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2014-05-29 Asterisk Development Team <asteriskteam@digium.com>
* Asterisk 1.8.28.0 Released.
2014-05-22 Asterisk Development Team <asteriskteam@digium.com>
* Asterisk 1.8.28.0-rc1 Released.
2014-05-22 15:47 +0000 [r414401] Richard Mudgett <rmudgett@digium.com>
* apps/app_meetme.c: app_meetme: Don't interrupt MOH for waitmarked
users. Occasionally, when the last marked user leaves the
conference, waitmarked users don't get MOH if MOH is supposed to
be played while a waitmarked user is waiting for another marked
user. * Made not interrupt MOH when the user is a waitmarked
user. The waitmarked user doesn't need to hear any leave
announcements from the conference as the user would have already
heard different leave announcements if they were enabled.
Apparently DAHDI occasionally sends unending non-silent streams
to these users or a normal user still in the conference has
continuous high background noise. These non-silent streams cause
MOH to be suspended while the never ending "announcement" is
played. Issue caused by ASTERISK-13680. AST-1349 #close Reported
by: Tyler Stewart Review:
https://reviewboard.asterisk.org/r/3543/
2014-05-22 13:58 +0000 [r414345] Matthew Jordan <mjordan@digium.com>
* UPGRADE.txt: UPGRADE: Add note for REF_DEBUG flag
2014-05-21 22:01 +0000 [r414269] Richard Mudgett <rmudgett@digium.com>
* channels/chan_local.c: chan_local: Only block media frames when a
generator is on both ends of a local channel. The fix for
ASTERISK-12292 was a bit too aggressive. You could have
generators pointed at each other on local channels but need to
get other kinds of frames such as DTMF or CONNECTED_LINE frames
accross.
2014-05-21 18:58 +0000 [r414214] Scott Griepentrog <sgriepentrog@digium.com>
* funcs/func_strings.c: pbx.c: prevent potential crash from
recursive replace() Recurisve usage of replace() resulted in
corruption of the temporary string storage and potential crash.
By changing the string to be allocated separtely per instance,
this is eliminated. ASTERISK-23650 #comment Reported by: Roel van
Meer ASTERISK-23650 #close Review:
https://reviewboard.asterisk.org/r/3539/
2014-05-19 13:31 +0000 [r414152] Alexandr Anikin <may@telecom-service.ru>
* addons/chan_ooh323.c: chan_ooh323: fix h323_log full path name *
fix to use astlogdir option for h323_log file instead of
hardcoded ASTERISK-23754 #close Reported by: Igor Goncharovsky
Patches: ooh323_logger_patch.diff
2014-05-16 20:00 +0000 [r413991-414067] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c: chan_dahdi: Fix analog dialtone detection.
* Check if waitingfordt (waitfordialtone) is enabled in
dahdi_read() to allow the DSP to operate early enough to detect
dialtone. * Made use the correct variable in
my_check_waitingfordt(). ASTERISK-23709 #close Reported by: Steve
Davies Patches: dialtone_detect_fix (license #5012) patch
uploaded by Steve Davies Review:
https://reviewboard.asterisk.org/r/3534/
* apps/app_meetme.c: app_meetme: Fix overwrite of DAHDI conference
data structure. Starting a conference recording using the admin
menu overwrites the DAHDI conference data structure used to
modify the admin user's conference mute mode. * Made no longer
pass the user's DAHDI conference data structure into the menu
functions. The menu now uses its own DAHDI conference data
structure to start the recording channel. * Moved the unlock
conf->playlock to before playing the conf-full message. No sense
keeping the lock while that prompt is playing. The user is never
going to get into the conference at that point.
2014-05-15 15:32 +0000 [r413949] Walter Doekes <walter+asterisk@wjd.nu>
* apps/app_dial.c, channels/chan_local.c, UPGRADE.txt:
chan_local+app_dial: Propagagate call answered elsewhere over
local channels. AST_FLAG_ANSWERED_ELSEWHERE was not propagated
back from local channels. It is now. That means that when a call
is picked up from a callgroup of local channels, the other
channels will now properly see it as "picked up". This occurs
when you use a construct like
Dial(Local/a@context&Local/b@context) where a@context and
b@context dial two chan_sip devices respectively. If one device
picks up, the other will not see "1 missed call" anymore. In this
respect, it now behaves the same as when doing Dial(SIP/a&SIP/b).
Review: https://reviewboard.asterisk.org/r/3540/
2014-05-14 15:27 +0000 [r413894] Walter Doekes <walter+asterisk@wjd.nu>
* res/res_musiconhold.c: res_musiconhold: Minor cleanup. Fix a few
free()'s that should be ast_free()'s. Reverted an old workaround
that isn't necessary. Reorder a tiny bit of code. Remove a bit of
commented-out code. Review:
https://reviewboard.asterisk.org/r/3536/
2014-05-13 14:32 +0000 [r413787-413832] Walter Doekes <walter+asterisk@wjd.nu>
* channels/chan_sip.c: chan_sip+CEL: Add missing ANSWER and PICKUP
events to INVITE/w/replaces pickup. When doing a "BLF-style call
pickup" -- an INVITE with Replaces: header -- the CEL log would
lack the ANSWER and PICKUP events. This patch adds the two
missing events to the handle_invite_replaces() function.
ASTERISK-22977 #close Review:
https://reviewboard.asterisk.org/r/3073/
* main/rtp_engine.c: rtp: Fix case typo in H263+ mime.
http://tools.ietf.org/html/rfc3555#section-4.2.6 says the
canonical mime subtype is "H263-1998", not "h263-1998". Original
code was added in r183101 on 2009-03-19 02:26:50 +0100. This
fixes issues with Polycom phones. ASTERISK-23665 #close
ASTERISK-23665 #comment Patch r3529.patch uploaded by Guillaume
Maudoux, backported by me. Review:
https://reviewboard.asterisk.org/r/3529/
2014-05-12 23:08 +0000 [r413714] Richard Mudgett <rmudgett@digium.com>
* configure, include/asterisk/autoconfig.h.in, configure.ac,
channels/sig_pri.c: chan_dahdi/sig_pri: Prevent unnecessary
PROGRESS events when overlap dialing is enabled. When overlap
dialing is enabled, the lack of inband audio available
information in the SETUP_ACKNOWLEDGE events causes an
interoperability problem with SIP. sig_pri doesn't know if there
is dialtone present when a SETUP_ACKNOWLEDGE is received so it
assumes it is there and posts an AST_CONTROL_PROGRESS frame. The
SIP channel driver then sends out a 183 Session Progress and
blocks the desired 180 Ringing message when the ALERTING message
comes in. * Made the configure script detect if the installed
version of libpri supports the SETUP_ACKNOWLEDGE enhancements. *
Using the new API, made generate an AST_CONTROL_PROGRESS frame on
an incoming SETUP_ACKNOWLEDGE message when the message indicates
inband audio is present instead of assuming that dialtone is
present. * Using the new API, made SETUP_ACKNOWLEDGE send out an
inband audio available indication only if dialtone is expected.
The change also makes the fallback behaviour of sending the
PROGRESS message better by sending it only if dialtone is
expected. * Changed receiving a PROCEEDING message to not
generate an AST_CONTROL_PROGRESS frame if the progress indication
ie indicates non-end-to-end-ISDN. This helps interoperability
with SIP. * Changed sending a PROCEEDING message in response to
an AST_CONTROL_PROCEEDING frame to not indicate inband audio
available. It was silly to do so anyway because the channel
driver doesn't know if inband audio is even available. This helps
interoperability with SIP. This patch and a corresponding change
in libpri work together to allow Asterisk to control the inband
audio available progress indication ie on the SETUP_ACKNOWLEDGE
message when dialtone is present. AST-1338 #close Reported by:
Tyler Stewart Review: https://reviewboard.asterisk.org/r/3521/
2014-05-09 23:02 +0000 [r413586-413592] Kinsey Moore <kmoore@digium.com>
* funcs/func_env.c: Fix 32bit build for func_env
* channels/chan_sip.c: Fix 32bit build for chan_sip
* channels/chan_dahdi.c, channels/sig_analog.c,
include/asterisk/astobj.h, main/event.c, funcs/func_iconv.c,
channels/sip/config_parser.c, apps/app_stack.c, res/res_odbc.c,
apps/app_adsiprog.c, res/res_calendar.c, main/udptl.c,
main/stun.c, main/frame.c, channels/chan_sip.c,
apps/app_festival.c, funcs/func_env.c, main/taskprocessor.c,
channels/chan_iax2.c, apps/app_getcpeid.c, res/res_monitor.c,
res/ael/pval.c, main/channel.c, main/manager.c,
formats/format_pcm.c, funcs/func_srv.c, main/file.c,
main/callerid.c, main/app.c, channels/chan_alsa.c, main/adsi.c,
pbx/pbx_dundi.c, main/stdtime/localtime.c, res/res_fax_spandsp.c,
main/sched.c, res/res_rtp_asterisk.c, cel/cel_pgsql.c,
cdr/cdr_adaptive_odbc.c, res/res_musiconhold.c,
channels/chan_gtalk.c, channels/sig_pri.c, res/res_srtp.c,
main/io.c, channels/chan_jingle.c, channels/chan_phone.c,
funcs/func_enum.c, res/res_config_odbc.c, apps/app_minivm.c,
res/res_agi.c, main/features.c, apps/app_dumpchan.c,
main/abstract_jb.c, main/logger.c, apps/app_sms.c,
main/audiohook.c, pbx/pbx_config.c, main/bridging.c, main/dsp.c,
apps/app_voicemail.c, apps/app_dial.c,
res/res_calendar_exchange.c, main/security_events.c,
res/res_fax.c, res/res_timing_dahdi.c, funcs/func_sysinfo.c,
main/utils.c, main/devicestate.c, res/res_jabber.c,
res/res_pktccops.c, main/cli.c, main/data.c, cel/cel_odbc.c,
channels/chan_skinny.c, main/asterisk.c,
channels/sip/include/sip.h, channels/chan_mgcp.c, main/xmldoc.c,
channels/chan_unistim.c, main/pbx.c,
res/res_calendar_icalendar.c, channels/chan_local.c,
main/rtp_engine.c, main/ccss.c, main/translate.c,
res/res_crypto.c, res/res_calendar_caldav.c, main/aoc.c,
pbx/dundi-parser.c, main/cel.c, apps/app_queue.c, main/enum.c,
channels/iax2-parser.c, main/config.c, res/res_calendar_ews.c,
main/netsock.c, main/loader.c: Allow Asterisk to compile under
GCC 4.10 This resolves a large number of compiler warnings from
GCC 4.10 which cause the build to fail under dev mode. The vast
majority are signed/unsigned mismatches in printf-style format
strings.
2014-05-08 00:33 +0000 [r413485] Joshua Colp <jcolp@digium.com>
* apps/app_queue.c, main/manager.c: app_queue: Extend documentation
for various Manager actions and events.
2014-05-07 17:46 +0000 [r413396] Mark Michelson <mmichelson@digium.com>
* res/res_config_odbc.c: Fix encoding of custom prepare extra data.
Patches: res_config_odbc-take2.patch by John Hardin (License
#6512)
2014-05-06 16:57 +0000 [r413304] Mark Michelson <mmichelson@digium.com>
* res/res_config_odbc.c: Ensure that all parts of SQL UPDATEs and
DELETEs are encoded. Patches: res_config_odbc.patch by John
Hardin (License #6512)
2014-05-02 20:21 +0000 [r413224-413241] Mark Michelson <mmichelson@digium.com>
* res/res_config_odbc.c: Prevent crashes in res_config_odbc due to
uninitialized string fields. Patches: odbc-crash.patch by John
Hardin (License #6512)
* res/res_config_pgsql.c: Return the number of rows affected by a
SQL insert, rather than an object ID. The realtime API specifies
that the store callback is supposed to return the number of rows
affected. res_config_pgsql was instead returning an Oid cast as
an int, which during any nominal execution would be cast to 0.
Returning 0 when more than 0 rows were inserted causes problems
to the function's callers. To give an idea of how strange code
can be, this is the necessary code change to fix a device state
issue reported against chan_pjsip in Asterisk 12+. The issue was
that the registrar would attempt to insert contacts into the
database. Because of the 0 return from res_config_pgsql, the
registrar would think that the contact was not successfully
inserted, even though it actually was. As such, even though the
contact was query-able and it was possible to call the endpoint,
Asterisk would "think" the endpoint was unregistered, meaning it
would report the device state as UNAVAILABLE instead of
NOT_INUSE. The necessary fix applies to all versions of Asterisk,
so even though the bug reported only applies to Asterisk 12+, the
code correction is being inserted into 1.8+. Closes issue
ASTERISK-23707 Reported by Mark Michelson
2014-04-23 17:47 +0000 [r412922] Richard Mudgett <rmudgett@digium.com>
* main/http.c: http: Fix spurious ERROR message in responses with
no content. Backport -r411687 and fix the fix because
content_length is the length of out plus the length of the file
controlled by fd. When a response has an out content length of 0,
fwrite would be called to write a buffer with no data in it. This
resulted in the following classic error message: [Apr 3 11:49:17]
ERROR[26421] http.c: fwrite() failed: Success This patch makes it
so that we only attempt to write the content of out if the out
string is non-zero.
2014-04-21 17:51 +0000 [r412764-412821] Jonathan Rose <jrose@digium.com>
* CHANGES: chan_sip: trust_id_outbound CHANGES message improvement
(closes issue AST-1301) (closes issue ASTERISK-19465) Reported
by: Krzysztof Chmielewski
* CHANGES: Typo in CHANGES
2014-04-21 15:50 +0000 [r412745] Kinsey Moore <kmoore@digium.com>
* main/manager.c, main/http.c: HTTP: Add TCP_NODELAY to accepted
connections This adds the TCP_NODELAY option to accepted
connections on the HTTP server built into Asterisk. This option
disables the Nagle algorithm which controls queueing of outbound
data and in some cases can cause delays on receipt of response by
the client due to how the Nagle algorithm interacts with TCP
delayed ACK. This option is already set on all non-HTTP AMI
connections and this change would cover standard HTTP requests,
manager HTTP connections, and ARI HTTP requests and websockets in
Asterisk 12+ along with any future use of the HTTP server.
Review: https://reviewboard.asterisk.org/r/3466/
2014-04-21 15:25 +0000 [r412744] Jonathan Rose <jrose@digium.com>
* channels/chan_sip.c, configs/sip.conf.sample, CHANGES,
channels/sip/include/sip.h: chan_sip: Add sendrpid trust options
In r411189, some behavior was changed which made sendrpid
behavior act in a more trusting manner by sending full user data
for peers set with private caller presence in P-Asserted-Identity
headers. Since this changed long time expected behaviors, we
decided to pull that patch when that was pointed out by the
community. Instead, this patch provides a trust_id_outbound
setting which will expose the data per RFC-3325 if set to 'yes'
and simply not send the PAI/RPID headers at all if set to 'no'.
By default trust_id_outbound will be set to 'legacy' which will
preserve the behavior prior to these patches. Extra special
thanks to Walter Doekes for providing advice and feedback.
(closes issue AST-1301) (closes issue ASTERISK-19465) Reported
by: Krzysztof Chmielewski Review:
https://reviewboard.asterisk.org/r/3447/
2014-04-19 01:01 +0000 [r412655] Matthew Jordan <mjordan@digium.com>
* apps/app_sms.c: app_sms: Fix uninitialized values; hangup channel
when REL is sent successfully This patch fixes two issues in
app_sms: (1) Firstly, the 'flags' field on the stack in
sms_exec() is uninitialised, causing it to use the wrong protocol
in some cases. This patch correctly initializes the flags fields.
(2) Secondly, when disconnect supervision is not working or
inbanddisconnect=yes is set in chan_dahdi.conf, app_sms was
failing to terminate the call after it sent the REL(ease) message
and the peer stopped talking to it. This patch fixes the code to
handle the 'bad stop bit' message more gracefully in that case,
and hang up the call. Review:
https://reviewboard.asterisk.org/r/1392/ ASTERISK-18331 #close
Reported by: David Woodhouse patches: asterisk-fix-sms.patch
uploaded by David Woodhouse (License 5754)
2014-04-18 17:12 +0000 [r412585] Rusty Newton <rnewton@digium.com>
* sounds/sounds.xml, sounds/Makefile: sounds: Fix Sounds Makefile
and XML that didn't support new sound prompt sets In
sounds/Makefile 1 Adds and moves some lines necessary for the
en_GB core set. I'm just following how the other sets are defined
here. 2 removes the ES extra sounds related lines as we don't
have ES extra sound sets. In sounds/sounds.xml 3 Adds
<support_level> definitions to all the sound sets as we have
these defined in 11,12,Trunk, but not in 1.8 4 Adds member
definitons for EN_AU, EN_GB, IT for core sound sets, and EN_GB in
extra sound sets ASTERISK-23550 Reported by: Rusty Newton Review:
https://reviewboard.asterisk.org/r/3464/
2014-04-17 20:23 +0000 [r412480] Matthew Jordan <mjordan@digium.com>
* channels/chan_oss.c: channels/chan_oss: Fix compilation problem
on SmartOS/Illumos/SunOS THis patch fixes an issue in chan_oss
when building on certain platforms. It ensures that soundcard.h
is found. Review: https://reviewboard.asterisk.org/r/3426 Note
that this patch is a part of the patch on ASTERISK-23576; the
Makefile portion only applies to Asterisk 11+. (issue
ASTERISK-23576) Reported by: Sebastian Wiedenroth patches:
fix-sunos.diff uploaded by Sebastian Wiedenroth (License 6597)
2014-04-15 15:21 +0000 [r412328] Jonathan Rose <jrose@digium.com>
* configs/sip.conf.sample, channels/chan_sip.c: Reverting r411189
so that it can be put up for public review --- r411189 | jrose |
2014-03-26 10:50:48 -0500 (Wed, 26 Mar 2014) | 12 lines chan_sip:
Send real CallerID information with P-Assserted-Identity
(RFC-3325) Prior to this patch, the P-Asserted-Identity header
would include anonymous caller id information which seems to go
against the point of the P-Asserted-Identity header. Now the real
caller ID information will be included in this header. Also, no
privacy header would be included. This patch adds 'Privacy: id'
to outgoing SIP messages that include the P-Asserted-Identity
header. (closes issue AST-1301) ---
2014-04-11 21:37 +0000 [r412225] Richard Mudgett <rmudgett@digium.com>
* apps/app_stack.c: app_stack: Add missing unlock in off-nominal
path of STACK_PEEK function. ASTERISK-23620 #close Reported by:
Bradley Watkins Patches: ASTERISK-23620_unlock_oldlist.patch
(license #5021) patch uploaded by Bradley Watkins
2014-04-11 01:33 +0000 [r412114] Matthew Jordan <mjordan@digium.com>
* main/utils.c, main/astobj2.c, contrib/scripts/refcounter.py
(added), main/asterisk.c, build_tools/cflags.xml,
include/asterisk/utils.h, channels/chan_sip.c,
include/asterisk/astobj2.h, main/logger.c: main/astobj2: Make
REF_DEBUG a menuselect item; improve REF_DEBUG output This patch
does the following: (1) It makes REF_DEBUG a meneselect item.
Enabling REF_DEBUG now enables REF_DEBUG globally throughout
Asterisk. (2) The ref debug log file is now created in the
AST_LOG_DIR directory. Every run will now blow away the previous
run (as large ref files sometimes caused issues). We now also no
longer open/close the file on each write, instead relying on
fflush to make sure data gets written to the file (in case the
ao2 call being performed is about to cause a crash) (3) It goes
with a comma delineated format for the ref debug file. This makes
parsing much easier. This also now includes the thread ID of the
thread that caused ref change. (4) A new python script instead
for refcounting has been added in the contrib/scripts folder.
Review: https://reviewboard.asterisk.org/r/3377/
2014-04-08 21:15 +0000 [r411960-411964] Richard Mudgett <rmudgett@digium.com>
* main/asterisk.c: Internal timing: Add notice that the -I and
internal_timing option are no longer needed. Add notice messages
during execution that the -I command line option and the
astersik.conf internal_timing option are no longer needed. The
internal timing functionality is now always enabled if there is a
timing module loaded. NOTE: Since the command line options and
the asterisk.conf config file are processed before the logging
system is initialized, the messages are output to stderr. Change
requested as a result of asterisk-dev list comments about the
commit for ASTERISK-22846 that removed the -I and internal_timing
options. Review: https://reviewboard.asterisk.org/r/3423/
* main/config.c: config: Fix CB_ADD_LEN() to work as originally
intended. Fix a long standing bug in CB_ADD_LEN() behaving like
CB_ADD(). ASTERISK-23546 #close Reported by: Walter Doekes
2014-04-07 14:45 +0000 [r411807] Walter Doekes <walter+asterisk@wjd.nu>
* configs/res_odbc.conf.sample: configs: Clean up long line and
typo in res_odbc.conf.sample.
2014-04-04 18:32 +0000 [r411715] Richard Mudgett <rmudgett@digium.com>
* include/asterisk/options.h, main/asterisk.c, main/channel.c,
channels/chan_sip.c, configs/asterisk.conf.sample, UPGRADE.txt:
internal_timing: Remove the option and always make it enabled if
a timing module is loaded. The masquerade supertest frequently
fails because either the local channel chain doesn't completely
optimize out or the DTMF handshake doesn't completely get
accross. Local channel optimization requires frames flowing to
trigger when optimization can happen. When optimization happens
the media frame that triggered the optimization is dropped.
Sending DTMF requires frames to flow in the other direction for
timing purposes while sending nothing. If internal timing is not
enabled when MOH is playing, Asterisk switches to received timing
when an audio frame is received. With optimization dropping media
frames and MOH not sending frames unless it receives frames,
occasionaly there are no more frames being passed and the test
fails. * The asterisk command line -I option and the
asterisk.conf internal_timing option are removed. Asterisk now
always uses internal timing when needed if any timing module is
loaded. The issue ASTERISK-14861 did this quite awhile ago in
v1.4 but effectively is broken if other internal timing modules
besides DAHDI are used. The ast_read_generator_actions() now only
does received timing if it has no choice for frame generators
like MOH, silence, and playback streaming. * Cleaned up some code
dealing with frame generators in ast_deactivate_generator(),
generator_write_format_change(), ast_activate_generator(), and
ast_channel_stop_silence_generator(). ASTERISK-22846 #close
Reported by: Matt Jordan Review:
https://reviewboard.asterisk.org/r/3414/
2014-04-01 16:48 +0000 [r411584] Joshua Colp <jcolp@digium.com>
* apps/app_queue.c: app_queue: Fix a bug where realtime members
would be deleted during reload causing waiting callers to get
ejected. This patch causes realtime queue members to remain in
queues during the reload process. Previously these members would
be removed causing any waiting callers to be ejected from the
queue with a reason of "EXITEMPTY". ASTERISK-23547 #close
ASTERISK-23547 #comment Patch
app_queue_fix_realtime_reload_1.8_trunk.patch submitted by Italo
Rossi (license 6409) Review:
https://reviewboard.asterisk.org/r/3404/
2013-04-23 Asterisk Development Team <asteriskteam@digium.com>
* Asterisk 1.8.27.0 Released.
2013-04-21 Asterisk Development Team <asteriskteam@digium.com>
* Asterisk 1.8.27.0-rc2 Released.
* chan_sip: Add sendrpid trust options
In r411189, some behavior was changed which made sendrpid behavior
act in a more trusting manner by sending full user data for peers
set with private caller presence in P-Asserted-Identity headers.
Since this changed long time expected behaviors, we decided to pull
that patch when that was pointed out by the community. Instead, this
patch provides a trust_id_outbound setting which will expose the data
per RFC-3325 if set to 'yes' and simply not send the PAI/RPID headers
at all if set to 'no'. By default trust_id_outbound will be set to
'legacy' which will preserve the behavior prior to these patches.
Extra special thanks to Walter Doekes for providing advice and
feedback.
2014-03-28 Asterisk Development Team <asteriskteam@digium.com>
* Asterisk 1.8.27.0-rc1 Released.
2014-03-28 16:16 +0000 [r411462] Scott Griepentrog <sgriepentrog@digium.com>
* main/http.c, main/tcptls.c, main/manager.c: http: response body
often missing after specific request This patch works around a
problem with the HTTP body being dropped from the response to a
specific client and under specific circumstances: a) Client
request comes from node.js user agent "Shred" via use of
swagger-client library. b) Asterisk and Client are *not* on the
same host or TCP/IP stack In testing this problem, it has been
determined that the write of the HTTP body is lost, even if the
data is written using low level write function. The only solution
found is to instruct the TCP stack with the shutdown function to
flush the last write and finish the transmission. See review for
more details. ASTERISK-23548 #close (closes issue ASTERISK-23548)
Reported by: Sam Galarneau Review:
https://reviewboard.asterisk.org/r/3402/
2014-03-28 15:42 +0000 [r411372-411457] Matthew Jordan <mjordan@digium.com>
* UPGRADE.txt: UPGRADE: Note IAX2 compatibility issue between 1.4
and 1.8+ systems.
* res/res_config_odbc.c, res/res_odbc.exports.in, UPGRADE.txt,
res/res_odbc.c, configs/res_odbc.conf.sample,
include/asterisk/res_odbc.h: res_config_odbc/res_odbc: Fix
handling of non-text columns updates with empty values. This
patch fixes setting nullable integer columns to NULL instead of
an empty string, which fails for PostgreSQL, for example. The
current code is supposed to do so, but the check is broken. The
patch also allows the first column in the list to be a nullable
integer. This patch also adds a compatibility setting in
res_odbc.conf, allow_empty_string_in_nontext. It is enabled by
default. It should be disabled for database backends (such as
PostgreSQL) that require NULL instead of an empty string for
Integer columns. Review: https://reviewboard.asterisk.org/r/3375
(issue ASTERISK-23459) Reported by: zvision patches:
res_config_odbc.diff uploaded by zvision (License 5755)
* channels/sip/include/sip.h: chan_sip: Add MESSAGE request to
allowed methods The allowed methods advertised by chan_sip did
not previously note the MESSAGE request. Even in Asterisk 1.8, we
do accept in-dialog MESSAGE requests; we should advertise that we
support MESSAGE requests. ASTERISK-23504 #close ASTERISK-23504
#comment Reported by: Martin Kontsek ASTERISK-23504 #comment
Patch sip.h_patch.diff uploaded by Martin Kontsek (license 6587)
Review: https://reviewboard.asterisk.org/r/3396/
2014-03-27 19:06 +0000 [r411313] Corey Farrell <git@cfware.com>
* funcs/func_groupcount.c, funcs/func_callcompletion.c,
funcs/func_pitchshift.c, funcs/func_odbc.c, funcs/func_volume.c,
funcs/func_frame_trace.c, funcs/func_channel.c,
funcs/func_blacklist.c, funcs/func_callerid.c, apps/app_stack.c,
res/res_calendar.c, apps/app_jack.c, funcs/func_speex.c,
funcs/func_dialplan.c, channels/chan_sip.c, funcs/func_math.c,
apps/app_readexten.c, funcs/func_strings.c, res/res_jabber.c,
channels/chan_iax2.c, res/res_mutestream.c, funcs/func_global.c,
apps/app_speech_utils.c: Fix dialplan function NULL channel
safety issues (closes issue ASTERISK-23391) Reported by: Corey
Farrell Review: https://reviewboard.asterisk.org/r/3386/
2014-03-26 22:43 +0000 [r411243] Joshua Colp <jcolp@digium.com>
* main/say.c: say: Fix a bug where SayNumber in Polish tries to
play incorrect sound. This change fixes a bug where calling
SayNumber with a number divisible by 100 using the Polish
language would cause the code to attempt to play a sound file
with an empty name. (closes issue ASTERISK-23509) Reported by:
zvision Review: https://reviewboard.asterisk.org/r/3378/
2014-03-26 15:50 +0000 [r411189] Jonathan Rose <jrose@digium.com>
* channels/chan_sip.c, configs/sip.conf.sample: chan_sip: Send real
CallerID information with P-Assserted-Identity (RFC-3325) Prior
too this patch, the P-Asserted-Identity header would include
anonymous caller id information which seems to go against the
point of the P-Asserted-Identity header. Now the real caller ID
information will be included in this header. Also, no privacy
header would be included. This patch adds 'Privacy: id' to
outgoing SIP messages that include the P-Asserted-Identity
header. (closes issue AST-1301)
2014-03-25 15:50 +0000 [r411088] Kinsey Moore <kmoore@digium.com>
* channels/chan_sip.c: chan_sip: Fix incorrect use of timers If
update_provisional_keepalive() is called while
send_provisional_keepalive_full() is waiting on the PVT lock,
then pvt->provisional_keepalive_sched_id will be changed to a new
sched_id value by update_provisional_keepalive(), but that new
sched_id then may be overwritten with -1 by
send_provisional_keepalive_full(), killing the pvt's reference to
a schedule and "leaking" the reference. (closes issue
ASTERISK-22079) Review: https://reviewboard.asterisk.org/r/3368/
Reported by: Jamuel Starkey, Matteo, Leif Madsen, Steve Davies
Patches: provisional_keepalive_fix.diff uploaded by Steve Davies
(license 5012)
2014-03-24 21:36 +0000 [r411021] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: chan_sip: Always use fromdomain if set for
domain, even if callerid is set to restricted. (closes issue
ASTERISK-20841) Reported by: Kelly Goedert
2014-03-17 21:54 +0000 [r410710-410748] Russ Meyerriecks <rmeyerreicks@digium.com>
* main/callerid.c: !fixup: callerid: Logic error in checksum
processing Fixes syntax error in previous commit :-(
* main/callerid.c: callerid: Logic error in checksum processing
Callerid checksum-ing was being handled incorrectly here. When
the checksum is calculated to be 0x00, it will perform 0x100-0x00
which results in 0x100. This value will then fail the otherwise
correct callerid message. This patch changes the logic to simply
add the calculated checksum to the transmitted 2's compliment
checksum. Review: https://reviewboard.asterisk.org/r/3356/
(closes issue ASTERISK-23488)
2014-03-10 17:00 +0000 [r410380] Richard Mudgett <rmudgett@digium.com>
* main/http.c: AST-2014-001: Stack overflow in HTTP processing of
Cookie headers. Sending a HTTP request that is handled by
Asterisk with a large number of Cookie headers could overflow the
stack. Another vulnerability along similar lines is any HTTP
request with a ridiculous number of headers in the request could
exhaust system memory. (closes issue ASTERISK-23340) Reported by:
Lucas Molas, researcher at Programa STIC, Fundacion; and Dr.
Manuel Sadosky, Buenos Aires, Argentina
2014-03-10 13:15 +0000 [r410308] Kinsey Moore <kmoore@digium.com>
* channels/chan_sip.c: AST-2014-002: chan_sip: Exit early on bad
session timers request This change allows chan_sip to avoid
creation of the channel and consumption of associated file
descriptors altogether if the inbound request is going to be
rejected anyway. (closes issue ASTERISK-23373) Reported by: Corey
Farrell Patches: chan_sip-earlier-st-1.8.patch uploaded by Corey
Farrell (license 5909) chan_sip-earlier-st-11.patch uploaded by
Corey Farrell (license 5909)
2014-03-07 22:50 +0000 [r410224] Corey Farrell <git@cfware.com>
* channels/chan_sip.c: chan_sip: Fix deadlock of monlock between
unload_module and do_monitor Release monlock before calling
pthread_join. This ensures do_monitor cannot freeze by locking
monlock during module unload. (closes issue ASTERISK-21406)
Reported by: Corey Farrell Review:
https://reviewboard.asterisk.org/r/3284/
2014-03-07 04:35 +0000 [r410105] Matthew Jordan <mjordan@digium.com>
* channels/chan_sip.c: chan_sip: Allow static realtime members to
be qualified during module load. When a static realtime peer with
qualify=yes is loaded, Asterisk will fail to send an OPTIONS
request due to the lastms being equal to 0. This results in the
peer being unable to receive calls from Asterisk because the
status is permanently UNKNOWN. This patch allows an OPTIONS
request to be sent during module load by ignoring the lastms
value on startup only. Review:
https://reviewboard.asterisk.org/r/3294/ (closes issue
ASTERISK-17523) Reported by: Maciej Krajewski Tested by:
wushumasters patches: realtime_fix_11.7.0.txt uploaded by Trevor
Peirce (license 6112)
2014-03-06 23:01 +0000 [r410043] Russell Bryant <russell@russellbryant.com>
* res/res_musiconhold.c: moh: fix a refcount error with realtime
MOH I observed a crash in res_musiconhold on an Asterisk 11
system using realtime MOH. Investigation of the backtrace showed
a corrupt mohclass, implying that it got destroyed before the
code expected it to. I went looking for reference counting errors
that could have caused this crash and this patch this result. It
contains 2 changes. 1) Remove a usless block of code that was
impossible to reach. There was even a comment indicating that it
was impossible to reach. The conditional includes
"!ast_test_flag(global_flags, MOH_CACHERTCLASSES)" and it's
inside of an if block with the opposite check
"ast_test_flag(global_flags, MOH_CACHERTCLASSES)". There's no
good reason to keep it around. 2) A similar block to #1 contained
a reference counting error. It stores state->class in the local
variable mohclass without increasing its reference count. The
reference count on mohclass is decremented at the end of the
function. This block of code probably very rarely runs, which
would help explain why this system was working fine for many
months before experiencing a crash. Review:
https://reviewboard.asterisk.org/r/3282/
2014-03-05 20:31 +0000 [r409916] Kinsey Moore <kmoore@digium.com>
* main/config.c: config: Fix inverted test The test of the result
of the stat() call was inverted such that its output was only
used if the call failed. This inverts the test so that the output
of stat() is used correctly. This was causing full reloads on
unchanged files. (closes issue ASTERISK-23383) Reported by: David
Woolley
2014-03-05 16:50 +0000 [r409833] David M. Lee <dlee@digium.com>
* main/config.c, configure, include/asterisk/autoconfig.h.in,
configure.ac: Corrected cross-platform stat nanosecond code When
nanosecond time resolution was added for identifying config file
changes, it didn't cover all of the myriad of ways that one might
obtain nanosecond time resolution off of struct stat. Rather than
complicate the #if even further figuring out one system from the
next, this patch directly tests for the three struct members I
know about today, and #ifdef's accordingly. Review:
https://reviewboard.asterisk.org/r/3273/
2014-03-05 12:04 +0000 [r409777] Sean Bright <sean@malleable.com>
* contrib/scripts/astgenkey, contrib/scripts/astgenkey.8: Fix
references to 'keys' CLI commands in astgenkey
2014-03-05 05:10 +0000 [r409705] Igor Goncharovskiy <igor.goncharovsky@gmail.com>
* channels/chan_unistim.c: Add update_peer function to
unistim_rtp_glue, improve other unistim_rtp_glue functions
conforming to other channel drivers. Do not forget auto-detected
and user-selected phone settings on 'unistim reload'
2014-03-04 19:32 +0000 [r409623] Michael L. Young <elgueromexicano@gmail.com>
* funcs/func_audiohookinherit.c: func_audiohookinheritance: Check
If A Channel Was Specified This patch prevents a crash when using
the function audiohookinheritance without setting the channel.
(closes issue ASTERISK-23104) Reported by: Joel Vandal Tested by:
Joel Vandal Patches:
asterisk-23104_audiohook_inherit_no_channel-11.diff uploaded by
Michael L. Young (license 5026) Review:
https://reviewboard.asterisk.org/r/3272/
2014-03-04 16:50 +0000 [r409521-409566] Kinsey Moore <kmoore@digium.com>
* main/astobj2.c: AO2: Add an assert for bad objects This adds an
assert that will only be active if Asterisk is compiled with
DO_CRASH and allows the testsuite to fail tests that would
otherwise require log file parsing.
* main/rtp_engine.c: rtp_engine: Clean up after a failed remote
bridge Upon failure of an INVITE transaction meant to initiate a
remote native bridge, rtp_engine.c would not clean up
non-reference-counted bridge instance pointers leaving a dangling
pointer which was being used to perform a local native bridge
after the other channel had hung up. This lead to dereferencing
into freed memory and plenty of AO2 errors. This change allows
the remote native bridge loop to clean up properly when the
bridge fails. (closes issue ASTERISK-23310) Reported by: Jeremy
Laine
2014-03-04 14:50 +0000 [r409472] Sean Bright <sean@malleable.com>
* channels/chan_sip.c: Minor whitespace change to 'sip show peers'
output. (closes issue ASTERISK-23406) Reported by: ibercom Tested
by: ibercom Patches: asterisk-11.patch uploaded by ibercom
2014-03-04 13:39 +0000 [r409436] Walter Doekes <walter+asterisk@wjd.nu>
* Makefile: buildsystem: Unbreak 'make -qp' on 1.8. r408083 caused
trouble with make -qp. Backport r408193 to 1.8 as well. (closes
issue ASTERISK-23382) Reported by: Corey Farrell
2014-03-03 02:06 +0000 [r409361] Matthew Jordan <mjordan@digium.com>
* main/asterisk.c: doxygen: Tweak the link back to ye olde Digium
website
2014-03-02 10:58 +0000 [r409308] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
* Makefile.rules: Makefile: replace -O6 with -O3 -O6 is not a legal
option of gcc. Unofficially gcc considers it to be equivalent of
-O3. clang chalks on it, though. This commit sets the default
optimization flag to be -O3, like gcc actually considered it.
Review: https://reviewboard.asterisk.org/r/3280/
2014-02-28 21:00 +0000 [r409156-409207] Richard Mudgett <rmudgett@digium.com>
* channels/chan_sip.c: chan_sip: Add precautionary p->owner checks.
* Add precautionary p->owner checks in sip_hangup(),
get_refer_info(), get_also_info(), and
interpret_t38_parameters(). * Simplify some tangled logic in
get_refer_info(), get_also_info(), and add_rpid(). * Removed some
dead code in handle_request_invite(). (closes issue
ASTERISK-23323) Reported by: Walter Doekes Patches:
issueA23323-more_p_owner_checks-1.8.x.patch (license #5674)
uploaded by wdoekes (modified)
issueA23323-more_p_owner_checks-11.x.patch (license #5674)
uploaded by wdoekes (modified)
issueA23323-more_p_owner_checks-12.x.patch (license #5674)
uploaded by wdoekes (modified)
issueA23323-more_p_owner_checks-trunk.patch (license #5674)
uploaded by wdoekes (modified)
* channels/chan_sip.c: chan_sip: Fix crash in
ast_channel_hangupcause_set(). * Fix crash in
ast_channel_hangupcause_set() because p->owner not checked before
calling. Regression introduced by the fix for ASTERISK-22621.
(closes issue ASTERISK-23135) Reported by: OK (issue
ASTERISK-23323) Reported by: Walter Doekes
2014-02-27 16:23 +0000 [r409077] David M. Lee <dlee@digium.com>
* utils/astman.c: Fix memory stomping bug in astman. This memset
complained in dev mod on my Ubuntu box. The memset is both
unnecessary and dangerous. At this point, m hasn't been
initialized yet, so the memset will write off to whatever address
happens to be on the stack at the time.
2014-02-27 15:59 +0000 [r409052] Corey Farrell <git@cfware.com>
* res/res_fax.c, configs/res_fax.conf.sample: res_fax: Warn that
minrate=2400 is not valid for V.27 instead of failing load.
Change minrate from 2400 to 4800 on config reload in response to
changes from ASTERISK-22790 only. Any config with minrate of 2400
that would fail before r405693 will still fail. Comment out many
settings in res_fax.conf.sample. The defaults are set in
res_fax.c, so setting the same value in sample config does
nothing but make the sample config more fragile. (closes issue
ASTERISK-23231) Reported by: David Brillert Review:
https://reviewboard.asterisk.org/r/3261/
2014-02-27 12:39 +0000 [r409001] Matthew Jordan <mjordan@digium.com>
* include/asterisk/rtp_engine.h, main/rtp_engine.c: rtp_engine: fix
crash during remote native bridging when calling get_codecs When
two RTP channels are in a remote bridge, the remote bridging loop
in rtp_engine will periodically check to see if the two channels
can still be bridged. One of the many things it checks is whether
or not the codecs have changed on the channel. If the codec has
changed, it will break out of the loop to re-determine which type
of bridge is appropriate. In order to perform this check, the
ast_rtp_glue virtual table's get_codec callback is called for
each channel. The callback implementations assume that the
channel tech private is valid when called; as such, there has
always been some code in place to check whether or not the
channel pvt is NULL before calling. However, this check is
insufficient. The channels are unlocked during the remote
bridging loop. It is possible for a channel to get masqueraded
between the check for the pvt being NULL and the actual call to
get_codec. When this occurs, the callback is called with a ZOMBIE
channel, which now has a NULL pvt. Crash. While this has always
been possible in Asterisk 1.8, it is much more likely to occur in
Asterisk 11 and later versions due to the timing changes that
occur when getting the codec from a channel. Note that this is
much more likely to be reproduced on slow, boggy hardware running
Asterisk 11 - but fairly rarely otherwise. Also Note: This crash
was also caught by the various SIP blind transfer tests, in
addition to the bug report Alec filed. Review:
https://reviewboard.asterisk.org/r/3247/ (closes issue
ASTERISK-21737) Reported by: Alec Davis Tested by: Alec Davis
2014-02-25 17:41 +0000 [r408876] Rusty Newton <rnewton@digium.com>
* configs/voicemail.conf.sample: configs/voicemail.conf.sample -
Make mailcmd sample text more explicit Made the wording a bit
more explicit. Didn't really change the meaning.
2014-02-22 02:26 +0000 [r408785] Corey Farrell <git@cfware.com>
* utils/extconf.c, utils/conf2ael.c, res/ael/pval.c, main/pbx.c:
Remove extra defines of AST_PBX_MAX_STACK. * Ensure
AST_PBX_MAX_STACK is only defined in extconf.h and pbx.h. * Fix
incorrect function parameters in utils/extconf.c. (closes issue
ASTERISK-23141) Reported by: Maxim Review:
https://reviewboard.asterisk.org/r/3241/
2014-02-21 20:18 +0000 [r408642-408747] Kevin Harwell <kharwell@digium.com>
* apps/app_forkcdr.c: app_forkcdr: ForkCDR v option does not keep
CDR variables for subsequent records When the 'v' option is
specified to ForkCDR application, AST_CDR_FLAG_KEEP_VARS flag is
set only for the first CDR in the chain. So ForkCDR works fine
with this option only once. After the second and further calls to
ForkCDR, CDR variables get cleared on all CDRs besides the first
one and moved to the newly forked CDR. It always sets the
KEEP_VARS flag on the first CDR in the chain, instead of the most
recent CDR which is used as a base to fork a new CDR. This patch
sets KEEP_VARS flag on the most recent CDR on the stack (the CDR
used for forking). (closes issue ASTERISK-23260) Reported by:
zvision Patches: app_forkcdr.diff uploaded by zvision (license
5755)
* main/rtp_engine.c: rtp_engine: Output mixup in
${CHANNEL(rtpqos,audio,all)} Fixed the output of
CHANNEL(rtpqos,audio,all) to use txjitter instead of rxjitter.
(closes issue ASTERISK-23261) Reported by: rsw686 Patches:
rtpqos.patch uploaded by rsw686 (license 5887)
* channels/chan_sip.c, main/channel.c: channel.c: MOH is not
working for transferee after attended transfer Updated the code
to check to see if MOH is playing on the transferor and if so
then start it on the channel that replaces it during a
masquerade. Example scenario of the problem: Alice calls Bob and
then Bob begins the attended transfer process into a queue. Upon
going on hold Alice hears music and so does Bob once he is in the
queue. Bob then transfers Alice into the queue and then music for
Alice stops even though she should be hearing it since has now
replaced Bob in the queue. The problem that was occurring is that
once the channel was masqueraded the app (queues, confbridge,
etc...) had no way of knowing that the channel had just been
swapped out thus it did not start music for the present channel.
Credit to Olle Johansson for pointing me in the right direction
on this issue. (closes issue ASTERISK-19499) Reported by: Timo
Teräs Review: https://reviewboard.asterisk.org/r/3226/
2014-02-21 10:35 +0000 [r408589] Alexandr Anikin <may@telecom-service.ru>
* addons/ooh323c/src/ooCalls.h: Fix type of roundTripDelay
variables
2014-02-21 00:46 +0000 [r408536] Michael L. Young <elgueromexicano@gmail.com>
* apps/app_chanspy.c: app_chanspy: Documentation Update To Clarify
"x" Option When using the "x" option (specify a DTMF digit to
exit the application), it is not obvious in the documentation
that this only works when spying on a channel. If a channel being
used to spy on other channels is waiting to connect to a channel
or is no longer attached to a channel, the DTMF is ignored. As
noted on the issue tracker, since there are workarounds available
and this is a rarely used option we are opting for a
documentation change here. (closes issue ASTERISK-22661) Reported
by: Chris Hillman Patches:
asterisk-22661-doc-clarify-chan_spy.diff uploaded by Michael L.
Young (license 5026) Review:
https://reviewboard.asterisk.org/r/2990/
2014-02-20 02:39 +0000 [r408447] Rusty Newton <rnewton@digium.com>
* apps/app_queue.c: apps/app_queue - Fix incorrect Macro parameter
documentation Macro is executed on the called channel, not the
calling channel. (closes issue ASTERISK-23069) Reported By: Bryan
Anderson
2014-02-19 19:01 +0000 [r408387] Richard Mudgett <rmudgett@digium.com>
* main/config.c: config: Add file size and nanosecond resolution
fields to the cached modified config file information. Repeatedly
modifying config files and reloading too fast sometimes fails to
reload the configuration because the cached modification
timestamp has one second resolution. * Added file size and
nanosecond resolution fields to the cached config file
modification timestamp information. Now if the file size changes
or the file system supports nanosecond resolution the modified
file has a better chance of being detected for reload. * Added a
missing unlock in an off-nominal code path. (closes issue
AST-1303) Review: https://reviewboard.asterisk.org/r/3235/
2014-02-19 11:30 +0000 [r408328] Alexandr Anikin <may@telecom-service.ru>
* addons/ooh323c/src/ooCapability.c, addons/ooh323c/src/ooh245.c:
process receiveAndTransmit user input remote caps instead of
receive only send receiveAndTransmit user input our caps instead
of receive only
2014-02-16 03:14 +0000 [r408200] Matthew Jordan <mjordan@digium.com>
* main/pbx.c: pbx: Handle a completely empty dialplan during a
context merge It is highly unlikely, but - at least in Asterisk
12 - theoretically possible to load Asterisk with no dialplan
whatsoever. If that occurs, and some other module (that is not a
pbx module) attempts to merge its contexts into the dialplan, the
existing merge routine will crash. This is because it is not
insane, and rightly believes that you provided some sort of
dialplan, somewhere. This patch will gracefully merge the
contexts in such a case. Note that this is highly unlikely to
occur in 1.8/11, as features will most likely provide some
dialplan via parking. However, in Asterisk 12, parking is now
provided by res_parking, and hence may create its dialplan later.
(closes issue ASTERISK-23297) Reported by: CJ Oster Review:
https://reviewboard.asterisk.org/r/3222
2014-02-14 21:52 +0000 [r408142] Scott Griepentrog <sgriepentrog@digium.com>
* main/pbx.c: pbx: ast_custom_function_unregister resource leak In
pbx.c ast_custom_function_unregister(), a list of escalations
being removed from the list wasn't being free'd creating a leak.
This patch corrects that by freeing the records. Review:
https://reviewboard.asterisk.org/r/3213/ Reported by: Corey
Farrell Patches: acf_escalating_leak.patch uploaded by
coreyfarrell (license 5909)
2014-02-14 13:25 +0000 [r408083] Walter Doekes <walter+asterisk@wjd.nu>
* Makefile: buildsystem: Don't force main to depend on everything
else. Directory 'main' only needs to depend on embedded modules.
If no module embedding is selected, the dependency is dropped.
Review: https://reviewboard.asterisk.org/r/3212/
2014-02-14 01:22 +0000 [r408020] Rusty Newton <rnewton@digium.com>
* configs/agents.conf.sample: configs/agents.conf.sample - Remove
example for non-functional "goodbye" parameter The "goodbye"
parameter is not implemented in the source code, it does nothing.
(closes issue SWP-6518) Reported By: Steve Pitts
2014-02-10 16:33 +0000 [r407873] Walter Doekes <walter+asterisk@wjd.nu>
* res/res_config_pgsql.c: res_config_pgsql: Fix
ast_update2_realtime calls. Fix so multiple updates from a single
call works (add missing ','). Remove bogus ast_free's that
weren't supposed to be there. Moved a few spaces for readability.
Review: https://reviewboard.asterisk.org/r/3194/
2014-02-09 15:34 +0000 [r407817] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
* channels/chan_dahdi.c, /: chan_dahdi: handle DAHDI_EVENT_REMOVED
on a pri D-Channel When a DAHDI device is removed at run-time it
sends the event DAHDI_EVENT_REMOVED on each channel. This is
intended to signal the userspace program to close the respective
file handle, as the driver of the device will need all of them
closed to properly clean-up. This event has long since been
handled in chan_dahdi (chan_zap at the time). However the event
that is sent on a D-Channel of a "PRI" (ISDN) span simply gets
ignored. This commit adds handling for closing the file
descriptor (and shutting down the span, while we're at it). It
also adds a CLI command 'pri destroy span <N>' to destroy the
span and its DAHDI channels. Backported from trunk/12. Review:
https://reviewboard.asterisk.org/r/726/ ........ Merged revisions
394552 394567 from http://svn.asterisk.org/svn/asterisk/trunk
2014-02-07 20:42 +0000 [r407678-407764] Richard Mudgett <rmudgett@digium.com>
* channels/chan_iax2.c: chan_iax2: Add some more iaxs[] NULL checks
to a routine already full of them.
* channels/chan_iax2.c, include/asterisk/frame.h,
configs/iax.conf.sample: chan_iax2: Block unnecessary control
frames to/from the wire. Establishing an IAX2 call between
Asterisk v1.4 and v1.8 (or later) results in an unexpected call
disconnect. The problem happens because newer values in the enum
ast_control_frame_type are not consistent between the branch
versions of Asterisk. For example: 1) v1.4 calls v1.8 (or later)
using IAX2 2) v1.8 answers and sends a connected line update
control frame. (on v1.8 AST_CONTROL_CONNECTED_LINE = 22) 3) v1.4
receives the control frame as an end-of-q (on v1.4
AST_CONTROL_END_OF_Q = 22) 4) v1.4 disconnects the call once the
receive queue becomes empty. Several things are done by this
patch to fix the problem and attempt to prevent it from happening
again in the future: * Added a warning at the definition of enum
ast_control_frame_type about how to add new control frame values.
* Made block sending and receiving control frames that have no
reason to go over the wire. * Extended the connectedline iax.conf
parameter to also include the redirecting information updates. *
Updated the connectedline iax.conf parameter documentation to
include a notice that the parameter must be "no" when the peer is
an Asterisk v1.4 instance. (closes issue AST-1302) Review:
https://reviewboard.asterisk.org/r/3174/
2014-02-07 12:59 +0000 [r407622] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
* configs/indications.conf.sample: indications.conf: add stutter
tone; end properly * If the "stutter" (voicemail indication) tone
is indeed a stutter tone, and it ends with a constant tone, make
sure that it is the dial tone. This was done for India (in),
Mexico (mx) and the Philippines (ph). * If no "stutter" tone
exists for a country, provide one. This was done for Spain (es),
Malaysia (my) and Venezuela (ve). Review:
https://reviewboard.asterisk.org/r/3158/
2014-02-05 22:58 +0000 [r407511] Rusty Newton <rnewton@digium.com>
* formats/format_wav.c: formats/format_wav: enhancing log message
"Not a wav file" to be clear on what is supported Modifying the
log message to be more specific as to what is supported.
Specifically it seems format_wav supports only PCM encoded
versions with a lower-case '.wav' extension. (closes issues
ASTERISK-22310) Reported by: Jim Credland Review:
https://reviewboard.asterisk.org/r/3188/
2014-02-05 20:30 +0000 [r407455] Kinsey Moore <kmoore@digium.com>
* main/logger.c: Logger: Fix handling of absolute paths This fixes
path handling for log files so that an extra / is not appended to
the file path when the path is absolute (begins with /). This
would previously result in different but functionally equivalent
paths in the output of 'logger show channels'.
2014-02-04 19:48 +0000 [r407272-407337] Richard Mudgett <rmudgett@digium.com>
* include/asterisk/devicestate.h, main/devicestate.c: devicestate:
Make ast_devstate_changed_literal() return value and doxygen
consistent. Nothing actually cares about the value anyway.
(closes issue ASTERISK-23178) Reported by: Jonathan Rose
* configs/sip.conf.sample, main/tcptls.c: tcptls.c: Made TLS handle
a certificate chain file. Thanks to Guillaume Martres for doing
the necessary research to validate the change. (closes issue
ASTERISK-17727) Reported by: LN Patches:
use_certificate_chain.patch (license #5864) patch uploaded by st
documente_certificate_chain.patch (license #6576) patch uploaded
by Guillaume Martres
2014-02-04 02:19 +0000 [r407205] Joshua Colp <jcolp@digium.com>
* res/res_clialiases.c: res_clialiases: Fix crash when reloading
and re-aliasing an alias that is in use. The code assumed that
unregistering the alias would always succeed while in practice
this is not actually true. A common case is the "reload" command
itself. If the cli_aliases.conf configuration file was changed
and reload executed the command would fail to unregister and
ultimately point to freed memory. The reload process now checks
whether unregistering succeeded or not and if not the old CLI
alias is retained. (closes issue ASTERISK-19773) Reported by:
Joel Vandal (closes issue ASTERISK-22757) Reported by: Gareth
Blades
2014-02-01 00:22 +0000 [r407100] Corey Farrell <git@cfware.com>
* apps/app_stack.c: app_stack: protect against missing parameters
to STACK_PEEK and LOCAL_PEEK STACK_PEEK requires 2 parameters and
LOCAL_PEEK requires 1 parameter. This protects against situations
where those parameters are blank or missing by logging an error
and returning. (closes issue ASTERISK-23220) Reported by: James
Sharp
2014-01-31 23:18 +0000 [r407041] Matthew Jordan <mjordan@digium.com>
* apps/app_dial.c: app_dial: Allow macro/gosub pre-bridge execution
to occur on priorities The parsing for the destination of the
macro/gosub uses the '^' character to separate out context,
extension, and priority. However, the logic for the macro/gosub
execution was written such that it would only do the actual
macro/gosub jump if a '^' character existed. This doesn't apply
when the macro/gosub jump occurs in a priority/priority label.
This patch changes the logic so that the parsing still occurs,
but the jump will occur even for priorities/priority labels.
(issue ASTERISK-23164) Review:
https://reviewboard.asterisk.org/r/3154
2014-01-30 20:26 +0000 [r406933] Corey Farrell <git@cfware.com>
* main/udptl.c, res/res_rtp_asterisk.c: res_rtp_asterisk & udptl:
fix port selection to work with SELinux restrictions ast_bind to
a port reserved for another program by SELinux causes errno ==
EACCES. This caused random failures when binding rtp or udptl
sockets. Treat EACCES as a non-fatal error, try next port.
(closes issue ASTERISK-23134) Reported by: Corey Farrell
2014-01-29 00:36 +0000 [r406860] Russell Bryant <russell@russellbryant.com>
* configs/queues.conf.sample: queues.conf.sample Fix documented
default for persistentmembers Closes issue ASTERISK-22662
2014-01-28 23:02 +0000 [r406801] Kevin Harwell <kharwell@digium.com>
* cel/cel_radius.c, configure, include/asterisk/autoconfig.h.in,
configure.ac, cdr/cdr_radius.c: cdr_radius, cel_radius: build
agains libfreeradius-client Asterisk's RADIUS module currently
build against libradiusclient-ng, but this project has been
superseeded by libfreeradius-client. The API is 99% compatible
except that the header name has changed, the library name has
changed, and the configuration file location has changed. (closes
issue ASTERISK-22980) Reported by: Jeremy Lainé Patches:
freeradius-client.patch uploaded by sharky (license 6561)
2014-01-28 16:36 +0000 [r406721] Scott Griepentrog <sgriepentrog@digium.com>
* main/rtp_engine.c: rtp_engine: improved handling of get_rtp_info
failure In ast_rtp_instance_make_compatible(), after a failure of
channel tech call get_rtp_info() to return peer_instance, the
null pointer would be passed to ao2_ref, producing an error that
looked like a refernce counting problem but is not. This patch
corrects that and adds helpful LOG_ERROR messages to indicate
which failure path occurred. (issue AST-1276) Review:
https://reviewboard.asterisk.org/r/3156/
2014-01-27 20:34 +0000 [r406566-406643] Russell Bryant <russell@russellbryant.com>
* main/config.c: Allow nested #includes in extconfig.conf
extconfig.conf was hard-coded to not allow nested includes for
some reason. The code has been this way since a patch was merged
for ASTERISK-3333 (revision 4889), which was a significant update
to this code ("Merge config updates"). I can't figure out any
good reason why this should be limited. This patch just removes
the limit and uses the default nesting depth limit. Closes issue
ASTERISK-17837 Review: https://reviewboard.asterisk.org/r/3159/
* main/file.c, include/asterisk/channel.h, main/channel.c: Protect
ast_filestream object when on a channel The ast_filestream object
gets tacked on to a channel via chan->timingdata. It's a
reference counted object, but the reference count isn't used when
putting it on a channel. It's theoretically possible for another
thread to interfere with the channel while it's unlocked and
cause the filestream to get destroyed. Use the astobj2 reference
count to make sure that as long as this code path is holding on
the ast_filestream and passing it into the file.c playback code,
that it knows it's valid. Bug reported by Leif Madsen. Review:
https://reviewboard.asterisk.org/r/3135/
2014-01-26 22:59 +0000 [r406514] Richard Mudgett <rmudgett@digium.com>
* main/tcptls.c: tcptls.c: Add missing cleanup on off nominal path.
2014-01-24 22:56 +0000 [r406417] Richard Mudgett <rmudgett@digium.com>
* main/cel.c: CEL: Protect data structures during reload and
shutdown. The CEL data structures need to be protected during a
configuration reload and shutdown. Asterisk crashed during a
shutdown because CEL events were still in flight and the CEL data
structures were already destroyed. * Protected the appset and
linkedids ao2 containers using the reload_lock. * Added NULL
checks before use of the appset and linkedids ao2 containers in
case the CEL module is already shutdown. * Fixed overloading of
the linkedids held objects reference count. During shutdown any
held objects would be leaked. * Fixed memory leak of linkedids
held objects if the LINKEDID_END is not being tracked. The
objects in the linkedids container were not removed if the
LINKEDID_END event is not used. * Added access protection to the
appset container during the CLI "cel show status" command. * Made
CEL config reload not set defaults if the cel.conf file is
invalid. (closes issue AST-1253) Reported by: Guenther Kelleter
Review: https://reviewboard.asterisk.org/r/3127/
2014-01-24 20:57 +0000 [r406360] Jonathan Rose <jrose@digium.com>
* res/res_config_pgsql.c: res_config_pgsql: Fix a memory leak and
use RAII_VAR for cleanup when practical Review:
https://reviewboard.asterisk.org/r/3141/
2014-01-24 20:56 +0000 [r406359] Richard Mudgett <rmudgett@digium.com>
* main/manager.c: manager: Register atexit shutdown routine only
once. * Made register atexit shutdown routine only once in
__init_manager(). * Fixed some initial load failure conditions in
__init_manager(). * Made reset options to defaults on reload when
the reload will actually happen. * Fixed the order of
unreferencing a session object in session_destroy(). * Removed
unnecessary container traversals of the white/black filters
during session_destructor() and manager_free_user(). * ast_free()
does not need a NULL check before calling.
2014-01-22 22:16 +0000 [r406241] Scott Griepentrog <sgriepentrog@digium.com>
* utils/extconf.c, main/pbx.c: pbx.c: Pre-initialize timezone to
avoid crash on destroy In ast_build_timing, initialize the
timezone value to NULL in order to avoid deferencing an
uninitialized value later when calling ast_destroy_timing. The
timezone value could be uninitialized if ast_build_timing were to
fail due to a zero length time string. (closes issue
ASTERISK-22861) Reported by: Sebastian Murray-Roberts Review:
https://reviewboard.asterisk.org/r/3134/ Patches:
ast_build_timing-initialize-timezone.patch uploaded by
coreyfarrell (license 5909)
2014-01-22 18:27 +0000 [r406170] Kinsey Moore <kmoore@digium.com>
* channels/chan_sip.c: chan_sip: Decline image streams on
unsupported transports This change allows chan_sip to decline
individual image streams over unsupported transports in the SDP
of the 200 response. Previously, an image stream offer with
RTP/AVP as the transport would cause chan_sip to respond with a
488. (closes issue ASTERISK-22988) Reported by: adomjan Original
patch by: adomjan
2014-01-21 20:54 +0000 [r406079] Walter Doekes <walter+asterisk@wjd.nu>
* main/manager.c, configs/manager.conf.sample: manager: Clarify
eventfilter documentation. Textual changes only. Review:
https://reviewboard.asterisk.org/r/3133/
2014-01-21 19:58 +0000 [r406037] Kinsey Moore <kmoore@digium.com>
* res/res_pktccops.c, channels/chan_mgcp.c: chan_mgcp: Enforce
locking for oseq This restricts direct usage of global oseq so
that all accesses are locked and threads are not racing to get
oseq values that they did not claim. This also fixes a build
error in res_pktccops under dev mode. (closes issue
ASTERISK-23100) Reported by: adomjan Patch by: adomjan
2014-01-20 21:58 +0000 [r405926] Richard Mudgett <rmudgett@digium.com>
* channels/sig_pri.c: chan_dahdi/PRI: Suppress CONNECTED_LINE
updates when nothing in the udpate is valid. * Also simplified
some subddress handling code. (closes issue ASTERISK-23008)
Reported by: Michael Cargile
2014-01-17 15:39 +0000 [r405791] Rusty Newton <rnewton@digium.com>
* channels/chan_sip.c, doc/asterisk.8, main/features.c,
configs/sip.conf.sample, apps/app_queue.c, apps/app_transfer.c,
channels/chan_iax2.c: Documentation: doc fixes across various
parts of the code for ASTERISK issues 23061,23028,23046,23027
Fixes typos of "transfered" instead of "transferred" in various
code. Fixes incorrect gosub param help text for app_queue. Fixes
Asterisk man pages containing unquoted minus signs. Adds note
about the "textsupport" option in sip.conf.sample. (issue
ASTERISK-23061) (issue ASTERISK-23028) (issue ASTERISK-23046)
(issue ASTERISK-23027) (closes issue ASTERISK-23061) (closes
issue ASTERISK-23028) (closes issue ASTERISK-23046) (closes issue
ASTERISK-23027) Reported by: Eugene, Jeremy Laine, Denis
Pantsyrev Patches: transferred.patch uploaded by Jeremy Laine
(license 6561) hyphen.patch uploaded by Jeremy Laine (license
6561) sip.conf.sample.patch uploaded by Eugene (license 6360)
2014-01-16 18:52 +0000 [r405656-405692] Kevin Harwell <kharwell@digium.com>
* UPGRADE.txt: res_fax: check_modem_rate() returned incorrect rate
for V.27 Added some text to UPGRADES.txt about the V.27 mode rate
changes in r405656. (issue ASTERISK-22790) Reported by: Paolo
Compagnini
* res/res_fax.c, configs/res_fax.conf.sample: res_fax:
check_modem_rate() returned incorrect rate for V.27 According to
the new standard for V.27 and V.32 they are able to transmit at a
bit rate of 4,800 or 9,600. The check_mode_rate function needed
to be updated to reflect this. Also, because of this change the
default 'minrate' value was updated to be 4800. (closes issue
ASTERISK-22790) Reported by: Paolo Compagnini Patches:
res_fax.txt uploaded by looserouting (license 6548)
2014-01-15 16:34 +0000 [r405581] Joshua Colp <jcolp@digium.com>
* cel/cel_manager.c: cel_manager: Don't crash if configuration file
is invalid. The cel_manager module did not properly handle the
case where the configuration file was invalid. The module will
now output a warning message and disable itself if this occurs.
Reported by: Bryan Walters
2014-03-03 Asterisk Development Team <asteriskteam@digium.com>
* Asterisk 1.8.26.0 Released.
2014-01-14 Asterisk Development Team <asteriskteam@digium.com>
* Asterisk 1.8.26.0-rc1 Released.
2014-01-14 18:35 +0000 [r405433-405486] Scott Griepentrog <sgriepentrog@digium.com>
* channels/chan_sip.c: chan_sip: No BYE message sent after INVITE
with Replaces Setting channel state DOWN is an unnecessary step
that was only being done in handle_invite_replaces(). This
changes that by removing the call and reducing locking. (closes
issue ASTERISK-23010) Reported by: Ryan Tilton Review:
https://reviewboard.asterisk.org/r/3116/
* channels/chan_sip.c: chan_sip: fix Local From tag on outbound
register regression In ASTERISK-12117, an improvement to insure
consistant local from tags on outbound registrations resulted in
an undesirable behavior - caused by leftover unexpired sip_pvt
dialogs (with the previous cseq number), resulting in many
uncessary REGISTER requests. Instead of significant rework of
transmit_register(), this change deletes the dialogs after a 200
OK response indiciating a successful registration, keeping the
old dialogs from interfering with normal operation. (closes issue
ASTERISK-22946) Reported by: Stephan Eisvogel Review:
https://reviewboard.asterisk.org/r/3109/
2014-01-14 15:31 +0000 [r405379] Matthew Jordan <mjordan@digium.com>
* channels/chan_sip.c: chan_sip: Hangup transferer/transferee when
transfer to Parking fails When performing a SIP transfer to a
Park extension, if the Park fails, chan_sip will currently not
hang up either the transferer or the transfer target. This
results in the channels being orphaned with no thread to service
frames, resulting in stuck channels. This patch immediately hangs
up the two channels if a Park fails. (closes issue
ASTERISK-22834) Reported by: rsw686 (closes issue ASTERISK-23047)
Reported by: Tommy Thompson Review:
https://reviewboard.asterisk.org/r/3107
2014-01-09 14:11 +0000 [r405160] Walter Doekes <walter+asterisk@wjd.nu>
* apps/app_dumpchan.c: "Minimun" typo.
2014-01-08 16:00 +0000 [r405033-405090] Kinsey Moore <kmoore@digium.com>
* configure, configure.ac, pbx/pbx_lua.c: pbx_lua: Add support for
Lua 5.2 This adds support for Lua 5.2 in pbx_lua which is
available on newer operating systems. (closes issue
ASTERISK-23011) Review: https://reviewboard.asterisk.org/r/3075/
Reported by: George Joseph Patch by: George Joseph
* UPGRADE.txt: UPGRADE: Add a note about non-functionality Add a
note that the "retry on 403 response to REGISTER" for chan_sip is
non-functional in the versions in which it was first introduced.
* channels/chan_sip.c: Add the missing part of r400140 When the
patch to add retry-on-forbidden-response was committed, part of
the patch for chan_sip was not committed which caused the feature
to be entirely nonfunctional. This corrects the code in question.
(closes issue ASTERISK-17138) Review:
https://reviewboard.asterisk.org/r/2874
2014-01-06 17:31 +0000 [r404951] Scott Griepentrog <sgriepentrog@digium.com>
* funcs/func_strings.c: func_strings: fix for memmove patch test In
r404674 the AST_TEST_DEFINE(test_REPLACE) test was added that
made use of a function that doesn't exist in 1.8. This fixes that
by reverting to directly accessing chan varshead. Reported by:
Tzafrir Cohen (issue ASTERISK-22910)
2014-01-03 22:06 +0000 [r404861] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
* main/asterisk.c: asterisk.c: supress live_dangerously warning on
rasterisk Even since the fixes of AST-2013-007, Asterisk prints
the following warning on startup if the user decided to live
dangerously: Privilege escalation protection disabled! See
https://wiki.asterisk.org/wiki/x/1gKfAQ for more details. This
message is intended for the logs and interactive startup. No need
for it to appear on a remote console. This commit removes it from
there. (closes issue ASTERISK-23084) Review:
https://reviewboard.asterisk.org/r/3101/
2014-01-03 21:57 +0000 [r404742-404857] Kevin Harwell <kharwell@digium.com>
* cel/cel_pgsql.c: cel_pgsql: module not correctly reloading Upon
reload the module unconditionally "unloaded" the module (freeing
memory and setting pointers to NULL) and then when attempting a
"load" if the config file had not changed then nothing would be
reinitialized. By moving the "unload" to occur conditionally
(reload only) after an attempted configuration load, but before
module "loading" alleviates the issue. The module now
loads/unloads/reloads correctly. (closes issue ASTERISK-22871)
Reported by: Matteo
* channels/chan_dahdi.c: chan_dahdi: dahdi show channels slices PRI
channel dnid on output dahdi show channels output slices the
callerid (which is dnid copied over on PRI channels). If the
channel naming structures look like: 'DAHDI/i1/1408409XXXX-6'
then the output slices 1408409XXXX down to 1408409XXX. This patch
just opens it up to 15 chars so you can see the whole thing.
(closes issue ASTERISK-22918) Reported by: outtolunc Patches:
svn_chan_dahdi.c.format12_15.diff.txt uploaded by outtolunc
(license 5198)
* apps/app_meetme.c, channels/chan_unistim.c: chan_unistim.c,
app_meetme: compiler warnings Fixed a couple of compiler warnings
(errors in 'dev-mode') given by gcc version 4.8.1. The one in
app_meetme involved the 'sizeof-pointer-memaccess' (see:
http://gcc.gnu.org/gcc-4.8/porting_to.html) warning. The one in
chan_unistim was issuing an array out of bounds message. Fixed
both so they would no longer issue warnings and can compile again
in 'dev-mode'. Review: https://reviewboard.asterisk.org/r/3098/
2014-01-02 19:32 +0000 [r404674] Scott Griepentrog <sgriepentrog@digium.com>
* funcs/func_strings.c: func_strings: use memmove to prevent
overlapping memory on strcpy When calling REPLACE() with an empty
replace-char argument, strcpy is used to overwrite the the
matching <find-char>. However as the src and dest arguments to
strcpy must not overlap, it causes other parts of the string to
be overwritten with adjacent characters and the result is
mangled. Patch replaces call to strcpy with memmove and adds a
test suite case for REPLACE. (closes issue ASTERISK-22910)
Reported by: Gareth Palmer Review:
https://reviewboard.asterisk.org/r/3083/ Patches:
func_strings.patch uploaded by Gareth Palmer (license 5169)
2013-12-31 21:25 +0000 [r404603] Kevin Harwell <kharwell@digium.com>
* cel/cel_pgsql.c: cel_pgsql: deadlock on unload and
core_event_dispatcher A deadlock can happen between a thread
unloading or reloading the cel_pgsql module and the
core_event_dispatcher taskprocessor thread. Description of what
is happening: Thread 1 (for example, a netconsole thread): a
"module reload cel_pgsql" is launched the thread enter the
"my_unload_module" function (cel_pgsql.c) the thread acquire the
write lock on psql_columns the thread enter the
"ast_event_unsubscribe" function (event.c) the thread try to
acquire the write lock on ast_event_subs[sub->type] Thread 2
(core_event_dispatcher taskprocessor thread): the taskprocessor
pop a CEL event the thread enter the "handle_event" function
(event.c) the thread acquire the read lock on
ast_event_subs[sub->type] the thread callback the "pgsql_log"
function (cel_pgsql.c), since it's a subscriber of CEL events the
thread try to acquire a read lock on psql_columns (closes issue
ASTERISK-22854) Reported by: Etienne Lessard Patches:
cel_pgsql_fix_deadlock_event.patch uploaded by hexanol (license
6394)
2013-12-20 21:12 +0000 [r404456] Scott Griepentrog <sgriepentrog@digium.com>
* main/say.c: say.c: correct time for polish In
ast_say_date_with_format_pl(), change ast_say_number() to use
tm_sec instead of tm_mn. (closes issue ASTERISK-22856) Reported
by: Robert Mordec Review:
https://reviewboard.asterisk.org/r/3082/ Patches: say.c.patch
uploaded by veilen (license 6555)
2013-12-18 19:47 +0000 [r404212] Richard Mudgett <rmudgett@digium.com>
* addons/ooh323c/src/ooq931.c, addons/ooh323c/src/memheap.c,
addons/ooh323c/src/ooTimer.c, addons/ooh323c/src/ooCapability.c,
addons/ooh323c/src/perutil.c, addons/ooh323cDriver.c,
addons/ooh323c/src/ooSocket.c: ooh323c: Fix gcc 4.6.3 compiler
warnings.
2013-12-18 11:58 +0000 [r404135] Joshua Colp <jcolp@digium.com>
* res/res_calendar.c: res_calendar: Protect channel when adding
datastore. This change adds a missing channel lock when adding a
datastore to a channel.
2013-12-18 00:27 +0000 [r404044-404081] Rusty Newton <rnewton@digium.com>
* funcs/func_strings.c: func_strings: Documentation fix for QUOTE()
Example output was inaccurate. (issue ASTERISK-22970) (closes
issue ASTERISK-22970) Reported by: Gareth Palmer Patches:
func_strings.patch uploaded by Gareth Palmer (license 5169)
* channels/chan_iax2.c, apps/app_chanspy.c, apps/app_mixmonitor.c,
include/asterisk/test.h, main/channel.c: Several components:
fixing Typos in comments and code, "avaliable" instead of
"available" (issue ASTERISK-23021) (closes issue ASTERISK-23021)
Reported by: Jeremy Lainé Tested by: Rusty Newton Patches:
available.patch uploaded by Jeremy Lainé (license 6561)
2013-12-16 16:36 +0000 [r403913] David M. Lee <dlee@digium.com>
* include/asterisk/pbx.h, main/asterisk.c, funcs/func_realtime.c,
main/pbx.c, main/tcptls.c, funcs/func_db.c,
README-SERIOUSLY.bestpractices.txt, configs/asterisk.conf.sample,
funcs/func_shell.c, funcs/func_env.c, funcs/func_lock.c,
UPGRADE.txt: security: Inhibit execution of privilege escalating
functions This patch allows individual dialplan functions to be
marked as 'dangerous', to inhibit their execution from external
sources. A 'dangerous' function is one which results in a
privilege escalation. For example, if one were to read the
channel variable SHELL(rm -rf /) Bad Things(TM) could happen;
even if the external source has only read permissions. Execution
from external sources may be enabled by setting
'live_dangerously' to 'yes' in the [options] section of
asterisk.conf. Although doing so is not recommended. (closes
issue ASTERISK-22905) Review:
http://reviewboard.digium.internal/r/432/
2013-12-16 15:53 +0000 [r403853-403862] Scott Griepentrog <sgriepentrog@digium.com>
* main/pbx.c: pbx.c: put copy of ast_exten.data on stack to prevent
memory corruption During dialplan execution in
pbx_extension_helper(), the contexts global read lock prevents
link list corruption, but was released with a pointer to the
ast_exten and data later used in variable substitution. Instead,
this patch removes pbx_substitute_variables() and locates a copy
of the ast_exten data on the stack before releasing the lock,
where ast_exten could get free'd by another thread performing a
module reload. (issue AST-1179) Reported by: Thomas Arimont
(issue AST-1246) Reported by: Alexander Hömig Review:
https://reviewboard.asterisk.org/r/3055/
* apps/app_sms.c: app_sms: BufferOverflow when receiving odd length
16 bit message This patch prevents an infinite loop overwriting
memory when a message is received into the unpacksms16()
function, where the length of the message is an odd number of
bytes. (closes issue ASTERISK-22590) Reported by: Jan Juergens
Tested by: Jan Juergens
2013-12-11 19:11 +0000 [r403634] Russell Bryant <russell@russellbryant.com>
* channels/chan_sip.c: Reset peer outboundproxy on sip.conf reload
If you set a peer's outboundproxy and then removed it from the
config, this would not get picked up in a config reload. This
patch fixes that by resetting it in set_peer_defaults(). Closes
ASTERISK-19454 Review: https://reviewboard.asterisk.org/r/3065/
2013-12-09 03:10 +0000 [r403449] Matthew Jordan <mjordan@digium.com>
* res/res_fax_spandsp.c: res_fax_spandsp: Always init T.38 session
to avoid crashes during state change Prior to this patch,
res_fax_spandsp was conservative with how it initialized the
spandsp T.38 context. It would only initialize it if the driver
thought the current state was a T.38 fax. While this works fine
in nominal situations, in certain off nominal situations,
res_fax_spandsp can believe that a T.38 fax will not occur when
in fact one has started. In particular, this was discovered when
res_fax would fall back to audio after timing out on a T.38
upgrade. The SIP channel driver would continue to retry the
re-INVITE and - if the remote end responded after res_fax timed
out with a 200 OK - a T.38 frame would be delivered to the
res_fax stack when it no longer expected it. As it turns out,
there does not appear to be any downside to always initializing
the T.38 context, other than the actual memory allocation. Since
that avoids this off nominal situation (and others which are
equally likely hard to predict), this is the safest way to avoid
this problem. Much thanks to Torrey as well for providing a
scenario that reproduces this issue. (closes issue
ASTERISK-21242) Reported by: Ashley Winters Tested by: Torrey
Searle patches: always-init-t38.patch uploaded by awinters
(License 6477) A_PARTY.xml uploaded by tsearle (License 5334)
2013-11-22 17:10 +0000 [r403014] Joshua Colp <jcolp@digium.com>
* main/translate.c: translate: Move freeing of frame to after it is
used. When translating from one format to another it is possible
to inform the translation function that the source frame should
be freed. This was previously done immediately but shortly
afterwards the frame that was freed was accessed and used again.
This change moves code around a bit so that the frame is now
freed after it has been completely used. (closes issue
ASTERISK-22788) Reported by: Corey Farrell Patches:
translate-access-after-free-11up.patch uploaded by coreyfarrell
(license 5909) translate-access-after-free-1.8.patch uploaded by
coreyfarrell (license 5909)
2013-11-12 14:55 +0000 [r402645-402708] Kinsey Moore <kmoore@digium.com>
* channels/chan_dahdi.c: chan_dahdi: Fix crash during caller ID
read Asterisk will sometimes core dump during caller id read on
analog channels due to a negative return value from the read() in
my_get_callerid that slips through as a negative length argument
to callerid_feed() if the errno returned by DAHDI is ELAST. This
change ensures that the negative return is treated properly even
when it is ELAST. (closes issue ASTERISK-22746) Reported by:
Michael Walton Patches: chan_dahdi_cid_crash_fix.r401410.patch
uploaded by Michael Walton (License 6502)
* apps/app_queue.c: app_queue: Honor penalty limits of 0 In the
current app_queue code from 1.8 up to trunk the upper and lower
penalties can be set to 0 but the value is interpreted to be
disabled instead of actually setting limits. This is especially
evident if min and max limits are set to 0 and members with
penalties of 0 and 1 are in the queue since the member with
penalty 1 will still receive calls. This patch adjusts the
special disabled value to be INT_MAX instead of 0. (closes issue
ASTERISK-20862) Review: https://reviewboard.asterisk.org/r/2995/
Reported by: Schmooze Com
2013-11-08 22:46 +0000 [r402604] Scott Griepentrog <sgriepentrog@digium.com>
* channels/sip/include/sip.h, channels/chan_sip.c: chan_sip: keep
same local (from) tag for outgoing register requests For outbound
register requests the tag on the From line was updated every 20
seconds prior to a successful registration and also once for each
registration renewal. That behavior can possibly cause the
registration to be denied because of the different tag, and is
not aligned with the intention of RFC 3261 8.1.3.5 "... request
constitutes a new transaction and SHOULD have the same value of
the Call-ID, To, and From of the previous request...". This
updates chan_sip to have a field to keep the local tag in the
registration structure and use that tag for registration requests
where the callid is also unchanged. (closes issue ASTERISK-12117)
Reported by: Pawel Pierscionek Review:
https://reviewboard.asterisk.org/r/2988/
2013-11-05 15:08 +0000 [r402468] Kevin Harwell <kharwell@digium.com>
* channels/chan_sip.c: chan_sip: notify dialog info ignores
presentation indicator in callerid The presentation indicator in
a callerid (e.g. set by dialplan function
Set(CALLERID(name-pres)= ...)) is not checked when SIP Dialog
Info Notifies are generated during extension monitoring. Added a
check to make sure the name and/or number presentations on the
callee (remote identity) are set to allow. If they are restricted
then "anonymous" is used instead. (closes issue AST-1175)
Reported by: Thomas Arimont Review:
https://reviewboard.asterisk.org/r/2976/
2013-10-31 15:57 +0000 [r402287] Matthew Jordan <mjordan@digium.com>
* main/loader.c: core/loader: Don't call dlclose in a while loop
For awhile now, we've noticed continuous integration builds
hanging on CentOS 6 64-bit build agents. After resolving a number
of problems with symbols, strange locks, and other shenanigans,
the problem has persisted. In all cases, gdb shows the Asterisk
process stuck in loader.c on one of the infinite while loops that
calls dlclose repeatedly until success. The documentation of
dlclose states that it returns 0 on success; any other value on
error. It does not state that repeatedly calling it will
eventually clear those errors. Most likely, the repeated calls to
dlclose was to force a close by exhausting the references on the
library; however, that will never succeed if: (a) There is some
fundamental error at work in the loaded library that precludes
unloading it (b) Some other loaded module is referencing a symbol
in the currently loaded module This results in Asterisk sitting
forever. Since we have matching pairs of dlopen/dlclose, this
patch opts to only call dlclose once, and log out as an ERROR if
dlclose fails to return success. If nothing else, this might help
to determine why on the CentOS 6 64-bit build agent things are
not closing successfully. Review:
https://reviewboard.asterisk.org/r/2970
2013-10-29 23:41 +0000 [r402224] Rusty Newton <rnewton@digium.com>
* sounds/Makefile: Updates for 1.4.25 core sounds and 1.4.14 extra
sounds, plus new en_GB language set The new sound packages relate
to issues: ASTERISK-22544, ASTERISK-22411, ASTERISK-21413,
ASTERISK-20782 Modified sounds/Makefile for the new sound
versions and to account for the new en_GB language set. (issue
ASTERISK-22659) (closes issue ASTERISK-22659) (closes issue
ASTERISK-22411) (closes issue ASTERISK-22544)
2013-10-29 14:52 +0000 [r402192] David M. Lee <dlee@digium.com>
* configure, configure.ac, makeopts.in, Makefile: Backport r373119
from 11 to go along with RAII_VAR support. In order to use nested
functions on some versions of GCC (e.g. GCC on OS X), the
-fnested-functions flag must be passed to the compiler. This
patch adds detection logic to ./configure to add the flag if
necessary.
2013-10-29 12:40 +0000 [r402150] Matthew Jordan <mjordan@digium.com>
* main/translate.c, main/xmldoc.c, main/channel.c, main/pbx.c:
Remove some spammy debug messages; improve clarity of others
Debug messages aren't free. Even when the debug level is
sufficiently low such that the messages are never evaluated,
there is a cost to having to parse Asterisk logs that contain
debug messages that (a) fail to convey sufficient information or
(b) occur so frequently as to be next to meaningless. Based on
having to stare at lots of DEBUG messages, this patch makes the
following changes: * channel.c: When copying variables from a
parent channel to a child channel, specify the channels involved.
Do not log anything for a variable that is not inherited; the
fact that it doesn't have an _ or __ already signifies that it
won't be inherited. * pbx.c: Specify what function evaluation has
occurred that created the result. * translate.c: Bump up the
translator path messages to 10. I've never once had to use these
debug messages, and for each format that is registered (on
startup) and unregistered (on shutdown) the entire f^2 matrix is
logged out. For short tests in the Asterisk Test Suite, this
should make finding the actual test much easier. * xmldoc.c: The
debug message that 'blah' is not found in the tree is expected.
Often, description elements - which are not required - are not
provided. This debug message adds no additional value, as it is
not indicative of an error or helpful in debugging which element
did not contain a 'blah' element as a child. If an element is
supposed to contain a child element, then that XML tree should
have failed validation in the first place. Review:
https://reviewboard.asterisk.org/r/2966/
2013-12-17 Asterisk Development Team <asteriskteam@digium.com>
* Asterisk 1.8.25.0 Released.
2013-12-16 Asterisk Development Team <asteriskteam@digium.com>
* Asterisk 1.8.25.0-rc2 Released.
* AST-2013-006 - app_sms: BufferOverflow when receiving odd length 16
bit message
This patch prevents an infinite loop overwriting memory when a
message is received into the unpacksms16() function, where the length
of the message is an odd number of bytes.
(closes issue ASTERISK-22590)
* AST-2013-007 - security: Inhibit execution of privilege escalating
functions
This patch allows individual dialplan functions to be marked as
'dangerous', to inhibit their execution from external sources.
A 'dangerous' function is one which results in a privilege
escalation. For example, if one were to read the channel variable
SHELL(rm -rf /) Bad Things(TM) could happen; even if the external
source has only read permissions.
Execution from external sources may be enabled by setting
'live_dangerously' to 'yes' in the [options] section of
asterisk.conf. Although doing so is not recommended.
(closes issue ASTERISK-22905)
2013-10-28 Asterisk Development Team <asteriskteam@digium.com>
* Asterisk 1.8.25.0-rc1 Released.
2013-10-25 21:51 +0000 [r401959-402000] Scott Griepentrog <sgriepentrog@digium.com>
* include/asterisk/rtp_engine.h, main/rtp_engine.c: rtp_engine: fix
rtp payloads copy and improve argument names In function
ast_rtp_instance_early _bridge_make_compatible the use of
instance 0/1 as arguments doesn't clearly communicate a direction
that the copying of payloads from the source channel to the
destination channel will occur, making it more probable to have
the arguments to ast_rtp_codecs_payloads_copy() put in the
reverse order. This patch renames the arguments with _dst and
_src suffixes and corrects the copy direction.
* include/asterisk/pbx.h, main/pbx.c: pbx.c: fix confused match
caller id that deleted exten still in hash This fixes a bug where
a zero length callerid match adjacent to a no match callerid
extension entry would be deleted together, which then resulted in
hashtable references to free'd memory. A third state of the
matchcid value has been added to indicate match to any extension
which allows enforcing comparison of matchcid on/off without
errors. (closes issue AST-1235) Reported by: Guenther Kelleter
Review: https://reviewboard.asterisk.org/r/2930/
2013-10-25 17:21 +0000 [r401619-401914] Jonathan Rose <jrose@digium.com>
* utils/clicompat.c: Put clicompat-r2.patch back in We've figured
out how to resolve the problems this was causing in 12/trunk, so
this can go back in now. (issue ASTERISK-22467) Reported by:
Corey Farrell Patches: clicompat-r2.patch uploaded by
coreyfarrell (license 5909)
* utils/clicompat.c: revert clicompat-r2.patch from r401704 Patch
caused the following build errors against testsuite
https://bamboo.asterisk.org/bamboo/browse/AST-ATRUNKBUILD4-244
(issue ASTERISK-22467) Reported by: Corey Farrell
* main/utils.c: utils: Fix memory leaks and missed unregistration
of CLI commands on shutdown Final set of patches in a series of
memory leak/cleanup patches by Corey Farrell (closes issue
ASTERISK-22467) Reported by: Corey Farrell Patches:
main-utils-1.8.patch uploaded by coreyfarrell (license 5909)
main-utils-11.patch uploaded by coreyfarrell (license 5909)
main-utils-12up.patch uploaded by coreyfarrell (license 5909)
* tests/test_linkedlists.c: test_linkedlists: Fix memory leak
(issue ASTERISK-22467) Reported by: Corey Farrell Patches:
test_linkedlists-1.8.patch uploaded by coreyfarrell (license
5909) test_linkedlists-11up.patch uploaded by coreyfarrell
(license 5909)
* main/jitterbuf.c: jitterbuf: Fix memory leak on jitter buffer
reset (issue ASTERISK-22467) Reported by: Corey Farrell Patches:
jitterbuf-jb_reset-leak-1.8.patch
jitterbuf-jb_reset-leak-11up.patch
* main/astobj2.c: astobj2: Unregister debug CLI commands at exit
(issue ASTERISK-22467) Reported by: Corey Farrell Patches:
astobj2-clean-debug-cli-1.8-11.patch uploaded by coreyfarrell
(license 5909) astobj2-clean-debug-cli-12up.patch uploaded by
coreyfarrell (license 5909)
* apps/app_voicemail.c: app_voicemail: Memory Leaks against tests
(issue ASTERISK-22467) Reported by: Corey Farrell Patches:
app_voicemail-1.8.patch uploaded by coreyfarrell (license 5909)
app_voicemail-11up.patch uploaded by coreyfarrell (license 5909)
* main/asterisk.c, utils/clicompat.c, channels/chan_dahdi.c,
codecs/ilbc/doCPLC.c, main/data.c, main/app.c: memory leaks:
Memory leak cleanup patch by Corey Farrell (second set) Also
covers ast_app_parse_timelen-fail-zero-length.patch, but the
patch was replaced with one of my own. (issue ASTERISK-22467)
Reported by: Corey Farrell Patches: chan_dahdi-cleanup_push.patch
uploaded by coreyfarrell (license 5909) clicompat-r2.patch
uploaded by coreyfarrell (license 5909) codecs-ilbc-doCPLC.patch
uploaded by coreyfarrell (license 5909)
data-cleanup-test-registration.patch uploaded by coreyfarrell
(license 5909) main-asterisk-kill-listener.patch uploaded by
coreyfarrell (license 5909)
* tests/test_dlinklists.c, funcs/func_math.c,
channels/sip/reqresp_parser.c, main/test.c,
main/editline/readline.c: memory leaks: Memory leak cleanup patch
by Corey Farrell (first set) (issue ASTERSIK-22467) Reported by:
Corey Farrell Patches:
chan_sip-parse_contact_header_test-free-contacts.patch uploaded
by coreyfarrell (license 5909) cli-filename-completion-leak.patch
uploaded by coreyfarrell (license 5909) func_math.patch uploaded
by corefarrell (license 5909) main-test-cleanup.patch uploaded by
coreyfarrell (license 5909) test_dlinklists.patch uploaded by
coreyfarrell (license 5909)
* main/translate.c, res/res_rtp_asterisk.c: res_rtp_asterisk:
Address jittery DTMF events in RTP streams (closes issue
ASTERISK-21170) Reported by: NITESH BANSAL Patches:
dtmf-timestamp.patch uploaded by NITESH BANSAL (license 6418)
Review: https://reviewboard.asterisk.org/r/2938/
2013-10-23 16:34 +0000 [r401577] Richard Mudgett <rmudgett@digium.com>
* cdr/cdr_adaptive_odbc.c: cdr_adaptive_odbc: Also apply a filter
when the CDR value is empty. Extra CDR records are written if a
filtered CDR value is empty because the filter is not checked.
(closes issue ASTERISK-22272) Reported by: Jordi Llull Chavarria
2013-10-23 15:19 +0000 [r401537] Kinsey Moore <kmoore@digium.com>
* channels/chan_mgcp.c: chan_mgcp: Properly handle malformed media
lines This corrects a situation in which a media line was not
parsed properly and resulted in a crash. (closes issue
ASTERISK-21190) Reported by: adomjan Patches:
chan_mgcp.c-sscnaf_fix uploaded by adomjan (License 5448)
2013-10-23 11:10 +0000 [r401497] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: Fix an issue where an incompatible audio
format may be added to SDP. If preferred codecs included any
non-audio format the code would mistakenly add the audio format,
even if it was not a joint capability with the remote side.
(closes issue ASTERISK-21131) Reported by: nbougues Patches:
patch_unsupported_codec_1.8.patch uploaded by nbougues (license
6470)
2013-10-22 22:36 +0000 [r401445] Matthew Jordan <mjordan@digium.com>
* res/res_rtp_asterisk.c: res_rtp_asterisk: Fix crash when RTCP is
not available during SSRC change In r400089, a patch was put in
to correct erroneous RTCP statistic resets. Unfortunately,
ast_rtp_read can be called on an RTP instance that does not have
RTCP information. This patch prevents that crash by only
resetting the statistics if we do actually have an RTCP instance.
(issue AST-1174) (closes issue ASTERISK-22667) Reported by: John
Bigelow
2013-10-22 00:13 +0000 [r401378] Richard Mudgett <rmudgett@digium.com>
* channels/sig_analog.c: chan_dahdi: Fix unable to get index
warning when transferring an analog call. Transferring an analog
call using flashhooks generated an unable to get index WARNING
message when the transfer is completed. * Removed unnecessary
analog subchannel shell games when transferring a call using
flashhooks. Thanks to Tzafrir Cohen for mentioning this in a
comment on issue ASTERISK-22720.
2013-10-21 19:45 +0000 [r401325] Kevin Harwell <kharwell@digium.com>
* main/editline/term.c: Segfault in LIBEDIT_INTERNAL after
tgetstr(), when libncurses5-dev isn't installed Include the
appropriate declarations when not using termcap, but term+curses
and [n]curses do not exist. (closes issue ASTERISK-22351)
Reported by: A. Iglesias Patches:
issueA22351_libedit_internal_without_ncurses_dev.patch uploaded
by wdoekes (license 5674)
2013-10-18 14:40 +0000 [r401178] Walter Doekes <walter+asterisk@wjd.nu>
* main/channel.c: Properly copy/remove the device state cache flag
over a masquerade. In r378303 the AST_FLAG_DISABLE_DEVSTATE_CACHE
flag was added that tells the devstate system to not cache states
for non-real devices. However, when optimizing away channels
(ast_do_masquerade), that flag wasn't copied. In my case, using
Local devices as queue members created a situation where the
endpoint was considered in use, but the state change of the
device being available again was ignored (not cached). The
endpoint channel was optimized into the (previously) Local
channel, but kept the do-not-cache flag. The end result being
that the queue member apparently stayed in use forever. (closes
issue ASTERISK-22718) Reported by: Walter Doekes Review:
https://reviewboard.asterisk.org/r/2925/
2013-10-17 15:22 +0000 [r401119] Kinsey Moore <kmoore@digium.com>
* res/res_jabber.c: Reduce log level of a non-pubsub error message
Drop an error log message to debug level 1 since distributed
device state functions correctly when receiving this message and
it spams the logs. (closes issue ASTERISK-22410) Reported by:
abelbeck Patches:
asterisk-1.8-res_jabber-log-nonpubsub-error-to-debug.patch
uploaded by abelbeck (License 5903)
2013-10-16 11:04 +0000 [r401049] Walter Doekes <walter+asterisk@wjd.nu>
* apps/app_queue.c: Don't check all realtime queues when doing
"queue show some_queue". When using realtime queues, queues have
to be fetched from the database every now and then to see if any
info has been changed or to see if the queue has been removed.
When fetching info for an individual queue, the pruning of other
queues is unnecessarily costly. Review:
https://reviewboard.asterisk.org/r/2907/
2013-10-15 14:52 +0000 [r400970] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Prevent chan_sip from sending duplicate
BYEs. When a 200 OK for an initial INVITE is received, we were
doing the right thing by ACKing and sending an immediate BYE.
However, we also were doing the wrong thing and queuing an answer
frame, thus causing the call to be answered. This would cause the
call to be hung up by the channel thread, thus resulting in a
second BYE being sent out. In this fix, I also have set the
hangupcause to be correct since the initial BYE being sent by
Asterisk had an unknown hangup cause. I have changed to using
"Bearer capabilty not available" since the call was hung up due
to an SDP offer/answer error. (closes issue ASTERISK-22621)
reported by Kinsey Moore
2013-10-14 21:40 +0000 [r400907] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c: chan_dahdi: Reflect the set software gain
in the CLI "dahdi show channel" output. * Remember the swgain
setting from CLI "dahdi set swgain" command so the CLI "dahdi
show channel" output will reflect the current setting. * Updated
CLI "dahdi set hwgain" and "dahdi set swgain" documentation.
(issue ASTERISK-22429) Reported by: Jaco Kroon Patches:
jira_asterisk_22429_v1.8_v2.patch (license #5621) patch uploaded
by rmudgett
2013-10-14 21:32 +0000 [r400906] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Do not increment the SDP version between 183
and 200 responses. Bumping the SDP version number can cause
interoperability problems since receivers of the responses will
expect that a 200 SDP will be identical to a previous 183 SDP.
(closes issue ASTERISK-21204) reported by NITESH BANSAL Patches:
dont-increment-session-version-in-2xx-after-183.patch uploaded by
NITESH BANSAL (License #6418)
2013-10-08 22:26 +0000 [r400694-400767] Kinsey Moore <kmoore@digium.com>
* configure, configure.ac: Add warning when compiling with iODBC
support When running configure, libiodbc2 development headers
will fulfill the requirement for ODBC development headers, but
will not function properly. This adds a warning when libiodbc2
development headers are detected instead of unixodbc development
headers. (closes issue ASTERISK-22459) Reported by: Patrick
Maille Tested by: Walter Doekes Patches:
issueA22459_warn_when_using_iodbc.patch uploaded by Walter Doekes
(License 5674)
* funcs/func_config.c: Fix func_config list entry allocation The
AST_CONFIG dialplan function defined in func_config.c allocates
its config file list entries using ast_malloc. List entry
allocations destined for use with Asterisk's linked list API must
be ast_calloc()d or otherwise initialized so that list pointers
are set to NULL. These uses of ast_malloc have been replaced by
ast_calloc to prevent dereferencing of uninitialized pointer
values when traversing the list. (closes issue ASTERISK-22483)
Reported by: Brian Scott
2013-10-06 17:07 +0000 [r400622] Michael L. Young <elgueromexicano@gmail.com>
* apps/app_queue.c: Fix Regression With Queuelog EXITWITHKEY Only
Logging Two Out Of Four Fields Commit r62462 added two extra
fields for logging "the original position the caller entered the
queue at, and the amount of time the caller was waiting in the
queue." But when r75969 was merged from 1.4 into trunk (r75977),
these two fields disappeared. Those two extra fields were not
logged in 1.4 and when the patch was merged, those fields went
away. Therefore, this is a regression and was caught by the
reporter because he was reading the awesome "Asterisk: The
Definitive Guide" book. (closes issue ASTERISK-22197) Reported
by: Dalius M. Tested by: Dalius M. Patches:
asterisk-22197-q-log-exitwithkey.diff uploaded by Michael L.
Young (license 5026) Review:
https://reviewboard.asterisk.org/r/2901/
2013-10-03 22:51 +0000 [r400469] Jonathan Rose <jrose@digium.com>
* channels/chan_sip.c: chan_sip: Don't ignore expires value in
contact header if it lacks semicolon (closes issue
ASTERISK-22574) Reported by: Filip Jenicek Patches:
chan_sip_expires.patch uploaded by Filip Jenicek (license 6277)
2013-10-03 18:25 +0000 [r400393] Kinsey Moore <kmoore@digium.com>
* res/res_rtp_multicast.c: Ensure res_rtp_mutlicast sets SSRC
properly This fixes a bug where the SSRC field on multicast RTP
can be stuck at 0 which can cause problems for endpoints trying
to make sense of incoming streams. (closes issue ASTERISK-22567)
Reported by: Simone Camporeale Patches:
22567_res_mulitcast_ssrc.patch uploaded by Simone Camporeale
(License 6536)
2013-10-02 21:30 +0000 [r400314] Michael L. Young <elgueromexicano@gmail.com>
* channels/chan_iax2.c: Cast Integer Argument To Unsigned Char The
member reg in the peercnt structure is an unsigned char and
peercnt_modify() is expecting an unsigned char argument which
gets assigned to peercnt->reg. This patch fixes that by casting
the integer argument being passed to peercnt_modify to unsigned
char.
2013-09-30 15:19 +0000 [r400137] Kinsey Moore <kmoore@digium.com>
* channels/chan_sip.c, UPGRADE.txt, configs/sip.conf.sample: Allow
Asterisk to retry after 403 on register This adds a global option
in chan_sip to allow it to continue attempting registration if a
403 is received, clearing the cached nonce and treating it as a
non-fatal response. Normally, this would cause registration
attempts to that endpoint to stop. (closes issue ASTERISK-17138)
Review: https://reviewboard.asterisk.org/r/2874/ Reported by:
Rudi
2013-09-28 22:20 +0000 [r400073-400089] Matthew Jordan <mjordan@digium.com>
* res/res_rtp_asterisk.c: res_rtp_asterisk: Correct erroneous lost
packet information in RTCP reports RTCP's calculation of the
number of lost packets in an RTP stream is based on that stream's
sequence number count, the number of received packets, and how
many packets we expect to receive. When the SSRC for an RTP
stream changes, there can - and almost always will be - a large
jump in the next packet's timestamp and sequence number. If we
don't reset the number of received packets, sequence number
count, and other metrics used by RTCP, the next RR/SR report will
use the previous SSRC's values to calculate the lost packet count
for the new SSRC - resulting in a very large number of lost
packets. This patch modifies res_rtp_asterisk such that, if it
detects a SSRC change, it will reset the various values used by
the RTCP calculations. From the perspective of RTCP, this appears
as a new media stream - which is what it is. Review:
https://reviewboard.asterisk.org/r/2886/ (closes issue AST-1174)
Reported by: Thomas Arimont
* configure.ac, configure: Add check for openSUSE when detecting
bfd library In ASTERISK-17842, some additional library checks
were added to the configure script so that the bfd library could
be found on CentOS and Fedora systems. As it turns out, openSUSE
requires an additional library. This patch adds another check to
the configure script for openSUSE that will add that library.
Review: https://reviewboard.asterisk.org/r/2885/ (closes issue
AST-1169) Reported by: Guenther Kelleter
2013-09-27 21:31 +0000 [r400013] Richard Mudgett <rmudgett@digium.com>
* channels/chan_sip.c, channels/sip/reqresp_parser.c: chan_sip:
Increase some scratch buffer sizes dealing with caller id. *
Eliminated an unnecessary initialization in check_user_full().
(closes issue ASTERISK-22477) Reported by: Michael Shepelev
2013-09-27 17:13 +0000 [r399939] Jonathan Rose <jrose@digium.com>
* channels/sip/include/sip.h, channels/chan_sip.c: chan_sip: Reject
calls on 200 OKs if no SDP has been received When Asterisk
receives a 200 OK in response to an invite, that peer should have
sent an SDP at some point by then. If the channel has never
received an SDP, media won't have been set and the remote address
won't be known. Endpoints in general should not be doing this.
This patch makes it so that Asterisk will simply hang up a call
if it sends a 200 OK at this point. So far this odd behavior for
endpoints has only been observed in tests which involved manually
created SIP transactions in SIPp. (closes issue ASTERISK-22424)
Reported by: Jonathan Rose Review:
https://reviewboard.asterisk.org/r/2827/
2013-09-25 20:23 +0000 [r399818] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c, channels/sig_ss7.c: chan_dahdi: CLI "core
stop gracefully" has needless delay for PRI and SS7. The PRI and
SS7 link control threads are not stopped correctly when the
chan_dahdi.so module is unloaded. The link control threads
pri_dchannel() and ss7_linkset() are not awakened from a poll()
to cancel the thread. * Added a SIGURG signal after requesting
the thread cancel to break the link control thread poll()
immediately. For SS7 it was slightly worse, the link poll()
timeout would always be whatever was the last libss7 scheduled
event time used. If no libss7 scheduled event was pending, the
thread could run more often than necessary. * Set nextms to 60
seconds for the ss7_linkset() poll() if there is no other libss7
scheduled event.
2013-09-25 19:25 +0000 [r399794] Michael L. Young <elgueromexicano@gmail.com>
* channels/chan_sip.c: Fix Realtime Peer Update Problem When
Un-registering And Expires Header In 200ok 1st Issue When a
realtime peer sends an un-REGISTER request, Asterisk un-registers
the peer but the database table record still has regseconds and
fullcontact for the peer. This results in calls attempting to be
routed to the peer which is no longer registered. The expected
behavior is to get busy/congested when attempting to call an
un-registered peer through the dialplan. What was discovered is
that we are clearing out the peer's registration in the database
in parse_register_contact() when calling expire_register() but
then upon returning from parse_register_contact(), update_peer()
is run which stores back in the database table regseconds and
fullcontact. 2nd Issue The reporter pointed out that the 200 ok
being returned by Asterisk after un-registering a peer contains a
Contact header with ;expires= and the Expires header is not set
to 0. This is actually a regression. Tests were created for this
second issue (ASTERISK-22548). The tests have been reviewed and a
Ship It! was received on those tests. This patch does the
following: * Do not ignore the Expires header value even when it
is set to 0. The patch sets the pvt->expiry earlier on in the
function so that it is set properly and used. * If pvt->expiry is
0, do not call update_peer since that means the peer has already
been un-registered and there is no need to update the database
record again since nothing has changed. (closes issue
ASTERISK-22428) Reported by: Ben Smithurst Tested by: Ben
Smithurst, Michael L. Young Patches:
asterisk-22428-rt-peer-update-and-expires-header.diff by Michael
L. Young (license 5026) Review:
https://reviewboard.asterisk.org/r/2869/
2013-09-24 20:03 +0000 [r399697] Richard Mudgett <rmudgett@digium.com>
* channels/chan_iax2.c: chan_iax2: Prevent some needless breaking
of the native IAX2 bridge. * Clean up some twisted code in the
iax2_bridge() loop. * Add AST_CONTROL_VIDUPDATE and
AST_CONTROL_SRCCHANGE to a list of frames to prevent the native
bridge loop from breaking. * Passing the
AST_CONTROL_T38_PARAMETERS frame should also allow FAX over a
native IAX2 bridge. (issue ABE-2912) Review:
https://reviewboard.asterisk.org/r/2870/
2013-09-19 16:34 +0000 [r399456] Jonathan Rose <jrose@digium.com>
* channels/chan_sip.c: chan_sip: Make direct media reinvites for
T38 put Asterisk in the media path Prior to this patch, Asterisk
would incorrectly use the previous endpoint addresses in SDP in
spite of providing its own port. T38 is never meant to be done
through directmedia and Asterisk should always be in the media
path for these streams. (closes issue ASTERISK-17273) Reported
by: Kevin Stewart (closes issue ASTERISK-18706) Reported by:
Jeremy Kister Review: https://reviewboard.asterisk.org/r/2853/
2013-10-21 Asterisk Development Team <asteriskteam@digium.com>
* Asterisk 1.8.24.0 Released.
2013-10-18 Asterisk Development Team <asteriskteam@digium.com>
* Asterisk 1.8.24.0-rc2 Released.
* Properly copy/remove the device state cache flag over a masquerade.
In r378303 the AST_FLAG_DISABLE_DEVSTATE_CACHE flag was added that
tells the devstate system to not cache states for non-real devices.
However, when optimizing away channels (ast_do_masquerade), that flag
wasn't copied.
In my case, using Local devices as queue members created a situation
where the endpoint was considered in use, but the state change of the
device being available again was ignored (not cached). The endpoint
channel was optimized into the (previously) Local channel, but kept
the do-not-cache flag. The end result being that the queue member
apparently stayed in use forever.
2013-09-19 Asterisk Development Team <asteriskteam@digium.com>
* Asterisk 1.8.24.0-rc1 Released.
2013-09-18 19:54 +0000 [r399402] Kinsey Moore <kmoore@digium.com>
* main/abstract_jb.c: Fix jitter buffer log file creation This
adjusts '/'-to-'#' replacement to replace all instances of '/'
instead of just the first to ensure that the jitter buffer log
file gets the correct name as per Richard Kenner's suggestion.
(closes issue ASTERISK-21036) Reported by: Richard Kenner
2013-09-18 17:15 +0000 [r399351] Matthew Jordan <mjordan@digium.com>
* build_tools/prep_tarball: Update prep_tarball with new
documentation files on the Asterisk wiki This will now pull both
a command reference for the version being prepared, as well as an
Admin Guide that applies to all versions of Asterisk. (issue
ASTERISK-22439) Reported by: Olle Johansson
2013-09-18 01:32 +0000 [r399304] Michael L. Young <elgueromexicano@gmail.com>
* main/features.c: Fix Segfault When Syntax Of A Line Under
[applicationmap] Is Invalid When processing the lines under the
[applicationmap] context in features.conf, a segfault occurs from
attempting to process a line with an invalid syntax (basically
missing most of the arguments). Example: [applicationmap]
automon=*6 * This patch moves the checking for empty arguments to
before they are accessed. * Also, checked the "todo" comment and
removed it. Some applications do not require arguments. (closes
issue ASTERISK-22416) Reported by: CGI.NET Tested by: CGI.NET
Patches: asterisk-22416-check-syntax-first_v2.diff by Michael L.
Young (license 5026) Review:
https://reviewboard.asterisk.org/r/2803
2013-09-16 16:37 +0000 [r399158] Richard Mudgett <rmudgett@digium.com>
* channels/chan_iax2.c: chan_iax2: Fix saving the wrong expiry time
in astdb. When a new IAX2 client registers, the astdb database is
updated with the value of minregexpire defined in iax.conf
instead of using the expiry time that is provided by the client.
The provided expiry time of the client is updated after inserting
the astdb entry. As a consequence, restarting or reloading
asterisk creates clients whose registration may expire before
they reregister. The clients are therefore unavailable after
minregexpire seconds until they reregister. * Move updating of
the expiry time to before inserting into the astdb. (closes issue
ASTERISK-22504) Reported by: Stefan Wachtler Patches:
chan_iax2.c.patch (license #6533) patch uploaded by Stefan
Wachtler
2013-09-13 20:47 +0000 [r399098] David M. Lee <dlee@digium.com>
* main/astobj2.c: Don't write to /tmp/refs when REF_DEBUG is not
defined. If MALLOC_DEBUG is enabled, then the debug destructor
for the container is used, which would erroneously write to
/tmp/refs. This patch only uses the debug destructor if ref_debug
is used. (closes issue ASTERISK-22536)
2013-09-13 13:31 +0000 [r399033] Kinsey Moore <kmoore@digium.com>
* apps/app_meetme.c: Fix several crashes in MeetMeAdmin This change
ensures that MeetMeAdmin commands requiring a user actually get a
user and fixes another issue where an extra dereference could
occur for a last-entered user being ejected if a user identifier
was also provided. (closes issue ASTERISK-21907) Reported by:
Alex Epshteyn Review: https://reviewboard.asterisk.org/r/2844/
2013-09-12 20:09 +0000 [r398937-398977] Jonathan Rose <jrose@digium.com>
* channels/sip/include/sip.h, channels/chan_sip.c: chan_sip: Revert
r398835 due to failing tests involving originate (issue
ASTERISK-22424) Reported by: Jonathan Rose
* res/res_musiconhold.c: res_musiconhold: Fix reference leaks
caused when reloading with REF_DEBUG set Due to a faulty function
for debugging reference decrementing, it was possible to reduce
the refcount on the wrong object if two moh classes of the same
name were in the moh class container. (closes issue
ASTERISK-22252) Reported by: Walter Doekes Patches:
18_moh_debug_ref_patch.diff Uploaded by Jonathan Rose (license
6182)
2013-09-12 00:00 +0000 [r398880-398884] Rusty Newton <rnewton@digium.com>
* apps/app_queue.c: 'queue add member' help text correction You are
adding dial strings to the queue, not channels. An aribitrary
string could be used, but you are typically referencing a
channel. Correcting the command help text. (issue ASTERISK-22263)
(closes issue ASTERISK-22263) Reported By: Rusty Newton
* configs/chan_dahdi.conf.sample: Documentation fix -
waitfordialtone is not boolean, it's time in milliseconds
Changing text in chan_dahdi.conf sample to be accurate. (issue
ASTERISK-22308) (closes issue ASTERISK-22308) Reported By:
Malcolm Davenport
2013-09-11 19:39 +0000 [r398835] Jonathan Rose <jrose@digium.com>
* channels/sip/include/sip.h, channels/chan_sip.c: chan_sip: Reject
calls without prior SDP on 200 OK If we receive a 200 OK without
SDP, we will now check to see if the remote address has been
established for that channel's RTP session and if the to tag for
that channel has changed from the most recent to tag in a
response less than 200. If either a change has been made since
the last to-tag was received or the remote address is unset, then
we will drop the call. (closes issue ASTERISK-22424) Reported by:
Jonathan Rose Review:
https://reviewboard.asterisk.org/r/2827/diff/#index_header
2013-09-10 17:53 +0000 [r398757] Richard Mudgett <rmudgett@digium.com>
* main/xmldoc.c, main/cli.c, funcs/func_dialgroup.c, main/heap.c,
main/event.c, res/res_musiconhold.c, main/indications.c,
main/asterisk.c: Fix incorrect usages of ast_realloc(). There are
several locations in the code base where this is done: buf =
ast_realloc(buf, new_size); This is going to leak the original
buf contents if the realloc fails. Review:
https://reviewboard.asterisk.org/r/2832/
2013-09-10 17:47 +0000 [r398748-398752] David M. Lee <dlee@digium.com>
* utils/check_expr.c: Fixed utils directory breakage from r398748,
this time with extra hate.
* utils/check_expr.c, utils/ael_main.c, utils/conf2ael.c: Fixed
utils directory breakage from r398648
2013-09-09 23:15 +0000 [r398703] Richard Mudgett <rmudgett@digium.com>
* main/astmm.c: MALLOC_DEBUG: Change fence magic number to be
completely different from the freed magic number. Race conditions
between freeing a nul terminated string and ast_strdup()'ing it
are more likely to be detected if the fence and freed magic
numbers are completely different.
2013-09-09 19:56 +0000 [r398648] David M. Lee <dlee@digium.com>
* main/utils.c, include/asterisk/lock.h, main/lock.c: Fix
DEBUG_THREADS when lock is acquired in __constructor__ This patch
fixes some long-standing bugs in debug threads that were
exacerbated with recent Optional API work in Asterisk 12. With
debug threads enabled, on some systems, there's a lock ordering
problem between our mutex and glibc's mutex protecting its module
list (Ubuntu Lucid, glibc 2.11.1 in this instance). In one
thread, the module list will be locked before acquiring our
mutex. In another thread, our mutex will be locked before locking
the module list (which happens in the depths of calling
backtrace()). This patch fixes this issue by moving backtrace()
calls outside of critical sections that have the mutex acquired.
The bigger change was to reentrancy tracking for
ast_cond_{timed,}wait, which wrongly assumed that waiting on the
mutex was equivalent to a single unlock (it actually suspends all
recursive locks on the mutex). (closes issue ASTERISK-22455)
Review: https://reviewboard.asterisk.org/r/2824/
2013-09-06 20:56 +0000 [r398523-398576] Kinsey Moore <kmoore@digium.com>
* res/res_jabber.c: Commit the remainder of r398523 This is a
missing part of the commit in revision 398523 that corrects the
name of a variable. (issue ASTERISK-22435)
* res/res_jabber.c: Fix Jabber/XMPP distributed MWI The mailbox and
context are swapped on the receiving end for all users of Jabber
and XMPP distributed MWI in Asterisk 1.8 and all more recent
versions. This swaps those values to be correct when publishing
to the internal event system from Jabber/XMPP distributed MWI
state. (closes issue ASTERISK-22435) Reported by: abelbeck Tested
by: Michael Keuter Patches:
asterisk-1.8-res_jabber-aji_handle_pubsub_event.patch uploaded by
abelbeck asterisk-11-res_xmpp-xmpp_pubsub_handle_event.patch
uploaded by abelbeck
2013-09-05 19:00 +0000 [r398235-398456] Richard Mudgett <rmudgett@digium.com>
* channels/chan_iax2.c: chan_iax2: Reduce indentation in
__attempt_transmit(). * Reduce indentation in
__attempt_transmit(). * Don't update the static last error time
variable every time in __schedule_action() and socket_read().
* channels/chan_iax2.c: chan_iax2: Fix stray reference to worker
thread idle_list. * Fix stray reference to idle_list in
cleanup_thread_list(). This may be the reason for the note in
iax2_process_thread() about threads not being removed from the
task lists. * Move cleanup_thread_list(&idle_list) to after the
other lists are cleaned up.
* channels/chan_iax2.c: chan_iax2: Fix bridgecallno deadlock
avoidance. * Fix bridgecallno deadlock avoidance. When doing
deadlock avoidance, you need to retest the status of values for
each loop to see if you still need the lock for bridgecallno. *
As a safety check, after acquiring the bridgecallno lock you
should check if iaxs[bridgecallno] is NULL just like the current
callno checks. * Move setting thread->iostate to IAX_IOSTATE_IDLE
to after processing any deferred frames to ensure that the
iostate is IDLE when it is placed back into the idle list.
defer_full_frame() tries to ensure iax2_process_thread() wakes up
to process the frame.
* channels/iax2-parser.c: chan_iax2: Add missing control frame
names to debug frame decode output. (Part 2)
* channels/iax2-parser.c: chan_iax2: Add missing control frame
names to debug frame decode output.
* channels/chan_misdn.c: chan_misdn: Fix misdn debug output printed
with arbitrary verbose levels. Fix the misdn debug output to
remote consoles. chan_misdn uses ast_console_puts() which doesn't
know about verbose levels. Better to use ast_verbose() instead.
Without this patch the misdn debug messages are appended to the
verbose level which ever was set by the message sent to the
console before, i.e. any undefined level. (closes issue AST-1218)
Reported by: Guenther Kelleter Patches: misdnlog.patch (license
#6372) patch uploaded by Guenther Kelleter
2013-09-02 07:24 +0000 [r398167] Walter Doekes <walter+asterisk@wjd.nu>
* cel/cel_custom.c: Be a little more verbose when loading
cel_custom.conf. Review: https://reviewboard.asterisk.org/r/2805/
2013-08-30 18:55 +0000 [r398021-398102] Kevin Harwell <kharwell@digium.com>
* channels/chan_sip.c, main/config.c, res/res_security_log.c: Fix
various memory leaks main/config.c - cleanup cache fie includes
res/res_security_log.c - unregister logger level
channesl/chan_sip.c - cleanup io context and notify_types (closes
issues ASTERISK-22378) Reported by: Corey Farrell Patches:
config_shutdown.patch uploaded by coreyfarrell (license 5909)
res_security_log.patch uploaded by coreyfarrell (license 5909)
chan_sip-1.8.patch uploaded by coreyfarrell (license 5909)
* main/manager.c, res/res_agi.c: Memory leak fix
ast_xmldoc_printable returns an allocated block that must be
freed by the caller. Fixed manager.c and res_agi.c to stop
leaking these results. (closes issue ASTERISK-22395) Reported by:
Corey Farrell Patches: manager-leaks-1.8.patch uploaded by
coreyfarrell (license 5909) res_agi-xmldoc-leaks.patch uploaded
by coreyfarrell (license 5909)
* main/features.c: Fix memory leak Fixed a features.c test that
leaked a reference to a parked call. This caused chancount to
never reach 0, so graceful shutdown stops. Also added an
unregister test. (closes issue ASTERISK-22413) Reported by: Corey
Farrell Patches: features-TEST_FRAMEWORK.patch uploaded by
coreyfarrell (license 5909)
2013-08-30 16:46 +0000 [r398018] Richard Mudgett <rmudgett@digium.com>
* tests/test_substitution.c: test_substituition: Fix failed test
reporting to actually report failure. You cannot put the "Testing
<blah> pass/fail" on a single line before actually performing the
test. Now any additional failure information is logged before the
test pass/fail announcement. * Added an additional CDR(answer,u)
test.
2013-08-27 17:55 +0000 [r397710-397756] Matthew Jordan <mjordan@digium.com>
* channels/chan_sip.c: AST-2013-005: Fix crash caused by invalid
SDP If the SIP channel driver processes an invalid SDP that
defines media descriptions before connection information, it may
attempt to reference the socket address information even though
that information has not yet been set. This will cause a crash.
This patch adds checks when handling the various media
descriptions that ensures the media descriptions are handled only
if we have connection information suitable for that media. Thanks
to Walter Doekes, OSSO B.V., for reporting, testing, and
providing the solution to this problem. (closes issue
ASTERISK-22007) Reported by: wdoekes Tested by: wdoekes patches:
issueA22007_sdp_without_c_death.patch uploaded by wdoekes
(License 5674)
* channels/chan_sip.c: AST-2013-004: Fix crash when handling ACK on
dialog that has no channel A remote exploitable crash
vulnerability exists in the SIP channel driver if an ACK with SDP
is received after the channel has been terminated. The handling
code incorrectly assumed that the channel would always be
present. This patch adds a check such that the SDP will only be
parsed and applied if Asterisk has a channel present that is
associated with the dialog. Note that the patch being applied was
modified only slightly from the patch provided by Walter Doekes
of OSSO B.V. (closes issue ASTERISK-21064) Reported by: Colin
Cuthbertson Tested by: wdoekes, Colin Cutherbertson patches:
issueA21064_fix.patch uploaded by wdoekes (License 5674)
2013-08-23 15:34 +0000 [r397525] Richard Mudgett <rmudgett@digium.com>
* include/asterisk/astmm.h, main/logger.c, main/utils.c,
include/asterisk/lock.h, main/astmm.c, channels/sig_pri.c,
main/astobj2.c, include/asterisk/logger.h, main/lock.c,
include/asterisk/utils.h: Fix memory corruption when trying to
get "core show locks". Review
https://reviewboard.asterisk.org/r/2580/ tried to fix the
mismatch in memory pools but had a math error determining the
buffer size and didn't address other similar memory pool
mismatches. * Effectively reverted the previous patch to go in
the same direction as trunk for the returned memory pool of
ast_bt_get_symbols(). * Fixed memory leak in ast_bt_get_symbols()
when BETTER_BACKTRACES is defined. * Fixed some formatting in
ast_bt_get_symbols(). * Fixed sig_pri.c freeing memory allocated
by libpri when MALLOC_DEBUG is enabled. * Fixed
__dump_backtrace() freeing memory from ast_bt_get_symbols() when
MALLOC_DEBUG is enabled. * Moved __dump_backtrace() because of
compile issues with the utils directory. (closes issue
ASTERISK-22221) Reported by: Matt Jordan Review:
https://reviewboard.asterisk.org/r/2778/
2013-08-22 08:19 +0000 [r397377] Walter Doekes <walter+asterisk@wjd.nu>
* default.exports, main/asterisk.exports.in: Add _IO_stdin_used in
version-script to fix SIGBUSes on Sparc. The
--version-script,asterisk.exports linker flag (and the module
exports) didn't provide _IO_stdin_used in the list of exported
symbols. That causes some kind of libc compatibility mode to kick
in, where stdio file structures (stdout/stderr) land somewhere
else. In the case of the Sparc, they landed on misaligned memory.
This became apparent first after r376428 (Reorder startup
sequence) when a lot of ast_log's were replaced with fprintf's.
Writing to stderr triggered a SIGBUS. (Compared to x86 and amd64
architectures, the Sparc is very picky about memory alignment.)
(issue ASTERISK-21763) (issue ASTERISK-21665) Reported by: Jeremy
Kister Review: https://reviewboard.asterisk.org/r/2760/
2013-08-21 17:00 +0000 [r397308] David M. Lee <dlee@digium.com>
* main/http.c: Complete http_shutdown. This patch frees up some
resources allocated in http.c. * tcp listeners stopped * tls
settings freed * uri redirects freed * unregister internal http.c
uri's (closes issue ASTERISK-22237) Reported by: Corey Farrell
Patches: http.patch uploaded by Corey Farrell (license 5909)
2013-08-21 14:56 +0000 [r397256] Matthew Jordan <mjordan@digium.com>
* include/asterisk/frame.h: Set 14400 as the default max bit rate
if T38MaxBitRate is not specified If an endpoint fails to include
the T38MaxBitRate attribute during negotiation, Asterisk will
negotiate a bit rate of 2400 instead of the ITU recommended bit
rate of 14400. This patch fixes this by making AST_T38_RATE_14400
the 'default' value of the enum by assigning it a value of 0,
such that if an endpoint fails to include the attribute, the
default will be 14400. Note that Walter Doekes included the nice
comment in frame.h about why we are purposefully assigning
AST_T38_RATE_14400 a value of 0. (closes issue ASTERISK-22275)
Reported by: Andreas Steinmetz patches: fax-fix.patch uploaded by
anstein (License 6523)
2013-08-21 02:09 +0000 [r397204] Michael L. Young <elgueromexicano@gmail.com>
* channels/chan_sip.c: Fix Not Storing Current Incoming Recv
Address In 1.8, r384779 introduced a regression by retrieving an
old dialog and keeping the old recv address since recv was
already set. This has caused a problem when a proxy is involved
since responses to incoming requests from the proxy server, after
an outbound call is established, are never sent to the correct
recv address. In 11, r382322 introduced this regression. The fix
is to revert that change and always store the recv address on
incoming requests. Thank you Walter Doekes for helping to point
out this error and Mark Michelson for your input/review of the
fix. (closes issue ASTERISK-22071) Reported by: Alex Zarubin
Tested by: Alex Zarubin, Karsten Wemheuer Patches:
asterisk-22071-store-recvd-address.diff by Michael L. Young
(license 5026)
2013-08-20 17:40 +0000 [r397112-397156] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Remove REF_DEBUG definition.
* channels/sip/dialplan_functions.c, channels/chan_sip.c: Fix
refcounting of sip_pvt in test_sip_rtpqos test and unlink it from
the list of pvts. (closes issue ASTERISK-22248) reported by Corey
Farrell patches: test_sip_rtpqos.patch uploaded by Corey Farrell
(license #5909)
2013-08-20 15:26 +0000 [r397033-397106] Kinsey Moore <kmoore@digium.com>
* main/threadstorage.c, main/astfd.c: Unregister CLI commands on
exit This patch ensures that CLI commands enabled by
DEBUG_FD_LEAKS and DEBUG_THREADLOCALS are cleaned up properly on
exit. (closes issue ASTERISK-22238) Reported by: Corey Farrell
Tested by: Corey Farrell Patches: debug_cli_unregister.patch
uploaded by Corey Farrell
* main/xmldoc.c: Fix xmldoc memory leak This fixes a
single-attribute memory leak that was occurring when the
"required" attribute was not true. (closes issue ASTERISK-22249)
Reported by: Corey Farrell Tested by: Corey Farrell Patches:
xmldoc-free_attr_required.patch uploaded by Corey Farrell
* main/cel.c: Protect CEL from an invalid config on reload This
patch fixes CEL to properly handle an invalid config on reload.
(closes issue ASTERISK-22259) Reported by: Corey Farrell Tested
by: Corey Farrell Patches: cel-config.patch uploaded by Corey
Farrell
2013-08-20 11:46 +0000 [r396994] Walter Doekes <walter+asterisk@wjd.nu>
* configs/h323.conf.sample, configs/sip.conf.sample: Add
"autoframing" option to sip.conf.sample and h323.conf.sample. The
autoframing option was added to chan_sip.c in r43243 (mogorman,
2006-09-19 01:32:57), but never made its way into the sample
configs. Review: https://reviewboard.asterisk.org/r/2768/
2013-08-20 01:17 +0000 [r396958] Matthew Jordan <mjordan@digium.com>
* main/data.c: Fix invalid access to disposed memory in main/data
unit test It is not safe to iterate over a macro'd list of ao2
objects, deref them such that the item's destructor is called,
and leave them in the list. The list macro to iterate over items
requires the item to be a valid allocated object in order to
proceed to the next item; with MALLOC_DEBUG on the corruption of
the linked list is caught in the crash. This patch fixes the
invalid access to free'd memory by removing the ao2 item from the
list before de-refing it. Note that this is a backport of r396915
from Asterisk trunk.
2013-08-15 16:21 +0000 [r396745] Kinsey Moore <kmoore@digium.com>
* main/cli.c, main/asterisk.c: Remove leading spaces from the CLI
command before parsing If you've mistakenly put a space before
typing in a command, the leading space will be included as part
of the command, and the command parser will not find the
corresponding command. This patch rectifies that situation by
stripping the leading spaces on commands. Review:
https://reviewboard.asterisk.org/r/2709/ Patch-by: Tilghman
Lesher
2013-08-14 19:05 +0000 [r396619-396656] Joshua Colp <jcolp@digium.com>
* tests/test_hashtab_thrash.c: Tweak comment for why usleep is
used.
* tests/test_hashtab_thrash.c: Tweak test_hashtab_thrash test to
allow the critical threads to execute. Depending on certain
conditions it was possible for the hashtab counting thread to
starve other threads, preventing them from executing in the
expected fashion. This change adds a sleep to allow the others to
do what they need to do. While this doesn't thrash the hashtab as
much as previously, it at least works. (closes issue
ASTERISK-22276) Reported by: Matt Jordan
2013-08-13 18:44 +0000 [r396579-396582] Walter Doekes <walter+asterisk@wjd.nu>
* channels/chan_sip.c: chan_sip: Convert 'just did sched_add
waitid...' from warning to debug message. Patches:
reviewboard-2377.patch uploaded by Paul Belanger Review:
https://reviewboard.asterisk.org/r/2377/
* channels/chan_sip.c: chan_sip: Fix IP-addr in warning when
rejecting a contact ACL. Patches: reviewboard-2155.patch uploaded
by Paul Belanger Review: https://reviewboard.asterisk.org/r/2155/
2013-08-08 20:14 +0000 [r396427] Walter Doekes <walter+asterisk@wjd.nu>
* main/logger.c, main/utils.c, main/astobj2.c,
include/asterisk/logger.h: Consistent memory allocation by
ast_bt_get_symbols. Always use ast_alloc/ast_free. This is
handled differently in trunk (r391012). Review:
https://reviewboard.asterisk.org/r/2580/
2013-08-06 08:14 +0000 [r396279] Walter Doekes <walter+asterisk@wjd.nu>
* pbx/pbx_dundi.c, utils/extconf.c, apps/app_stack.c,
apps/app_playback.c, funcs/func_global.c, main/cdr.c,
pbx/pbx_loopback.c, main/pbx.c, funcs/func_strings.c: Check
result of ast_var_assign() calls for memory allocation failure.
We try to keep the system running even when all available memory
is spent. Review: https://reviewboard.asterisk.org/r/2734/
2013-08-05 20:17 +0000 [r396196-396240] Michael L. Young <elgueromexicano@gmail.com>
* channels/chan_sip.c: Fix Registration Failure When A Peer And TLS
Are Used If a peer is used in a register line and TLS is defined
as the transport, the registration fails since the transport on
the dialog is never set properly resulting in UDP being used
instead of TLS. This patch sets the dialog's transport based on
the transport that was defined in the register line. If the
register line does not specify a transport, the parsing function
for the register line always defaults back to UDP. (closes issue
ASTERISK-21964) Reported by: Doug Bailey Tested by: Doug Bailey
Patches: asterisk-21964-set-reg-dialog-transport.diff by Michael
L. Young (license 5026)
* channels/chan_sip.c: Restore Extra Line Break Between Peers When
Running AMI Action SIPPeers The commit (r387133) for fixing
ASTERISK-21466 accidentally removed an extra line break between
the peers returned by the AMI action SIPPeers. This results in
some parsers breaking because they expect this extra line break.
This patch restores that extra line break. (closes issue
ASTERISK-22239) Reported by: Jacek Konieczny Tested by: Jacek
Konieczny, Michael L. Young Patches:
asterisk-ami_sippeers_separator.patch by Jacek Konieczny (license
6298)
* UPGRADE.txt: Adding a note to UPGRADE.txt about a change made to
res_agi in order to indicate when streaming an audio file fails
like it is done in other parts of the code to indicate an error.
Note was requested by Paul Belanger:
http://lists.digium.com/pipermail/asterisk-dev/2013-July/061420.html
(related to issue ASTERISK-21903)
2013-07-22 13:49 +0000 [r394886-395032] Matthew Jordan <mjordan@digium.com>
* main/asterisk.c: Update copyright year to 2013 in asterisk.c;
some whitespace fixes (closes issue ASTERISK-22179) Reported by:
Malcolm Davenport
* funcs/func_channel.c: Clean up documentation This patch cleans up
documentation in func_channel for the following items: *
rtpsource * secure_signaling * secure_media (closes issue
ASTERISK-20969) Reported by: snuffy patches:
func_chan-update.diff uploaded by snuffy (License 5024)
* configs/indications.conf.sample: Provide proper ring tone in
indications.conf for Malaysia The ring tone provided in the
sample indications.conf was incorrect. This patch modifies the
sample ring tone to be what it should: ring =
425/400,0/200,425/400,0/2000 This brings it in line with the tone
definition in DAHDI 2.7.0. (zonedata.c) (closes issue
ASTERISK-21997) Reported by: Filip Jenicek patches:
malaysia_ring.patch uploaded by phill (License 6277)
* main/http.c: Tolerate presence of RFC2965 Cookie2 header by
ignoring it This patch modifies parsing of cookies in Asterisk's
http server by doing an explicit comparison of the "Cookie"
header instead of looking at the first 6 characters to determine
if the header is a cookie header. This avoids parsing "Cookie2"
headers and overwriting the previously parsed "Cookie" header.
Note that we probably should be appending the cookies in each
"Cookie" header to the parsed results; however, while clients can
send multiple cookie headers they never really do. While this
patch doesn't improve Asterisk's behavior in that regard, it
shouldn't make it any worse either. Note that the solution in
this patch was pointed out on the issue by the issue reporter,
Stuart Henderson. (closes issue ASTERISK-21789) Reported by:
Stuart Henderson Tested by: mjordan, Stuart Henderson
* contrib/realtime/postgresql/realtime.sql: Update PostgreSQL
realtime scripts with schema for queue_log table This patch
updates the realtime SQL scripts with an entry that will create
the queue_log table. This brings the PostgreSQL scripts inline
with the MySQL scripts, with respect to what tables they will
create. (closes issue ASTERISK-21021) Reported by: Eugene
patches: queue_log.sql uploaded by varnav (license 6360)
* configs/iax.conf.sample: Document connectedline parameter for
chan_iax2 The connectedline parameter for a chan_iax2 peer was
undocumented. This patch documents the options in the sample
configuration file. (closes issue ASTERISK-21953) Reported by:
Birger "WIMPy" Harzenetter
2013-07-18 12:51 +0000 [r394640] Michael L. Young <elgueromexicano@gmail.com>
* res/res_agi.c: Properly indicate failure to open an audio stream
in res_agi If there is an error streaming an audio file, the
current return status makes it difficult for an AGI script to
determine that there was an error with the audio file. This
patches changes the result to return -1 and the function returns
RESULT_FAILURE instead of RESULT_SUCCESS. From looking at other
parts of res_agi, this would appear to be the proper way to
handle an error. (closes issue ASTERISK-21903) Reported by: Ariel
Wainer Tested by: Ariel Wainer Patches:
asterisk-21903-return-stream-res_1.8.diff by Michael L. Young
(license 5026) Review: https://reviewboard.asterisk.org/r/2625/
2013-07-14 01:53 +0000 [r394302] Matthew Jordan <mjordan@digium.com>
* funcs/func_strings.c: Clarify documentation for function PASSTHRU
It is not apparent to the average user that the PASSTHRU function
should not be passed as ${PASSTHRU(string)} but just as
PASSTHRU(string) to functions which take a variable name and not
its contents. This patch clarifies the behavior in the
documentation and provides an example. (closes issue
ASTERISK-21717) Reported by: Richard Miller patches:
func_strings.diff uploaded by Richard Miller (license 5685)
2013-07-11 16:25 +0000 [r394106] Moises Silva <moises.silva@gmail.com>
* channels/chan_dahdi.c: Fix a longstanding issue with MFC-R2
configuration that prevented users from mixing different variants
or general MFC-R2 settings within the same E1 line. Most users do
not have a problem with this since MFC-R2 lines are usually
fractional E1s, or the whole E1 has the same country variant and
R2 settings. In Venezuela however is common to have inbound
MFC-R2 and outbound DTMF-R2 within the same E1. This fix now
properly parses the chan_dahdi.conf file to generate a new openr2
context every time a new channel => section is found and the
configuration was changed. (closes issue ASTERISK-21117) Reported
by: Rafael Angulo Related Elastix issue:
http://bugs.elastix.org/view.php?id=1612
2013-07-10 01:41 +0000 [r393928] Russell Bryant <russell@russellbryant.com>
* configs/sla.conf.sample, include/asterisk/utils.h,
apps/app_meetme.c: astobj2-ify the SLA code The SLA code within
app_meetme was written before asotbj2 had been merged into
Asterisk. Worse, support for reloads did not exist at first and
was added later as a bolt-on feature. I knew at the time that
reloading was not safe at all while SLA was in use, so the reload
would be queued up to execute when the system was idle.
Unfortunately, this approach was still prone to errors beyond the
fact that this was the only place in Asterisk where configuration
was not reloaded instantly when requested. This patch converts
various SLA objects to be reference counted objects using
astobj2. This allows reloads to be processed while the system is
in use. The code ensures that the objects will not disappear
while one of the other threads is using them. However, they will
be immediately removed from the global trunk and station
containers so no new calls will use them if removed from
configuration. Review: https://reviewboard.asterisk.org/r/2581/
2013-07-03 23:27 +0000 [r393627] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c: chan_dahdi: Fix segfault reloading
chan_dahdi when round robin is used. * Clear round_robin[] in
dahdi_restart(). (closes issue ASTERISK-21847) Reported by: Ivo
Andonov Patches: jira_asterisk_21847_v1.8.patch (license #5621)
patch uploaded by rmudgett
2013-06-14 16:14 +0000 [r391778] Jonathan Rose <jrose@digium.com>
* apps/app_mixmonitor.c: app_mixmonitor: Fix crashes caused by
unloading app_mixmonitor Unloading app_mixmonitor while active
mixmonitors were running would cause a segfault. This patch fixes
that by making it impossible to unload app_mixmonitor while
mixmonitors are active. Review:
https://reviewboard.asterisk.org/r/2624/
2013-06-12 02:19 +0000 [r391489] Matthew Jordan <mjordan@digium.com>
* main/loader.c: Fix memory leak while loading priority modules
When we load a module with the LOAD_PRIORITY flag, we remove its
entry from the load order list. Unfortunately, we don't free the
memory associated with entry in the list. This patch corrects
that and properly frees the memory for the module in the list.
2013-06-11 08:03 +0000 [r391333] Alec L Davis <sivad.a@paradise.net.nz>
* channels/chan_iax2.c: IAX2: Transfer Reject: Lock bridgecallno
before touching it, refactor 1). When touching the bridgecallno,
we need to lock it. 2). Remove magic number '0' and replace with
TRANSFER_NONE. 3). Exit early if no bridgecallno. 4). Reduce
indentation. Reported by: alecdavis Tested by: alecdavis
alecdavis (license 585) Review
https://reviewboard.asterisk.org/r/2613/
2013-07-15 Asterisk Development Team <asteriskteam@digium.com>
* Asterisk 1.8.23.0 Released.
2013-07-12 Asterisk Development Team <asteriskteam@digium.com>
* Asterisk 1.8.23.0-rc2 Released.
* Properly lock and safely handle a transfer failure in IAX2
When touching the bridgecallno, we need to lock it - otherwise a
race condition can occur. This patch does the proper locking
of the bridgecallno before modifying its state.
2013-06-10 Asterisk Development Team <asteriskteam@digium.com>
* Asterisk 1.8.23.0-rc1 Released.
2013-06-10 14:15 +0000 [r391215] Matthew Jordan <mjordan@digium.com>
* UPGRADE.txt, apps/app_queue.c, configs/queues.conf.sample: Add
announce-to-first-user option for app_queue In r386792, the
ability to play prompts to the first caller in a call queue was
added. While this is arguably a bug fix for those who expect the
first caller to continue receiving prompts while the agent is
dialed, it has the side effect of preventing the first caller
from hearing the agent immediately upon bridging. This may not be
a problem for those who really want this option, but for those
who didn't care whether or not the first caller in queue heard
their position, it was an issue. This patch disables the ability
for the first caller in the queue to hear prompts and adds a new
option, announce-to-first-user, to queues.conf. Those who the
behavior can enable it by setting this value to True. Note that
if we ever implement the ability to have the prompts be stopped
upon bridging, this option can be removed. (closes issue
ASTERISK-21782) Reported by: Remi Quezada
2013-06-10 09:30 +0000 [r391062-391143] Alec L Davis <sivad.a@paradise.net.nz>
* channels/chan_iax2.c: chan_iax2: nativebridge refactor, missed
unlock bridgecallno
* channels/chan_iax2.c: fix bad edit after conflict resolution
* channels/chan_iax2.c: IAX2: refactor nativebridge transfer remove
triple checking of iaxs[fr->callno]->transferring reduce
indentation. Reported by: alecdavis Tested by: alecdavis
alecdavis (license 585) Review
https://reviewboard.asterisk.org/r/2602/
* channels/chan_iax2.c: IAX2: fix race condition with nativebridge
transfers. 1). When touching the bridgecallno, we need to lock
it. 2). stop_stuff() which calls iax2_destroy_helper() Assumes
the lock on the pvt is already held, when iax2_destroy_helper()
is called. Thus we need to lock the bridgecallno pvt before we
call stop_stuff(iaxs[fr->callno]->bridgecallno); 3). When
evaluating the state of 'callno->transferring' of the current
leg, we can't change it to READY unless the bridgecallno is
locked. Why, if we are interrupted by the other call leg before
'transferring = TRANSFER_RELEASED', the interrupt will find that
it is READY and that the bridgecallno is also READY so Releases
the legs. (closes issue ASTERISK-21409) Reported by: alecdavis
Tested by: alecdavis alecdavis (license 585) Review
https://reviewboard.asterisk.org/r/2594/
2013-05-31 08:10 +0000 [r390181] Alexandr Anikin <may@telecom-service.ru>
* addons/chan_ooh323.c: reject call attempts when gatekeeper is
configured but not registered (closes issue ASTERISK-21800)
Reported by: Dmitry Melekhov Patches: ASTERISK-21800-1.patch
Tested by: Dmitry Melekhov
2013-05-29 20:10 +0000 [r390044] Richard Mudgett <rmudgett@digium.com>
* main/channel.c: Fix segfault when dealing with chan_agent
channels. Check the returned bridged pointer for NULL to avoid a
crash. It looks like chan_agent is returning a NULL pointer when
it probably should be returning a pointer to the channel the
Agent channel is pretending to be. (closes issue ASTERISK-21793)
Reported by: Rodrigo P. Telles Patches:
jira_asterisk_21793_v1.8.patch (license #5621) patch uploaded by
rmudgett Tested by: Rodrigo P. Telles
2013-05-28 17:35 +0000 [r389895] Jonathan Rose <jrose@digium.com>
* main/slinfactory.c: Fix a memory copying bug in slinfactory which
was causing mixmonitor issues. Reported by: Michael Walton Tested
by: Jonathan Rose Patches: slinfactory.c.ASTERISK-21799.patch
uploaded by Michael Walton (license 6502) (closes issue
ASTERISK-21799)
2013-05-24 11:42 +0000 [r389676] Matthew Jordan <mjordan@digium.com>
* main/logger.c: Print all logger messages on shutdown When
Asterisk shuts down and shuts down the loggin gsubsystem, any
messages currently in flight will not get logged. This patch
prevents the loop writing messages from breaking out prematurely,
such that all of the messages are logged. (closes issue
ASTERISK-21716) Reported by: Corey Farrell patches:
logger-process-all-messages.patch uploaded by Corey Farrell
(license 5909)
2013-05-20 17:43 +0000 [r389244] Jason Parker <jparker@digium.com>
* /: Add doxygen.log to svn:ignore property.
2013-05-15 15:54 +0000 [r388838] kharwell <kharwell@localhost>:
* main/lock.c: Fix for segfault in __ast_rwlock_destroy with
DEBUG_THREADS If DEBUG_THREADS is enabled __ast_rwlock_destroy
causes a segfault while trying to access a possible NULL t->track
object. A NULL check has been added before trying to access the
memory. (closes issue ASTERISK-21724) Reported by: Corey Farrell
Fixed by: Corey Farrell Patches: ast_rwlock_destroy-segv.patch
uploaded by Corey Farrell (license 5909)
2013-05-15 12:37 +0000 [r388768] Kinsey Moore <kmoore@digium.com>
* res/res_srtp.c, configure, include/asterisk/autoconfig.h.in,
configure.ac: Use srtp_shutdown when available This allows the
SRTP library to be shut down properly when the functionality is
offered by libsrtp. Review:
https://reviewboard.asterisk.org/r/2538/ (closes issue
ASTERISK-21719)
2013-05-13 20:34 +0000 [r388596] Kinsey Moore <kmoore@digium.com>
* res/res_srtp.c: Revert r388529 for now Adding the cleanup
function needs some deeper thought since it apparently doesn't
exist for all variants of libsrtp.
2013-05-13 18:16 +0000 [r388532] Jonathan Rose <jrose@digium.com>
* main/pbx.c: pbx: Fix lack of cleanup on macrolock and
context_table (closes issue ASTERISK-21723) Reported by: Corey
Farrell Patches: core-pbx-cleanup.patch uploaded by Correy
Farrell (license 5909)
2013-05-13 18:05 +0000 [r388529] Kinsey Moore <kmoore@digium.com>
* res/res_srtp.c: Close libsrtp properly Ensure that libsrtp is
shutdown properly when res_srtp is unloaded. (closes issue
ASTERISK-21719) Reported by: Corey Farrell Patches:
res_srtp-library-shutdown.patch uploaded by Corey Farrell
2013-05-13 14:24 +0000 [r388477] Richard Mudgett <rmudgett@digium.com>
* main/manager.c: Fix SendText AMI action to never return non-zero.
AMI actions must never return non-zero unless they intend to
close the AMI connection. (Which is almost never.) (closes issue
ASTERISK-21779) Reported by: Paul Goldbaum
2013-05-10 22:09 +0000 [r388423-388425] Richard Mudgett <rmudgett@digium.com>
* channels/misdn/isdn_msg_parser.c: Allow mISDN to send PROGRESS
messsage. * Made isdn_msg_parser.c build a progress message with
the mandatory progress indicator IE. (The mISDNuser NT state
machine rejected sending the incomplete message.) Note: The
associated mISDN and mISDNuser patches respectively are viewable
here: http://svnview.digium.com/svn/thirdparty?view=rev&rev=200
http://svnview.digium.com/svn/thirdparty?view=rev&rev=201 (closes
issue AST-1153) Reported by: Guenther Kelleter Patches:
progress-chan_misdn.diff (license #6372) patch uploaded by
Guenther Kelleter progress-misdn.diff (license #6372) mISDN patch
uploaded by Guenther Kelleter progress-misdnuser.diff (license
#6372) mISDNuser patch uploaded by Guenther Kelleter
* utils: Add version.c to list of ignored files in the utils
directory.
2013-05-10 20:28 +0000 [r388376] Mark Michelson <mmichelson@digium.com>
* pbx/pbx_dundi.c: Fix memory leak in pbx_dundi pbx_dundi added an
io context without removing it. This caused a memory leak when
the module was unloaded. (closes ASTERISK-21718) Reported by
Corey Farrell Patches: pbx_dundi-ast_io_remove.patch uploaded by
Corey Farrell (License #5909)
2013-05-09 03:58 +0000 [r388111] Michael L. Young <elgueromexicano@gmail.com>
* res/res_rtp_asterisk.c: Fix The Payload Being Set On CN Packets
And Do Not Set Marker Bit When we send out a CN packet (for
instance, in the case of using rtpkeepalives), we are not setting
the payload code properly. Also, we are setting the marker bit
when we shouldn't be according to RFC 3389, section 4. AST_RTP_CN
is not defined by AST_FORMAT codes. Therefore, we should be using
ast_rtp_codecs_payload_code() rather than
ast_rtp_codecs_payload_lookup(). 11 and trunk already use the
appropriate function. * In 1.8, use ast_rtp_codecs_payload_code()
* Remove the setting of the marker bit * Fix the debug message by
incrementing the seqno after the debug message is set in order to
display the correct seqno that was sent out (closes issue
ASTERISK-21246) Reported by: Peter Katzmann Tested by: Peter
Katzmann, Michael L. Young Patches:
asterisk-21246-rtp-cng-payload-error_1.8_v2.diff uploaded by
Michael L. Young (license 5026) Review:
https://reviewboard.asterisk.org/r/2500/
2013-05-08 07:17 +0000 [r387875] Alec L Davis <sivad.a@paradise.net.nz>
* channels/chan_sip.c: chan_sip: NOTIFYs for BLF start queuing up
and fail to be sent out after retries fail RFC6665 4.2.2: ...
after a failed State NOTIFY transaction remove the subscription
The problem is that the State Notify requests rely on the 200OK
reponse for pacing control and to not confuse the notify
susbsystem. The issue is, the pendinginvite isn't cleared if a
response isn't received, thus further notify's are never sent.
The solution, follow RFC 6665 4.2.2's 'SHOULD' and remove the
subscription after failure. (closes issue ASTERISK-21677)
Reported by: Dan Martens Tested by: Dan Martens, David Brillert,
alecdavis alecdavis (license 585) Review
https://reviewboard.asterisk.org/r/2475/
2013-05-06 15:52 +0000 [r387688] Russell Bryant <russell@russellbryant.com>
* apps/app_meetme.c: Make SLA reload more paranoid. Reload support
was originally not included for SLA. It was added later, but in a
fairly non-traditional way. It basically sets a flag indicating
that a reload is pending, and then waits for a time where it
thinks everything SLA related is idle and unused, and *then*
executes the reload. It does this because the reload process is
destructive. It starts by throwing everything away and starting
over. There are a number of problems with this approach. One of
them is that the check to see if anything in use was incomplete.
This patch makes it more complete and thus less likely for a
crash to occur during reload processing. However, this approach
still has problems so some much more significant reworking of
this code will need to come in as a next step. Patch credit and
testing by CoreDial, LLC.
2013-05-02 17:11 +0000 [r387421] Matthew Jordan <mjordan@digium.com>
* utils/Makefile: Update utils Makefile to handle r387294 Alec's
patch that added the Asterisk version to 'core show locks'
angered the items in utils, as they exist somewhat outside of the
Asterisk build system. Some day, this Makefile should get nuked
from high orbit, but for now, include version.c in its list of
stuff to pile in.
2013-05-02 07:53 +0000 [r387294-387344] Alec L Davis <sivad.a@paradise.net.nz>
* channels/sip/include/sip.h, channels/chan_sip.c: chan_sip:
Session-Expires: Set timer to correctly expire at (~2/3) of the
interval when not the refresher RFC 4028 Section 10 if the side
not performing refreshes does not receive a session refresh
request before the session expiration, it SHOULD send a BYE to
terminate the session, slightly before the session expiration.
The minimum of 32 seconds and one third of the session interval
is RECOMMENDED. Prior to this asterisk would refresh at 1/2 the
Session-Expires interval, or if the remote device was the
refresher, asterisk would timeout at interval end. Now, when not
refresher, timeout as per RFC noted above. (closes issue
ASTERISK-21742) Reported by: alecdavis Tested by: alecdavis
alecdavis (license 585) Review
https://reviewboard.asterisk.org/r/2488/
* channels/chan_sip.c: chan_sip: Honor Session-Expires in 200OK
response when it's a RE-INVITE when asterisk is the refresher.
RFC 4028 Section 7.2 "UACs MUST be prepared to receive a
Session-Expires header field in a response, even if none were
present in the request." What changed After ASTERISK-20787,
inbound calls to asterisk with no Session-Expires in the INVITE
are now are offered a Session-Expires (1800 asterisk default) in
the response, with asterisk as the refresher. Symptom: After 900
seconds (asterisk default refresher period 1800), asterisk
RE-INVITEs the device, the device may respond with a much lower
Session-Expires (180 in our case) value that it is now using.
Asterisk ignores this response, as it's deemed both an INBOUND
CALL, and a RE-INVITE. After 180 seconds the device times out and
sends BYE (hangs up), asterisk is still working with the
refresher period of 1800 as it ignored the 'Session Expires: 180'
in the previous 200OK response. Fix: handle_response_invite()
when 200OK, remove check for outbound and reinvite. (closes issue
ASTERISK-21664) Reported by: alecdavis Tested by: alecdavis
alecdavis (license 585) Review
https://reviewboard.asterisk.org/r/2463/
* channels/chan_dahdi.c: chan_dahdi: fix lower bound check with -ve
integer conversion from a float Lower bound of a 16bit signed int
is -32768 not -32767 (closes issue ASTERISK-21744) Reported by:
alecdavis Tested by: alecdavis alecdavis (license 585)
* main/utils.c: Add Asterisk Version to core show locks Assist with
reporting 'core show locks' when submitting bug reports. Example
below: =========================== == SVN-branch-1.8-... ==
Currently Held Locks =========================== (closes issue
ASTERISK-21743) Reported by: alecdavis Tested by: alecdavis
alecdavis (license 585)
2013-05-01 21:15 +0000 [r387036-387213] Matthew Jordan <mjordan@digium.com>
* res/res_rtp_asterisk.c: Clear the DTMF sending digit tracking on
off nominal paths In certain situations, when the RTP engine goes
to send a DTMF end digit it may be in a situation where the
remote address is no longer available, or the digit that was
supposed to be sent is invalid. In such cases, we need to clear
the RTP counters appropriately. Otherwise, when the RTP source is
set again, we'll continue to think that we're in the middle of
sending a DTMF digit, which can confuse the remote party
(signficantly). (closes issue ASTERISK-21522) Reported by: Corey
Farrell patches: rtp_dtmf_process_end.patch uploaded by Corey
Farrell (License 5909)
* channels/chan_sip.c: Prevent crash in 'sip show peers' when the
number of peers on a system is large When you have lots of SIP
peers (according to the issue reporter, around 3500), the 'sip
show peers' CLI command or AMI action can crash due to a poorly
placed string duplication that occurs on the stack. This patch
refactors the command to not allocate the string on the stack,
and handles the formatting of a single peer in a separate
function call. (closes issue ASTERISK-21466) Reported by:
Guillaume Knispel patches:
fix_sip_show_peers_stack_overflow_asterisk_11.3.0-v2.patch
uploaded by gknispel (License 6492)
* main/features.c: Fix CDR not being created during an externally
initiated blind transfer Way back when in the dark days of
Asterisk 1.8.9, blind transferring a call in a context that
included the 'h' extension would inadvertently execute the hangup
code logic on the transferred channel. This was a "bad thing".
The fix was to properly check for the softhangup flags on the
channel and only execute the 'h' extension logic (and, in later
versions, hangup handler logic) if the channel was well and truly
dead (Jim). Unfortunately, CDRs are fickle. Setting the
softhangup flag when we detected that the channel was leaving the
bridge (but not to die) caused some crucial snippet of CDR code,
lying in ambush in the middle of the bridging code, to not get
executed. This had the effect of blowing away one of the CDRs
that is typically created during a blind transfer. While we live
and die by the adage "don't touch CDRs in release branches", this
was our bad. The attached patch restores the CDR behavior, and
still manages to not run the 'h' extension during a blind
transfer (at least not when it's supposed to). Thanks to Steve
Davies for diagnosing this and providing a fix. Review:
https://reviewboard.asterisk.org/r/2476 (closes issue
ASTERISK-21394) Reported by: Ishfaq Malik Tested by: Ishfaq
Malik, mjordan patches: fix_missing_blindXfer_cdr2 uploaded by
one47 (License 5012)
2013-04-30 13:45 +0000 [r386929] Sean Bright <sean@malleable.com>
* include/asterisk/utils.h: Use the proper lower bound when doing
saturation arithmetic. 16 bit signed integers have a range of
[-32768, 32768). The existing code was using the interval
(-32768, 32768) instead. This patch fixes that. Review:
https://reviewboard.asterisk.org/r/2479/
2013-04-29 23:34 +0000 [r386877] Rusty Newton <rnewton@digium.com>
* sounds/Makefile: Modifying sounds/Makefile to pull down 1.4.24
core sounds 1.4.24 core sounds includes a full set of Italian
prompts for core sounds and a fix for the missing voicemail
prompts in the Russian language. (closes issue ASTERISK-19431)
(closes issue ASTERISK-19721)
2013-04-29 08:36 +0000 [r386792] Olle Johansson <oej@edvina.net>
* CHANGES, apps/app_queue.c: Play periodic prompst for first call
in a call queue Review: https://reviewboard.asterisk.org/r/2263/
2013-04-26 21:26 +0000 [r386641-386672] Matthew Jordan <mjordan@digium.com>
* main/config.c: Clean up memory leak in config file on off nominal
paths when glob is allowed If a system allows for its usage,
Asterisk will use glob to help parse Asterisk .conf files. The
config file loading routine was leaking the memory allocated by
the glob() routine when the config file was in an unmodified or
invalid state. This patch properly calls globfree in those off
nominal paths. (closes issue ASTERISK-21412) Reported by: Corey
Farrell patches: config_glob_leak.patch uploaded by Corey Farrell
(license 5909)
* main/features.c: Clean up resources in features on exit This
patch cleans up two things features: * It properly unregisters
the CLI commands that features registered * It cancels and
performs a pthread_join on the created parking thread. This not
only properly joins a non-detached thread, but also prevents
disposing of the parking lots prior to the parking thread
completely exiting. (closes issue ASTERISK-21407) Reported by:
Corey Farrell patches: features_shutdown-r2.patch uploaded by
Corey Farrell (License 5909)
2013-04-25 02:43 +0000 [r386483] Michael L. Young <elgueromexicano@gmail.com>
* channels/chan_sip.c: Change Case On Forcerport For Consistency *
Change "ForcerPort" to "Forcerport" to match everywhere else it
is displayed
2013-04-22 16:10 +0000 [r386256] Richard Mudgett <rmudgett@digium.com>
* main/channel.c: Fix crash when AMI redirect action redirects two
channels out of a bridge. The two party bridging loops were
changing the bridge peer pointers without the channel locks held.
Thus when ast_channel_massquerade() tested and used the pointer
there is a small window of opportunity for the pointers to become
NULL even though the masquerade code has the channels locked.
(closes issue ASTERISK-21356) Reported by: William luke Patches:
jira_asterisk_21356_v11.patch (license #5621) patch uploaded by
rmudgett Tested by: William luke
2013-04-19 15:59 +0000 [r386109] Matthew Jordan <mjordan@digium.com>
* res/res_timing_pthread.c: Prevent res_timing_pthread from
blocking callers There were several reports of deadlock when
using res_timing_pthread. Backtraces indicated that one thread
was blocked waiting for the write to the pipe to complete and
this thread held the container lock for the timers. Therefore any
thread that wanted to create a new timer or read an existing
timer would block waiting for either the timer lock or the
container lock and deadlock ensued. This patch changes the way
the pipe is used to eliminate this source of deadlocks: 1) The
pipe is placed in non-blocking mode so that it would never block
even if the following changes someone fail... 2) Instead of
writing bytes into the pipe for each "tick" that's fired the pipe
now has two states--signaled and unsignaled. If signaled, the
pipe is hot and any pollers of the read side filedescriptor will
be woken up. If unsigned the pipe is idle. This eliminates even
the chance of filling up the pipe and reduces the potential
overhead of calling unnecessary writes. 3) Since we're tracking
the signaled / unsignaled state, we can eliminate the exta poll
system call for every firing because we know that there is data
to be read. (closes issue ASTERISK-21389) Reported by: Matt
Jordan Tested by: Shaun Ruffell, Matt Jordan, Tony Lewis patches:
0001-res_timing_pthread-Reduce-probability-of-deadlocking.patch
uploaded by sruffell (License 5417) (closes issue ASTERISK-19754)
Reported by: Nikola Ciprich (closes issue ASTERISK-20577)
Reported by: Kien Kennedy (closes issue ASTERISK-17436) Reported
by: Henry Fernandes (closes issue ASTERISK-17467) Reported by:
isrl (closes issue ASTERISK-17458) Reported by: isrl Review:
https://reviewboard.asterisk.org/r/2441/
2013-04-19 05:18 +0000 [r386049] David M. Lee <dlee@digium.com>
* main/cli.c: cli.c: Properly initialize debug_modules and
verbose_modules. This avoids some lock errors on the core set
{debug,verbose} commands.
2013-04-16 23:11 +0000 [r385916] Alec L Davis <sivad.a@paradise.net.nz>
* main/devicestate.c, res/res_jabber.c: Distributed Device State
broken at sites using res_xmpp or res_jabber where Secuity
Advisory AST-2012-015 is inplace res_jabber/res_xmpp were not
adding AST_EVENT_IE_CACHABLE to the event as each message came
in, then devstate_change_collector_cb() was unable to find
AST_EVENT_IE_CACHABLE in the event, so defaulted incorrectly to
AST_DEVSTATE_NOT_CACHABLE. (issue ASTERISK-20175) (closes issue
ASTERISK-21429) (closes issue ASTERISK-21069) (closes issue
ASTERISK-21164) Reported by: alecdavis Tested by: alecdavis
alecdavis (license 585) Review
https://reviewboard.asterisk.org/r/2452/
2013-04-15 17:07 +0000 [r385745] Jason Parker <jparker@digium.com>
* Makefile: Don't unnecessarily rebuild things on every run of
'make'. Review: https://reviewboard.asterisk.org/r/2449/
2013-04-15 14:38 +0000 [r385683] David M. Lee <dlee@digium.com>
* BSDmakefile, contrib/realtime/mysql/voicemail_data.sql,
build_tools/sha1sum-sh, res/res_mutestream.c,
configs/res_curl.conf.sample, tests/test_func_file.c,
include/asterisk/select.h, res/res_rtp_multicast.c,
include/asterisk/bridging_technology.h, tests/test_locale.c,
include/asterisk/bridging_features.h, doc/Makefile,
tests/test_poll.c, res/res_timing_kqueue.c,
contrib/realtime/mysql/musiconhold.sql,
contrib/realtime/mysql/queue_log.sql, channels/sig_ss7.c,
channels/sig_ss7.h, channels/chan_multicast_rtp.c,
tests/test_expr.c, apps/app_saycounted.c,
contrib/realtime/mysql/voicemail_messages.sql: Fix the
svn:keywords property on several files. Normally I think keyword
expansion is silly, but the one time it would have been good, it
didn't work because the property had quotes in it. This patch
fixes obviously busted svn:keywords properties.
2013-04-14 02:58 +0000 [r385633-385636] Matthew Jordan <mjordan@digium.com>
* res/res_rtp_multicast.c: Calculate the timestamp for outbound RTP
if we don't have timing information This patch calculates the
timestamp for outbound RTP when we don't have timing information.
This uses the same approach in res_rtp_asterisk. Thanks to both
Pietro and Tzafrir for providing patches. (closes issue
ASTERISK-19883) Reported by: Giacomo Trovato Tested by: Pietro
Bertera, Tzafrir Cohen patches: rtp-timestamp-1.8.patch uploaded
by tzafrir (License 5035) rtp-timestamp.patch uploaded by
pbertera (License 5943)
* channels/chan_alsa.c: Don't attempt to create a voice frame on a
read error Prior to this patch, a read error in snd_pcm_readi
would still be treated as a nominal result when constructing a
voice frame from the expected data. Since the value returned is
negative, as opposed to the number of samples read, this could
result in a crash. With this patch, we now return a null frame
when a read error is detected. Note that the patch on
ASTERISK-21329 was modified slightly for this commit, in that we
bail immediately on detecting the read error, rather than
bypassing the construction of the voice frame. (closes issue
ASTERISK-21329) Reported by: Keiichiro Kawasaki patches:
chan_alsa.diff uploaded by kawasaki (License 6489)
2013-04-12 22:34 +0000 [r385551-385593] Michael L. Young <elgueromexicano@gmail.com>
* apps/app_queue.c: Fix Manager Segfault When app_queue Is Unloaded
When app_queue is unloaded, some manager commands are not being
unregistered which result in a segfault. This patch corrects
this. (closes issue ASTERISK-21397) Reported by: Peter Katzmann,
Corey Farrell Tested by: Corey Farrell Patches:
asterisk-21397-missing-unreg-manager-cmd_1.8.diff Michael L.
Young (license 5026)
asterisk-21397-missing-unreg-manager-cmd_11.diff Michael L. Young
(license 5026) Review: https://reviewboard.asterisk.org/r/2444/
* apps/app_voicemail.c: Fix app_voicemail Segfault And A Few Memory
Leaks The original report was that app_voicemail would crash.
This was caused by ast_config_load() returning
CONFIG_STATUS_FILEINVALID but no checks being performed for that
return status. After adding the initial patch to fix this issue,
Jaco Kroon (jkroon) added some fixes to memory leaks he had
discovered. During review, Walter Doekes (wdoekes) suggested
adding a helper function in order to determine if we had a valid
configuration or not. This patch does the following: * Creates a
helper function to check if the configuration is valid * Adds
calls to the new helper function where appropiate * Fixes memory
leaks where the code returned without running
ast_config_destroy() on the configuration that was loaded (closes
issue ASTERISK-21302) Reported by: Jaco Kroon Tested by: Jaco
Kroon, Michael L. Young Patches:
asterisk-11.3.0-app_voicemail-ast_config-fixes.patch Jaco Kroon
(license 5671) asterisk-21302-valid_cfg_and_mem_leaks_v3-1.8.diff
Michael L. Young (license 5026) Review:
https://reviewboard.asterisk.org/r/2443/
2013-04-12 08:46 +0000 [r385402-385429] Alec L Davis <sivad.a@paradise.net.nz>
* channels/chan_iax2.c: IAX2 defer_full_frames fail to get sent
Ensure iax2_process_thread is signalled when a deferred frame is
queued to it. (issue ASTERISK-18827) Reported by: alecdavis
Tested by: alecdavis alecdavis (license 585) Review
https://reviewboard.asterisk.org/r/2426/
* channels/chan_iax2.c: IAX2, prevent network thread starting
before all helper threads are ready On startup, it's possible for
a frame to arrive before the processing threads were ready. In
iax2_process_thread() the first pass through falls into
ast_cond_wait, should a frame arrive before we are at
ast_cond_wait, the signal will be ignored. The result
iax2_process_thread stays at ast_cond_wait forever, with deferred
frames being queued. Fix: When creating initial idle
iax2_process_threads, wait for init_cond to be signalled after
each thread is started. (issue ASTERISK-18827) Reported by:
alecdavis Tested by: alecdavis alecdavis (license 585) Review
https://reviewboard.asterisk.org/r/2427/
2013-04-10 14:22 +0000 [r385170-385190] Matthew Jordan <mjordan@digium.com>
* res/res_config_ldap.c: Use LDAP memory management functions
instead of Asterisk's When MALLOC_DEBUG is enabled with
res_config_ldap, issues (munmap_chunk: invalid pointer errors)
can occur as the memory is being allocated with Asterisk's
wrappers around malloc/calloc/free/strdup, as opposed to the LDAP
library's wrappers. This patch uses the LDAP library's wrappers
where appropriate, so that compiling with MALLOC_DEBUG doesn't
cause more problems than it solves. Note that the patch listed
below was modified slightly for this commit to account for some
additional memory allocation/deallocations. (closes issue
ASTERISK-17386) Reported by: John Covert Tested by: Andrew Latham
patches: issue18789-1.8-r316873.patch uploaded by seanbright
(License 5060)
* channels/chan_sip.c: Fix crash in chan_sip when a core initiated
op occurs at the same time as a BYE When a BYE request is
processed in chan_sip, the current SIP dialog is detached from
its associated Asterisk channel structure. The tech_pvt pointer
in the channel object is set to NULL, and the dialog persists for
an RFC mandated period of time to handle re-transmits. While this
process occurs, the channel is locked (which is good).
Unfortunately, operations that are initiated externally have no
way of knowing that the channel they've just obtained (which is
still valid) and that they are attempting to lock is about to
have its tech_pvt pointer removed. By the time they obtain the
channel lock and call the channel technology callback, the
tech_pvt is NULL. This patch adds a few checks to some channel
callbacks that make sure the tech_pvt isn't NULL before using it.
Prime offenders were the DTMF digit callbacks, which would crash
if AMI initiated a DTMF on the channel at the same time as a BYE
was received from the UA. This patch also adds checks on
sip_transfer (as AMI can also cause a callback into this
function), as well as sip_indicate (as lots of things can queue
an indication onto a channel). Review:
https://reviewboard.asterisk.org/r/2434/ (closes issue
ASTERISK-20225) Reported by: Jeff Hoppe
2013-04-08 23:34 +0000 [r385047] Rusty Newton <rnewton@digium.com>
* configs/extconfig.conf.sample: Modified the list of keys for the
driver backends for sake of sample clarity Added a line showing
the mapping of "mysql" to res_config_mysql available in add-ons.
We used "mysql" as an example driver key in the sample, but
didn't show what module it mapped too. Also added a subtitle
above the list of keys for driver backends.
2013-04-08 19:55 +0000 [r385008] Michael L. Young <elgueromexicano@gmail.com>
* UPGRADE.txt, channels/chan_sip.c: Fix For Not Overriding The
Default Settings In chan_sip The initial report was that the
"nat" setting in the [general] section was not having any effect
in overriding the default setting. Upon confirming that this was
happening and looking into what was causing this, it was
discovered that other default settings would not be overriden as
well. This patch works similar to what occurs in build_peer(). We
create a temporary ast_flags structure and using a mask, we
override the default settings with whatever is set in the
[general] section. In the bug report, the reporter who helped to
test this patch noted that the directmedia settings were being
overriden properly as well as the nat settings. (closes issue
ASTERISK-21225) Reported by: Alexandre Vezina Tested by:
Alexandre Vezina, Michael L. Young Patches:
asterisk-21225-handle-options-default-prob_1.8_v4.diff.diff
Michael L. Young (license 5026) Review:
https://reviewboard.asterisk.org/r/2386/
2013-04-04 19:31 +0000 [r384779] Michael L. Young <elgueromexicano@gmail.com>
* contrib/realtime/postgresql/realtime.sql,
contrib/realtime/mysql/sippeers.sql, channels/chan_sip.c:
Backport Appropiate NAT Setting Cleanup In ASTERISK-20904, the
focus was around the changes to NAT that took place in Asterisk
11. Since the report stated that 1.8 was fine, we didn't take a
look at 1.8 at the time. While working on ASTERISK-21225, I could
see that 1.8 would benefit from having some of those changes
applied to it. This patch does the following: * The important
part of this patch is that it sets the peer's flags earlier in
build_peer so that the code properly uses the peer's flags based
on the peer's configuration. * constify req parameter in
check_via() * update realtime schemas under the contrib directory
to handle properly the NAT settings available in 1.8 as well as
to handle the changes made in 11 to make upgrading easier when
installing newer versions of Asterisk (closes issue
ASTERISK-21243) Reported by: Michael L. Young Patches:
asterisk-20904-changes_for_1.8.diff Michael L. Young (license
5026) Review: https://reviewboard.asterisk.org/r/2422/
2013-04-03 20:13 +0000 [r384685] Richard Mudgett <rmudgett@digium.com>
* channels/sig_pri.c, channels/sig_pri.h, channels/chan_dahdi.c,
configs/chan_dahdi.conf.sample: chan_dahdi: Add
inband_on_proceeding compatibility option. The new
inband_on_proceeding option causes Asterisk to assume inband
audio may be present when a PROCEEDING message is received. Q.931
Section 5.1.2 says the network cannot assume that the CPE side
has attached to the B channel at this time without explicitly
sending the progress indicator ie informing the CPE side to
attach to the B channel for audio. However, some non-compliant
ISDN switches send a PROCEEDING without the progress indicator ie
indicating inband audio is available and assume that the CPE
device has connected the media path for listening to ringback and
other messages. ASTERISK-17834 which causes this issue was
dealing with a non-compliant network switch. (closes issue
ASTERISK-21151) Reported by: Gianluca Merlo Tested by: rmudgett
2013-04-03 17:05 +0000 [r384640] Matthew Jordan <mjordan@digium.com>
* funcs/func_channel.c: Update documentation for CHANNEL function
Document that you can read/write the 'accountcode' and 'amaflags'
on a channel.
2013-04-02 17:33 +0000 [r384544] David M. Lee <dlee@digium.com>
* Makefile: Fixed spurious rebuilds of func_version.
func_version.so was being rebuilt every time, because build.h was
changing every build, because of the cleantest dependency that
was added in r384410 to fix parallel make bugs. Now build.h will
only be created if it does not exist, which was the original
behavior of the Makefile.
2013-04-01 13:18 +0000 [r384410] David M. Lee <dlee@digium.com>
* Makefile: Fix parallel make problems. Occasionally, make -j would
fail due to missing includes, or other unusual errors. This was
due to the 'cleantest' target, which was designed to force a make
clean when some change in the code would cause the typical
depedency checking to fail. Several targets in the main Makefile
did not depend upon cleantest, hence would run in parallel to it.
By adding the dependency, make -j runs happily now. Review:
https://reviewboard.asterisk.org/r/2418/
2013-03-29 16:23 +0000 [r384325] Jonathan Rose <jrose@digium.com>
* apps/app_voicemail.c: app_voicemail: Add blank argument to
externnotify if no context argument At least one call to
run_externnotify provides a NULL context parameter and because
the snprintf statement doesn't account for a NULL context
parameter, it simply writes '(null)' to the arguments string
instead. This patch makes it write two quotes back to back for
that argument instead in the event of a NULL context. (closes
issue ASTERISK-18207) Reported by: Barry L. Kline Patches:
modified from patch-20130306 uploaded by Karsten Wemheuer
(License 5930)
2013-05-17 Asterisk Development Team <asteriskteam@digium.com>
* Asterisk 1.8.22.0 Released.
2013-05-13 Asterisk Development Team <asteriskteam@digium.com>
* Asterisk 1.8.22.0-rc2 Released.
* Distributed Device State broken at sites using res_xmpp or res_jabber
where Secuity Advisory AST-2012-015 is inplace
res_jabber/res_xmpp were not adding AST_EVENT_IE_CACHABLE to the
event as each message came in, then devstate_change_collector_cb()
was unable to find AST_EVENT_IE_CACHABLE in the event, so defaulted
incorrectly to AST_DEVSTATE_NOT_CACHABLE.
* Fix CDR not being created during an externally initiated blind
transfer
Way back when in the dark days of Asterisk 1.8.9, blind transferring
a call in a context that included the 'h' extension would
inadvertently execute the hangup code logic on the transferred
channel. This was a "bad thing". The fix was to properly check for
the softhangup flags on the channel and only execute the 'h'
extension logic (and, in later versions, hangup handler logic) if
the channel was well and truly dead (Jim).
Unfortunately, CDRs are fickle. Setting the softhangup flag when we
detected that the channel was leaving the bridge (but not to die)
caused some crucial snippet of CDR code, lying in ambush in the
middle of the bridging code, to not get executed. This had the
effect of blowing away one of the CDRs that is typically created
during a blind transfer.
While we live and die by the adage "don't touch CDRs in release
branches", this was our bad. The attached patch restores the CDR
behavior, and still manages to not run the 'h' extension during a
blind transfer (at least not when it's supposed to).
Thanks to Steve Davies for diagnosing this and providing a fix.
* Prevent res_timing_pthread from blocking callers
There were several reports of deadlock when using res_timing_pthread.
Backtraces indicated that one thread was blocked waiting for the
write to the pipe to complete and this thread held the container lock
for the timers. Therefore any thread that wanted to create a new
timer or read an existing timer would block waiting for either the
timer lock or the container lock and deadlock ensued.
This patch changes the way the pipe is used to eliminate this source
of deadlocks:
1) The pipe is placed in non-blocking mode so that it would never
block even if the following changes someone fail...
2) Instead of writing bytes into the pipe for each "tick" that's
fired the pipe now has two states--signaled and unsignaled. If
signaled, the pipe is hot and any pollers of the read side
filedescriptor will be woken up. If unsigned the pipe is idle.
This eliminates even the chance of filling up the pipe and reduces
the potential overhead of calling unnecessary writes.
3) Since we're tracking the signaled / unsignaled state, we can
eliminate the exta poll system call for every firing because we know
that there is data to be read.
* Fix crash when AMI redirect action redirects two channels out of a
bridge.
The two party bridging loops were changing the bridge peer pointers
without the channel locks held. Thus when ast_channel_massquerade()
tested and used the pointer there is a small window of opportunity
for the pointers to become NULL even though the masquerade code has
the channels locked.
2013-03-28 Asterisk Development Team <asteriskteam@digium.com>
* Asterisk 1.8.22.0-rc1 Released.
2013-03-27 19:50 +0000 [r384162] Kinsey Moore <kmoore@digium.com>
* channels/chan_sip.c: Address uninitialized conditional that
valgrind found
2013-03-27 18:49 +0000 [r384118] Matthew Jordan <mjordan@digium.com>
* main/http.c: Fix a file descriptor leak in off nominal path While
looking at the security vulnerability in ASTERISK-20967, Walter
noticed a file descriptor leak and some other issues in off
nominal code paths. This patch corrects them. Note that this
patch is not related to the vulnerability in ASTERISK-20967, but
the patch was placed on that issue. (closes issue ASTERISK-20967)
Reported by: wdoekes patches:
issueA20967_file_leak_and_unused_wkspace.patch uploaded by
wdoekes (License 5674)
2013-03-27 17:02 +0000 [r384048] Kinsey Moore <kmoore@digium.com>
* res/res_rtp_asterisk.c: Fix white noise on SRTP decryption When
res_rtp_asterisk.c was altered to avoid attempting to apply
unprotect algorithms to non-audio RTP packets, the test used was
incorrect. This caused the audio packets to not be decrypted and
resulted in loud white noise on the other endpoint (or both
endpoints depending on the call legs involved). The test now
properly checks the version field in the RTP header to ensure
that RTP and RTCP are decrypted while other types of packets are
not. (closes issue ASTERISK-21323) Reported by: andrea Tested by:
Kinsey Moore, andrea, John Bigelow Patches: whitenoise_fix.diff
uploaded by Kinsey Moore
2013-03-27 14:53 +0000 [r383976-383981] Matthew Jordan <mjordan@digium.com>
* channels/sip/include/sip.h, channels/chan_sip.c: AST-2013-003:
Prevent username disclosure in SIP channel driver When
authenticating a SIP request with alwaysauthreject enabled,
allowguest disabled, and autocreatepeer disabled, Asterisk
discloses whether a user exists for INVITE, SUBSCRIBE, and
REGISTER transactions in multiple ways. The information is
disclosed when: * A "407 Proxy Authentication Required" response
is sent instead of a "401 Unauthorized" response * The presence
or absence of additional tags occurs at the end of "403
Forbidden" (such as "(Bad Auth)") * A "401 Unauthorized" response
is sent instead of "403 Forbidden" response after a
retransmission * Retransmission are sent when a matching peer did
not exist, but not when a matching peer did exist. This patch
resolves these various vectors by ensuring that the responses
sent in all scenarios is the same, regardless of the presence of
a matching peer. This issue was reported by Walter Doekes, OSSO
B.V. A substantial portion of the testing and the solution to
this problem was done by Walter as well - a huge thanks to his
tireless efforts in finding all the ways in which this setting
didn't work, providing automated tests, and working with Kinsey
on getting this fixed. (closes issue ASTERISK-21013) Reported by:
wdoekes Tested by: wdoekes, kmoore patches: AST-2013-003-1.8
uploaded by kmoore, wdoekes (License 6273, 5674) AST-2013-003-10
uploaded by kmoore, wdoekes (License 6273, 5674) AST-2013-003-11
uploaded by kmoore, wdoekes (License 6273, 5674)
* main/http.c: AST-2013-002: Prevent denial of service in HTTP
server AST-2012-014, fixed in January of this year, contained a
fix for Asterisk's HTTP server for a remotely-triggered crash.
While the fix put in place fixed the possibility for the crash to
be triggered, a denial of service vector still exists with that
solution if an attacker sends one or more HTTP POST requests with
very large Content-Length values. This patch resolves this by
capping the Content-Length at 1024 bytes. Any attempt to send an
HTTP POST with Content-Length greater than this cap will not
result in any memory allocation. The POST will be responded to
with an HTTP 413 "Request Entity Too Large" response. This issue
was reported by Christoph Hebeisen of TELUS Security Labs (closes
issue ASTERISK-20967) Reported by: Christoph Hebeisen patches:
AST-2013-002-1.8.diff uploaded by mmichelson (License 5049)
AST-2013-002-10.diff uploaded by mmichelson (License 5049)
AST-2013-002-11.diff uploaded by mmichelson (License 5049)
2013-03-26 02:23 +0000 [r383839-383863] Matthew Jordan <mjordan@digium.com>
* channels/chan_sip.c: Resolve deadlock between SIP registration
and channel based functions In r373424, several reentrancy
problems in chan_sip were addressed. As a result, the SIP channel
driver is now properly locking the channel driver private
information in certain operations that it wasn't previously. This
exposed two latent problems either in register_verify or by
functions called by register_verify. This includes: * Holding the
private lock while calling sip_send_mwi_to_peer. This can create
a new sip_pvt via sip_alloc, which will obtain the channel
container lock. This is a locking inversion, as any channel
related lock must be obtained prior to obtaining the SIP channel
technology private lock. * Holding the privat elock while calling
sip_poke_peer. In the same vein as sip_send_mwi_to_peer,
sip_poke_peer can create a new SIP private, causing the same
locking inversion. Note that this locking inversion typically
occured when CLI commands were run while a SIP REGISTER request
was being processed, as many CLI commands (such as 'sip show
channels', 'core show channels', etc.) have to obtain the channel
container lock. (issue ASTERISK-21068) Reported by: Nicolas
Bouliane (issue ASTERISK-20550) Reported by: David Brillert
(issue ASTERISK-21314) Reported by: Badalian Vyacheslav (issue
ASTERISK-21296) Reported by: Gabriel Birke
* main/cdr.c: Resolve deadlock between pending CDR and batch CDR
locks r375757 attempted to resolve a race condition between
multiple submissions of CDRs while in batch mode from attempting
to destroy the scheduled batch submission by extending the batch
CDR lock. Unfortunately, this causes a deadlock between the
pending CDR lock and the batch CDR lock. This patch resolves the
intent of r375757 by simply providing a new lock that protects
the scheduling of the batches. The original batch CDR lock is
kept to protect manipulation of the batch CDR settings, but has
been placed such that it is not held when the pending lock is
held. Thanks to Chase Venters for providing lock analysis on the
issue. (issue ASTERISK-21162) Reported by: Chase Venters
2013-03-26 01:32 +0000 [r383835] Russell Bryant <russell@russellbryant.com>
* apps/app_meetme.c: Fix multi-station answer race condition. When
an SLA trunk is ringing (inbound call on the trunk) Asterisk will
make outbound calls to the stations that have that trunk. If more
than one station answers the call at the same time, all channels
other than the first one to answer are left in a bad state. The
channel gets leaked, is not connected to anything, and there's no
way to get rid of it. We now properly clean up these losing
channels by hanging up on them. Since they lost the race, as we
process their answer, there is no ringing trunk for them to
answer.
2013-03-25 23:19 +0000 [r383796] Richard Mudgett <rmudgett@digium.com>
* channels/sig_pri.c: Set the CALLERID(dnid-num-plan) for incoming
ISDN calls. The CALLEDTON channel variable is set for incoming
ISDN calls to the lower 7 bits of the Q.931
type-of-number/numbering-plan octet. The CALLERID(dnid-num-plan)
should have the same value. (closes issue ASTERISK-21248)
Reported by: rmudgett
2013-03-25 12:35 +0000 [r383667] Sean Bright <sean@malleable.com>
* res/res_config_curl.c: Properly delimit post data in
res_config_curl.
2013-03-20 20:22 +0000 [r383460] Walter Doekes <walter+asterisk@wjd.nu>
* funcs/func_curl.c: Have func_curl log a warning when a curl
request fails. Review: https://reviewboard.asterisk.org/r/2403/
2013-03-19 15:50 +0000 [r383340] David M. Lee <dlee@digium.com>
* codecs/Makefile: Removed codecs/g722/*.i on make clean
2013-03-15 12:49 +0000 [r383165] Kinsey Moore <kmoore@digium.com>
* main/tcptls.c, main/manager.c, channels/chan_sip.c, main/http.c:
tcptls: Prevent unsupported options from being set AMI, HTTP, and
chan_sip all support TLS in some way, but none of them support
all the options that Asterisk's TLS core is capable of
interpreting. This prevents consumers of the TLS/SSL layer from
setting TLS/SSL options that they do not support. This also gets
tlsverifyclient closer to a working state by requesting the
client certificate when tlsverifyclient is set. Currently, there
is no consumer of main/tcptls.c in Asterisk that supports this
feature and so it can not be properly tested. Review:
https://reviewboard.asterisk.org/r/2370/ Reported-by: John
Bigelow Patch-by: Kinsey Moore (closes issue AST-1093)
2013-03-15 01:32 +0000 [r383120-383124] Matthew Jordan <mjordan@digium.com>
* channels/chan_sip.c: When a session timer expires during a T.38
call, re-invite with correct SDP When a session timer expires
during a dialog that has re-negotiated to T.38 and Asterisk is
the refresher, Asterisk will send a re-INVITE with an SDP
containing audio media only. This causes some hilarity with the
poor fax session under weigh. This patch corrects that by sending
T.38 parameters if we are in the middle of a T.38 session.
(closes issue ASTERISK-21232) Reported by: Nitesh Bansal patches:
dont-send-audio-reinvite-for-sess-timer-in-t38-call.patch
uploaded by nbansal (License 6418)
* pbx/pbx_spool.c: Fix processing of call files when using KQueue
on OS X In certain situations, call files are not processed when
using KQueue with pbx_spool. Asterisk was sending an invalid
timeout value when the spool directory is empty, causing the call
to kevent to error immediately. This can create a tight loop,
increasing the CPU load on the system. (closes issue
ASTERISK-21176) Reported by: Carlton O'Riley patches:
kqueue_osx.patch uploaded by coriley (License 6473)
2013-03-14 16:56 +0000 [r383061] Jason Parker <jparker@digium.com>
* autoconf/ast_ext_lib.m4: Fix whitespace in AST_EXT_LIB_CHECK
macro.
2013-03-12 21:15 +0000 [r382939-382942] Michael L. Young <elgueromexicano@gmail.com>
* addons/res_config_mysql.c: Fix Sorting Order For Parking Lots
Stored In Static Realtime When retrieving the parking lots from a
MySQL database table, the current order is "filename, cat_metric
desc, var_metric asc, category". If there are multiple parking
lots with the same cat_metric but different categories,
everything is being sorted on cat_metric first resulting in
errors when loading the parking lots. This patch fixes the
problem by sorting on the category field first, then the
cat_metric field. (closes issue ASTERISK-21035) Reported by: Alex
Epshteyn Patches: asterisk-21035-orderby.diff Michael L. Young
(license 5026)
* contrib/realtime/postgresql/realtime.sql,
contrib/realtime/mysql/sippeers.sql: Update Contributed Realtime
Schema Files - IPv6 Addresses This commit updates some fields in
the contributed realtime schema files to handle IPv6 addresses.
(closes issue ASTERISK-21173) Reported by: Torrey Searle Patches:
realtime_sql.patch Torrey Searle (license 5334)
asterisk-21173-update-ip-fields.diff Michael L. Young (license
5026)
2013-03-12 16:20 +0000 [r382847] Matthew Jordan <mjordan@digium.com>
* UPGRADE.txt, channels/chan_sip.c: Include the Username field in
SIP Registry events when Status is registered In ASTERISK-17888,
the AMI Registry event during SIP registrations was supposed to
include the Username field. Somehow, one of the events was
missed. This patch corrects that - the Username field should be
included in all AMI Registry events involving SIP registrations.
(issue ASTERISK-17888) (closes issue ASTERISK-21201) Reported by:
Dmitriy Serov patches: chan_sip.c.diff uploaded by Dmitriy Serov
(license 6479)
2013-03-06 18:22 +0000 [r382513] Kinsey Moore <kmoore@digium.com>
* apps/app_page.c: Correct app_page documentation The 'A' and 'n'
options for Page() mention that the announcement will be played
simultaneously. This is not necessarily the case.
2013-03-05 03:46 +0000 [r382409] Igor Goncharovskiy <igor.goncharovsky@gmail.com>
* channels/chan_unistim.c: Fix several unreleased mutex locks that
cause problem with processing calls Reported by: Daniel Bohling
Tested by: Daniel Bohling (Closes issue ASTERISK-21119)
2013-02-28 17:09 +0000 [r382227-382233] Matthew Jordan <mjordan@digium.com>
* channels/chan_iax2.c: Prevent deadlock in chan_iax2 when
attempting to set caller ID A deadlock can occur in chan_iax2
when it attempts to set the caller ID, as it already holds the
iax2 private lock and improperly fails to obtain the channel lock
before calling ast_set_callerid. By not safely obtaining the
channel lock, a locking inversion can take place, causing a
deadlock. This patch solves this by calling the required deadlock
avoidance functions that obtain the channel lock before setting
the caller ID. Thanks to Pavel for fixing my syntax errors and
testing this patch out. (closes issue ASTERISK-21128) Reported
by: Pavel Troller Tested by: Pavel Troller patches:
ASTERISK-21128-1.8.diff uploaded by mjordan (license 6283)
ASTERISK-21128-modified-1.8.diff uploaded by Pavel Troller
(license 6302)
* UPGRADE.txt, apps/app_meetme.c: Let channels joining a MeetMe
conference opt out of the denoiser For some channel drivers,
specifically those that have a varying rate in the number of
audio samples, the audio quality for a MeetMe conference can be
exceedingly poor. This is due to a unilateral application of the
DENOISE function in func_speex to channels joining the
conference. The denoiser function in the speex library is
initialized with the number of audio samples in each sample that
will be provided to it. If the number of audio samples changes,
the denoiser has to be thrown away and re-initialized. While this
could be worked around by removing func_speex, that doesn't help
if you actually use the denoiser with other channels on the
system. This patches does the following: * Checks for the
presence of func_speex as opposed to codec_speex when determining
if the DENOISE function is present (which is where the function
is actually implemented) * Adds an option to MeetMe 'n' that
causes the denoiser to not be applied to a channel when it joins.
This keeps the current behavior the default, but let's users
disable the denoiser if it causes problems on their system.
Review: https://reviewboard.asterisk.org/r/2358 (closes issue
AST-1062) Reported by: Thomas Arimont
2013-02-27 16:16 +0000 [r382153-382171] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: Relax dialog checking in
get_sip_pvt_byid_locked so it works when the dialog is forked.
(closes issue ASTERISK-20638) Reported by: eelcob Patches:
pedantic-call-pickup-from-tag.patch uploaded by eelcob (license
6442)
* configure, include/asterisk/autoconfig.h.in: Fix the configure
script over here as well.
2013-02-26 19:37 +0000 [r382110] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
* configure, configure.ac: Consider linux-gnuspe as linux-gnu * The
powerpcspe Linux port uses linux-gnuspe as the OS string. * Our
build system shouldn't really care for that, so just call it
linux-gnu. * Original report: Roland Stigge ,
http://bugs.debian.org/701505 Review:
https://reviewboard.asterisk.org/r/2357/
2013-02-26 19:30 +0000 [r382107] Walter Doekes <walter+asterisk@wjd.nu>
* channels/chan_sip.c: Correct RPID parsing for unquoted
display-name. Parsing Remote-Party-ID will now succeed if
display-name is of the *(token LWS) kind and not just the
quoted-string kind. Review:
https://reviewboard.asterisk.org/r/2341/
2013-02-26 19:06 +0000 [r382087] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
* main/Makefile: Remove unneeded linux-gnueabi* As of r380520 the
configure scripts converts the value of linux-gnueabi* of OSARCH
to "linux-gnu". So no point in testing for those values.
2013-02-25 12:48 +0000 [r381916-382021] Matthew Jordan <mjordan@digium.com>
* addons/res_config_mysql.c: Clean up use of va_end/va_args in
res_config_mysql There were several problems using variadic
argument macros in res_config_mysql. * Improper use of va_end.
Multiple calls to va_end were possible resulting in an unbalanced
matching of va_start/va_end. * Calls to va_arg after a possible
encounter of a SENTINEL value. This patch corrects those errors.
(closes issue ASTERISK-19451) Reported by: wdoekes patches:
ASTERISK-19451-1.8--2.diff uploaded by wdoekes (License 5674)
* channels/chan_jingle.c: Set the sin_family on the bind address
socket during initialization Somehow, chan_jingle has managed to
operate for years without setting the sin_family on its bindaddr
socket. This patch properly sets the field during initial module
load to AF_INET. Note that the patch on the issue was modified
slightly to change the initialization of the socket from
allocation of a chan_jingle private to the module initialization,
as the bindaddr object (which is static) only needs to have the
address set once. (closes issue ASTERISK-19341) Reported by:
andre valentin patches: 0105-chan_jingle.patch uploaded by
avalentin (License 6064)
* main/manager.c: Don't display the AMI ALL class authorization for
users if they don't have it When converting AMI class
authorizations to a string representation, the method always
appends the ALL class authorization. This is especially important
for events, as they should always communicate that class
authorization - even if the event itself does not specify ALL as
a class authorization for itself. (Events have always assumed
that the ALL class authorization is implied when they are raised)
Unfortunately, this did mean that specifying a user with
restricted class authorizations would show up in the 'manager
show user' CLI command as having the ALL class authorization.
Rather then modifying the existing string manipulation function,
this patch adds a function that will only return a string if the
field being compared explicitly matches class authorization field
it is being compared against. This prevents ALL from being
returned unless it is actually specified for the user. (closes
issue ASTERISK-20397) Reported by: Johan Wilfer
* apps/app_parkandannounce.c: Make ParkAndAnnounce return to
priority + 1 when return context is not defined The
ParkAndAnnounce application documentation for the optional
return_context parameter states the following: return_context The
goto-style label to jump the call back into after timeout.
Default 'priority+1'. Unfortunately, the application was sending
the channel back into the dialplan at 'priority', which is the
ParkAndAnnounce application call. This causes an infinite loop of
the channel constantly being parked, announced, timed out,
parked, announced, timed out... while fun, especially for those
callers you wish to drive to the end of madness, this was not the
intent of the application. (closes issue ASTERISK-20113) Reported
by: serginuez patches: app_parkandannounce.diff uploaded by
serginuez (License 6405)
2013-02-21 22:44 +0000 [r381847] Matthew Jordan <mjordan@digium.com>
* configure, configure.ac: Properly detect launchd Asterisk was a
little too pro-active in claiming that it found launchd. On
systems without launchd - such as FreeBSD - this resulted in
certain items in Asterisk that conflict with launchd to not be
selectable, such as res_timing_kqueue. (closes issue
ASTERISK-20749) Reported by: Oleg Baranov
2013-02-19 19:16 +0000 [r381770] kharwell <kharwell@localhost>:
* main/features.c: Write the correct callid to the data1 field in
queue_log for transfer events. The incorrect callid was being
written to the "data1" field in queue_log table for transfer
events. The callid of the queue was being written instead of the
transfer target's callid. This now gets the correct "transfer to"
number and places that in the "data1" field of the queue_log
table when a transfer event is triggered. (closes issue
ASTERISK-19960) Reported by: vladimir shmagin
2013-02-18 20:28 +0000 [r381668] Walter Doekes <walter+asterisk@wjd.nu>
* configs/sip.conf.sample: Remove "registertrying" and add
"rtp_engine" from/to sip.conf.sample The "registertrying" option
was removed in r343220. The "rtp_engine" option was added in
r186078 but erroneously named "engine" in the sample. Note that
there is no global sip setting for a different engine.
2013-02-14 19:41 +0000 [r381466] Richard Mudgett <rmudgett@digium.com>
* main/features.c: End stuck DTMF if AST_SOFTHANGUP_ASYNCGOTO
because it isn't a real hangup. It doesn't hurt to check
AST_SOFTHANGUP_UNBRIDGE either, but it should not be set outside
of a bridge. (issue ASTERISK-20492)
2013-02-14 03:42 +0000 [r381364] Matthew Jordan <mjordan@digium.com>
* apps/app_db.c: Don't throw a spurious error when using DBdeltree
The function call ast_db_deltree returns the number of row
deleted, or a negative number if it failed. DBdeltree was
treating any non-zero return as an error, causing a spurious
verbose error message to be displayed. This patch handles the
return code of ast_db_deltree correctly. (closes issue
ASTERISK-21070) Reported by: ianc patches: dbdeltree.diff
uploaded by ianc (License #5955)
2013-02-12 20:16 +0000 [r381281] Mark Michelson <mmichelson@digium.com>
* main/rtp_engine.c: Do not allow native RTP bridging if
packetization of media streams differs. The RTP engine will no
longer allow for local and remote native RTP bridges if
packetization of streams differs. Allowing native bridging in
this scenario has been known to cause FAX failures. (closes
ASTERISK-20650) Reported by: Maciej Krajewski Patches:
ASTERISK-20659.patch uploaded by Mark Michelson (License #5049)
Review: https://reviewboard.asterisk.org/r/2319
2013-02-11 20:46 +0000 [r381216] kharwell <kharwell@localhost>:
* apps/app_playback.c: Properly load say.conf upon reload of module
app_playback. If say.conf did not exists prior to originally
loading module app_playback it would not load on subsequent
reloads of the module once it had been created. This occurred
because upon reload of the app_playback module it would only load
a new configuration if an old one had previously existed. This
fix simply removed the association between checking if an old
configuration existed and the loading of the new one. (closes
issue ASTERISK-20800) Reported by: pgoergler
2013-02-06 20:10 +0000 [r380973] David M. Lee <dlee@digium.com>
* channels/chan_sip.c: Fixed failing test from r380696. When I
added my extensive suite of session timer unit tests, apparently
one of them was failing and I never noticed. If neither Min-SE
nor Session-Expires is set in the header, it was responding with
a Session-Expires of the global maxmimum instead of the
configured max for the endpoint. (issue ASTERISK-20787)
2013-02-05 18:09 +0000 [r380853] Richard Mudgett <rmudgett@digium.com>
* main/dial.c: Separate option_types[] from the struct definition.
Updated the option_types[] doxygen comment.
2013-01-31 19:56 +0000 [r380696] David M. Lee <dlee@digium.com>
* channels/chan_sip.c: Process session timers, even if
Session-Expires header is missing Previously, Asterisk only
processed session timer information if both the 'Supported:
timer' and 'Session-Expires' headers were present. However, the
Session-Expires header is optional. If we were to receive a
request with a Min-SE greater than our configured
session-expires, we would respond with a 'Session-Expires' header
that was too small. This patch cleans the situation up a bit,
always processing timer information if the 'Supported: timer'
header is present. (closes issue ASTERISK-20787) Reported by:
Mark Michelson Review: https://reviewboard.asterisk.org/r/2299/
2013-01-31 00:22 +0000 [r380572-380611] Richard Mudgett <rmudgett@digium.com>
* include/asterisk/channel.h: Make CHECK_BLOCKING() debug message
more useful. Change the displayed pthread value to hex format so
it can be easily matched with CLI core show threads or gdb.
* channels/chan_dahdi.c: chan_dahdi: Fix "dahdi show channels
group" for groups greater than 31. The variable type used was not
large enough to hold a group bit field.
2013-03-27 Asterisk Development Team <asteriskteam@digium.com>
* Asterisk 1.8.21.0-rc2 Released.
* Do not allow native RTP bridging if packetization of media streams
differs.
The RTP engine will no longer allow for local and remote native RTP
bridges if packetization of streams differs. Allowing native bridging
in this scenario has been known to cause FAX failures.
* Resolve deadlock between pending CDR and batch CDR locks
r375757 attempted to resolve a race condition between multiple
submissions of CDRs while in batch mode from attempting to destroy the
scheduled batch submission by extending the batch CDR lock. Unfortunately,
this causes a deadlock between the pending CDR lock and the batch CDR lock.
This patch resolves the intent of r375757 by simply providing a new lock
that protects the scheduling of the batches. The original batch CDR lock
is kept to protect manipulation of the batch CDR settings, but has been
placed such that it is not held when the pending lock is held.
Thanks to Chase Venters for providing lock analysis on the issue.
* Resolve deadlock between SIP registration and channel based
functions
In r373424, several reentrancy problems in chan_sip were addressed. As
a result, the SIP channel driver is now properly locking the channel
driver private information in certain operations that it wasn't previously.
This exposed two latent problems either in register_verify or by functions
called by register_verify. This includes:
* Holding the private lock while calling sip_send_mwi_to_peer. This
can create a new sip_pvt via sip_alloc, which will obtain the channel
container lock. This is a locking inversion, as any channel related lock
must be obtained prior to obtaining the SIP channel technology private
lock.
* Holding the private lock while calling sip_poke_peer. In the same vein as
sip_send_mwi_to_peer, sip_poke_peer can create a new SIP private, causing
the same locking inversion.
Note that this locking inversion typically occured when CLI commands were run
while a SIP REGISTER request was being processed, as many CLI commands (such
as 'sip show channels', 'core show channels', etc.) have to obtain the channel
container lock.
* AST-2013-002: Prevent denial of service in HTTP server
AST-2012-014, fixed in January of this year, contained a fix for
Asterisk's HTTP server for a remotely-triggered crash. While the fix put in
place fixed the possibility for the crash to be triggered, a denial of
service vector still exists with that solution if an attacker sends one or
more HTTP POST requests with very large Content-Length values. This patch
resolves this by capping the Content-Length at 1024 bytes. Any attempt to send
an HTTP POST with Content-Length greater than this cap will not result in any
memory allocation. The POST will be responded to with an HTTP 413 "Request
Entity Too Large" response.
This issue was reported by Christoph Hebeisen of TELUS Security Labs
* AST-2013-003: Prevent username disclosure in SIP channel driver
When authenticating a SIP request with alwaysauthreject enabled,
allowguest disabled, and autocreatepeer disabled, Asterisk discloses whether
a user exists for INVITE, SUBSCRIBE, and REGISTER transactions in
multiple ways. The information is disclosed when:
* A "407 Proxy Authentication Required" response is sent instead of a
"401 Unauthorized" response
* The presence or absence of additional tags occurs at the end of
"403 Forbidden" (such as "(Bad Auth)")
* A "401 Unauthorized" response is sent instead of "403 Forbidden"
response after a retransmission
* Retransmission are sent when a matching peer did not exist, but not
when a matching peer did exist.
This patch resolves these various vectors by ensuring that the responses sent
in all scenarios is the same, regardless of the presence of a matching peer.
This issue was reported by Walter Doekes, OSSO B.V. A substantial portion of
the testing and the solution to this problem was done by Walter as well - a
huge thanks to his tireless efforts in finding all the ways in which this
setting didn't work, providing automated tests, and working with Kinsey on
getting this fixed.
* Fix white noise on SRTP decryption
When res_rtp_asterisk.c was altered to avoid attempting to apply
unprotect algorithms to non-audio RTP packets, the test used was
incorrect. This caused the audio packets to not be decrypted and
resulted in loud white noise on the other endpoint (or both endpoints
depending on the call legs involved). The test now properly checks the
version field in the RTP header to ensure that RTP and RTCP are
decrypted while other types of packets are not.
2013-01-30 Asterisk Development Team <asteriskteam@digium.com>
* Asterisk 1.8.21.0-rc1 Released.
2013-01-30 17:44 +0000 [r380451-380520] Matthew Jordan <mjordan@digium.com>
* configure, configure.ac: Support building Asterisk for Raspberry
Pi/Raspbian with hard-float support Building Asterisk on Raspbian
with hard-float support fails as it uses the string
'linux-gnueabihf' for host os, as opposed to 'linux-gnueabi'.
This patch modifies the configure script for Asterisk such that
it will match on any string beginning with 'linux-gnueabi', as
opposed to requiring an explicit match. (closes issue
ASTERISK-21006) Reported by: Christian Hesse Tested by: Christian
Hesse patches: linux-gnueabihf.patch uploaded by Christian Hesse
(license 6459) linux-gnueabihf-autoconf.patch uploaded by
Christian Hesse (license 6459)
* channels/chan_sip.c: Perform case insensitive comparisons for
T.38 attributes RFC5347 section 2.5.2 states the following: ...
The attribute "T38MaxBitRate" was once incorrectly registered
with IANA as "T38maxBitRate" (lower-case "m"). In accordance with
T.38 examples and common implementation practice, the form
"T38MaxBitRate" SHOULD be generated by implementations conforming
to this package. In general, it is RECOMMENDED that
implementations of this package accept lowercase, uppercase, and
mixed upper/lowercase encodings of all the T.38 attributes. ...
Asterisk currently does not perform case insensitive matching on
the T.38 attributes. This causes the T38MaxBitRate attribute to
be negotiated at 2400 baud instead of 14400 (or whatever value
you actually wanted). This patch makes it so that when we compare
T.38 attributes, we do so in a case insensitive fashion. Note
that while the issue reporter did not directly write the patch,
they contributed to it (and would have provided one themselves if
the license had gone through a tad faster), and hence get
attribution for it. (closes issue ASTERISK-20897) Reported by:
Eric Hill Tested by: Eric Hill patches: -- uploaded by Eric Hill
* res/res_calendar_icalendar.c: Fix memory leak in
res_calendar_icalendar The ICalendar module had a systemic memory
leak on each fetch of data from the ICalendar source. The
previous fetched data was not being properly disposed. This patch
makes it so that before each fetch of data, we dispose of the
previously fetched data. (closes issue ASTERISK-21012) Reported
by: Joel Vandal Tested by: Joel Vandal
2013-01-29 17:22 +0000 [r380364] Richard Mudgett <rmudgett@digium.com>
* channels/chan_agent.c: chan_agent: Prevent multiple channels from
logging in as the same agent. Multiple channels logging in as the
same agent can result in dead channels waiting for a condition
signal that will never come because another channel thread stole
it. A symptom is chan_sip repeatedly generating warning messages
about rescheduling autodestruction of dialogs with an agent
channel owner. * Made only login_exec() (the app AgentLogin)
clear the agent_pvt->chan pointer to prevent multiple channels
from logging in as the same agent. agent_read(), agent_call(),
and agent_set_base_channel() no longer disconnect the agent
channel from the agent_pvt. This also eliminates the need to keep
checking for agent_pvt->chan being NULL. * Made agent_hangup()
not wake up the AgentLogin agent thread until it is done. * Made
agent_request() not able to get the agent until he has logged in
and any wrapup time has expired. * Made agent_request() use
ast_hangup() instead of agent_hangup() to correctly dispose of a
channel. * Removed agent_set_base_channel(). Nobody calls it and
it is a bad thing in general. * Made only agent_devicestate()
determine the current device state of an agent. Note: Agent group
device states have never been supported. Review:
https://reviewboard.asterisk.org/r/2260/
2013-01-29 17:05 +0000 [r380347] David M. Lee <dlee@digium.com>
* channels/sip/sdp_crypto.c: Corrected crypto tag in SDP ANSWER for
SRTP. (again) The original fix (r380043) for getting Asterisk to
respond with the correct tag overlooked some corner cases, and
the fact that the same code is in 1.8. This patch moves the
building of the crypto line out of sdp_crypto_process(). Instead,
it merely copies the accepted tag. The call to sdp_crypto_offer()
will build the crypto line in all cases now, using a tag of "1"
in the case of sending offers. (closes issue ASTERISK-20849)
Reported by: José Luis Millán Review:
https://reviewboard.asterisk.org/r/2295/
2013-01-29 02:02 +0000 [r380297] Matthew Jordan <mjordan@digium.com>
* autoconf/ast_check_pwlib.m4, configure: Update configure script
to be compatible with ptlib 2.10.9 With ptlib 2.10.9, the
configure script fails due to grep returning multiple matches for
the pattern it searches for. This patch updates the pattern
matching to return only the actual version for the symbol
searched for, PTLIB_VERSION. (closes issue ASTERISK-20980)
Reported by: Stefan Reuter patches: ASTERISK-20980-1.patch
uploaded by Stefan Reuter (license 5339)
2013-01-28 21:06 +0000 [r380254] Sean Bright <sean@malleable.com>
* channels/chan_iax2.c, channels/iax2.h: Correct the number of
available call numbers in IAX2. There is currently an edge case
where call number 32768 might be allocated for a call, even
though the IAX2 protocol requires call numbers be only 15 bits.
This resulted in some unpredictable behavior when call number
32678 is chosen. This patch was mostly written by Richard Mudgett
via ReviewBoard. I'm just committing it. Review:
https://reviewboard.asterisk.org/r/2293/
2013-01-28 01:52 +0000 [r380210] Russell Bryant <russell@russellbryant.com>
* main/file.c: Change cleanup ordering in filestream destructor.
This patch came about due to a problem observed where wav files
had an empty header. The header is supposed to be updated in
wav_close(). It turns out that this was broken when the
cache_record_files option from asterisk.conf was enabled. The
cleanup code was moving the file to its final destination
*before* running the close() method of the file destructor, so
the header didn't get updated. Another problem here is that the
move was being done before actually closing the FILE *. Finally,
the last bug fixed here is that I noticed that wav_close() checks
for stream->filename to be non-NULL. In the previous cleanup
order, it's checking a pointer to freed memory. This doesn't
actually cause anything to break, but it's treading on dangerous
waters. Now the free() of stream->filename is happening after the
format module's close() method gets called, so it's safer.
Review: https://reviewboard.asterisk.org/r/2286/
2013-01-23 00:19 +0000 [r379963] Richard Mudgett <rmudgett@digium.com>
* main/astobj2.c: Attempt to be more helpful when using a bad ao2
object pointer. Backport of -r360626 with some enhancements. Put
the external obj pointer in the message instead of the internal
version.
2013-01-22 18:21 +0000 [r379885] Jonathan Rose <jrose@digium.com>
* sounds/Makefile, apps/app_meetme.c: app_meetme: Use new prompts
for administrator menu The old prompts for the administrator menu
were inadequate. They didn't mention that the menu had additional
options through the 8 key and pressing the 8 key wouldn't reveal
what those options were. This patch fixes all of that while also
organizing code pertaining to each individual menu type which was
previously all stored in one gigantic function along with many of
the basic conference functions. (closes issue AST-996) Reported
by: John Bigelow Review:
http://reviewboard.digium.internal/r/360/
2013-01-22 14:43 +0000 [r379760-379825] Matthew Jordan <mjordan@digium.com>
* apps/app_meetme.c: Fix station ringback; trunk hangup issues in
SLA This patch fixes two bugs: * If an outbound call is made from
a SLA phone using SLAStation, then there is no ringtone audible
to the phone that originates the call. The indication of the
ringing was not being passed to the SLA station; this patch fixes
that by passing through the progress indications. * If an SLA
station hangs up before the called party answers, then the
channel to the called party continues to ring until a timeout
occurs. If the called party manages to answer, Asterisk attempts
to connect the called party to a non-existant MeetMe room. This
patch corrects the behavior by abandoning the call attempt if it
detects that the SLA station is no longer in use while attempting
to call the called party. Review:
https://reviewboard.asterisk.org/r/2275/ (closes issue
ASTERISK-20462) Reported by: dkerr patches:
asterisk-11-bugid20440+20462.patch uploaded by dkerr (license
5558) asterisk-11-bugid20462.patch uploaded by dkerr (license
5558) (closes issue ASTERISK-20440) Reported by: dkerr patches:
asterisk-11-bugid20440.patch uploaded by dkerr (license 5558)
asterisk-11-bugid20440+20462.patch uploaded by dkerr (license
5558)
* UPGRADE.txt, contrib/init.d/rc.gentoo.asterisk,
contrib/init.d/rc.slackware.asterisk,
contrib/init.d/rc.archlinux.asterisk,
contrib/scripts/safe_asterisk, main/asterisk.c,
contrib/init.d/rc.suse.asterisk,
contrib/init.d/rc.mandriva.asterisk,
contrib/init.d/rc.debian.asterisk,
contrib/init.d/rc.redhat.asterisk: Update init.d scripts to
handle stderr; readd splash screen for remote consoles When
r376428 was commited to re-order start up sequences to be more
tolerant of forking with thread primitives, a few items were
changed that caused changes in behavior on some distros. This
includes: * Not displaying the splash screen on a remote console.
* Displaying an error message on stderr when a remote console
cannot connect to a running instance of Asterisk. In the first
case, the splash screen was re-added (thanks to Michael L.
Young). In the second case, the various init.d scripts were
modified to pipe stderr to /dev/null, as the error message is
useful - if you execute a remote console or a remote console
command execution and it fail, it should tell you. Note that the
error message was always present, it just failed to be printed
prior to r376428. Much thanks to the folks who quickly reported
this problem, provided solutions, and promptly tested the various
init.d scripts on a variety of distros. (closes issue
ASTERISK-20945) Reported by: Warren Selby Tested by: Michael L.
Young, Jamuel Starkey, kaldemar, Danny Nicholas, mjordan patches:
asterisk-20945-remote-intro-msg.diff uploaded by elguero (license
5026) ASTERISK-20945-1.8-mjordan.diff uploaded by mjordan
(license 6283)
2013-01-21 18:27 +0000 [r379718] Kinsey Moore <kmoore@digium.com>
* codecs/codec_ilbc.c: Prevent segfault for interpolated iLBC
frames When iLBC is being used with a jitter buffer and the jb
has to interpolate frames, it generates frames with a null
pointer and a non-zero datalen. This is now handled properly.
(closes issue ASTERISK-20914) Reported By: John McEleney Patches:
ASTERISK-20914-1.8.diff uploaded by Matt Jordan (license 6283)
2013-01-21 04:59 +0000 [r379645] Andrew Latham <lathama@gmail.com>
* contrib/scripts/install_prereq: Add LDAP libraries to install
script Add LDAP dev package to Debian/Ubuntu install list.
Existed in Redhat already. Merged from 11 to Trunk in 379643.
Sorry for forgeting 1.8 (issue ASTERISK-20886)
2013-01-21 04:05 +0000 [r379608] Matthew Jordan <mjordan@digium.com>
* apps/app_minivm.c: Fix crash in app_minivm when mime encoding
string An incorrect string initializations was left in
ast_str_encode_mime from the patch that converted string
manipulations to use ast_str strings (r191140). The string
initialization causes a crash when ast_str_set is called on the
string later on in the function. (closes issue ASTERISK-18697)
Reported by: Chris Boot patches:
minivm-null-pointer-dereference-fix.patch uploaded by bootc
(license 6309) (issue ASTERISK-20854) Reported by: Chris Warr
Tested by: Chris Warr
2013-01-19 20:41 +0000 [r379547] Walter Doekes <walter+asterisk@wjd.nu>
* configure, include/asterisk/autoconfig.h.in,
include/asterisk/compat.h, main/strcompat.c, configure.ac: Add
builtin roundf() for systems lacking it. (closes issue
ASTERISK-16854) Review: https://reviewboard.asterisk.org/r/2276
Reported-by: Ovidiu Sas
2013-01-18 23:26 +0000 [r379509] Matthew Jordan <mjordan@digium.com>
* main/asterisk.c: Fix astcanary startup problem due to wrong pid
value from before daemon call When Asterisk forks itself into the
background via a call to daemon, it must re-set the pid value of
the new process. Otherwise, astcanary gets the pid value of the
process before the fork, which prevents it from running. Asterisk
eventually starts lowering its priority, as it can no longer
communicate with the proverbial canary in the coal mine. This
patch ensures that the correct process identifier is used by
astcanary. (closes issue ASTERISK-20947) Reported by: Jakob
Hirsch Tested by: mjordan patches:
asterisk-10.12.0.astcanary_ppid.diff uploaded by Jakob Hirsch
(license 6113)
2013-01-18 05:23 +0000 [r379392] David M. Lee <dlee@digium.com>
* channels/sip/include/reqresp_parser.h, channels/chan_sip.c,
channels/sip/reqresp_parser.c: Fix Record-Route parsing for large
headers. Record-Route parsing copied the header into a char[256]
array, which can be a problem if the header is longer than that.
This patch parses the header in place, without the copy, avoiding
the issue. In addition to the original patch, I added a unit test
for the new get_in_brackets_const function. (closes issue
ASTERISK-20837) Reported by: Corey Farrell Patches:
chan_sip-build_route-optimized-rev1.patch uploaded by Corey
Farrell (license 5909) (with minor changes by dlee)
2013-01-17 02:28 +0000 [r379342] Matthew Jordan <mjordan@digium.com>
* addons/chan_mobile.c: Fix issue where chan_mobile fails to bind
to first available port Per the bluez API, in order to bind to
the first available port, the rc_channel field of the socket
addressing structure used to bind the socket should be set to 0.
Previously, Asterisk had set the rc_channel field set to 1,
causing it to connect to whatever happens to be on port 1. We
could probably not explicitly set rc_channel to 0 since we memset
the struct earlier, but explicitly setting it will hopefully
prevent someone from coming in and setting it to some explicit
port in the future. (closes issue ASTERISK-16357) Reported by:
challado Tested by: Alexander Heinz, Nikolay Ilduganov, benjamin,
eliafino, David van Geyn patches: ASTERISK-16357.diff uploaded by
Nikolay Ilduganov (license 6253)
2013-01-16 22:45 +0000 [r379310] Mark Michelson <mmichelson@digium.com>
* main/manager.c: Further fix misinformation in the description of
manager MailboxStatus command. The description still claimed that
it returned the number of messages rather than whether there were
messages waiting.
2013-01-16 21:12 +0000 [r379276] Jason Parker <jparker@digium.com>
* contrib/scripts/install_prereq: Reduce number of packages
install_prereq installs on Debian systems. 'search' will look for
any package containing the name provided, so we need to force a
more exact search.
2013-01-16 17:40 +0000 [r379226] Richard Mudgett <rmudgett@digium.com>
* channels/chan_misdn.c: chan_misdn: Fix compile error. (issue
ASTERISK-15456)
2013-01-16 04:10 +0000 [r379091-379178] Matthew Jordan <mjordan@digium.com>
* addons/chan_mobile.c: Fix parsing SMSSRC for SMS messages The
parser for SMS messages would incorrectly parse out the from
number. The parsing would incorrectly start scanning for the from
number at the same index as the first double quote ("); this
would inadvertently cause it to treat the first double quote as
the terminating double quote for the from number as well. The
SMSSRC should now populate correctly. (closes issue
ASTERISK-16822) Reported by: menschentier Tested by: Jonas Falck
patches: fixSMSSRC.patch uploaded by jonax (license 6320) (closes
issue ASTERISK-19153) Reported by: Panos Gkikakis patches:
sms-sender-fix.diff uploaded by roeften (license 5884)
* channels/chan_misdn.c: Set the INVALID_EXTEN channel variable
when chan_misdn forces the 'i' extension The chan_misdn channel
driver will send a channel with an invalid destination to the 'i'
extension itself if said extension can be reached. It forgot,
however, to set the INVALID_EXTEN channel variable when it
bounces the channel to this extension. Dialplan writers
everywhere moaned at yet another inconsistency. This is yet
another example of why duplicating logic in multiple places
results in bugs that stick around in Jira for just under three
years. Yes: ASTERISK-15456 was created on January 18th, 2010.
Patch committed on January 15th, 2013. Ouch. (closes issue
ASTERISK-15456) Reported by: Thomas Omerzu patches:
chan_misdn_invalid.patch2 uploaded by Thomas Omerzu (license
5927)
* bridges/bridge_softmix.c: Prevent crash in ConfBridge due to race
condition when channels leave bridge When a channel leaves a
bridge, a race condition existed where the bridge_channel's pvt
structure would be accessed after it was disposed of. This patch
prevents that by setting the pointer to the pvt to NULL prior to
disposing of it. Note that this patch is a backport from Asterisk
10. This particular race condition was fixed as part of the
larger code rework that occurred for that release. The solution
to this problem was pointed out by Gunnar Harms in
ASTERISK-16640. (closes issue ASTERISK-16640) Reported by:
thomas987 (closes issue ASTERISK-16835) Reported by: saghul
2013-01-14 15:11 +0000 [r379001] David M. Lee <dlee@digium.com>
* channels/chan_sip.c: Fix XML encoding of 'identity display' in
NOTIFY messages, continued. When r378933 was merged into 1.8, it
should have also escaped remote_display, since it will have the
same XML encoding problem when the caller/callee roles are
reversed. (closes issue ABE-2902) Reported by: Guenther Kelleter
2013-01-13 21:15 +0000 [r378967] Matthew Jordan <mjordan@digium.com>
* res/res_rtp_asterisk.c: Reset RTP timestamp; sequence number on
SSRC change In r370252 for ASTERISK-18404, Asterisk's handling of
RTP was modified to better account for out of order RTP packets.
This was accomplished by using the RTP timestamp and sequence
number to check for out of order packets. However, when a SSRC
change occurs, the timestamp and sequence number will no longer
have any relation to the previously received packets. The
variables tracking the timestamp and sequence number therefore
have to be reset. (closes issue ASTERISK-20906) Reported by:
Eelco Brolman patches: dtmf_on_hold.patch uploaded by Eelco
Brolman (license #6442)
2013-01-12 06:26 +0000 [r378933] David M. Lee <dlee@digium.com>
* main/utils.c, include/asterisk/utils.h, channels/chan_sip.c,
tests/test_xml_escape.c (added): Fix XML encoding of 'identity
display' in NOTIFY messages. XML encoding in chan_sip is
accomplished by naively building the XML directly from strings.
While this usually works, it fails to take into account escaping
the reserved characters in XML. This patch adds an
'ast_xml_escape' function, which works similarly to
'ast_uri_encode'. This is used to properly escape the
local_display attribute in XML formatted NOTIFY messages. Several
things to note: * The Right Thing(TM) to do would probably be to
replace the ast_build_string stuff with building an ast_xml_doc.
That's a much bigger change, and out of scope for the original
ticket, so I refrained myself. * It is with great sadness that I
wrote my own ast_xml_escape function. There's one in libxml2, but
it's knee-deep in libxml2-ness, and not easily used to one-off
escape a string. * I only escaped the string we know is causing
problems (local_display). At least some of the other strings are
URI-encoded, which should be XML safe. Rather than figuring out
what's safe and escaping what's not, it would be much cleaner to
simply build an ast_xml_doc for the messages and let the XML
library do the XML escaping. Like I said, that's out of scope.
(closes issue ABE-2902) Reported by: Guenther Kelleter Tested by:
Guenther Kelleter Review:
http://reviewboard.digium.internal/r/365/ ........ Merged
revision 378919 from
https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
2013-01-09 20:26 +0000 [r378733-378776] David M. Lee <dlee@digium.com>
* main/rtp_engine.c: Fix end condition in
ast_rtp_lookup_mime_multiple2. The erroneous end condition would
never include the AST_RTP_CISCO_DTMF flag in the debug output.
(closes issue ASTERISK-20772) Reported by: Xavier Hienne
* include/asterisk/causes.h: Replace errant tabs with spaces in
causes.h. (closes issue ASTERISK-20826) Reported by: snuffy
Patches: notabs.dif uploaded by snuffy (license 5024)
2013-01-08 20:22 +0000 [r378663] Richard Mudgett <rmudgett@digium.com>
* apps/app_queue.c: app_queue: Fix multiple calls to a queue member
that is in only one queue. When ringinuse=no queue members can
receive more than one call if these calls happen at nearly the
same time. * Fix so a queue member does not receive more than one
call from a queue. NOTE: This fix does not prevent multiple calls
to a member if the member is in more than one queue. * Did some
refactoring to eliminate some code redundancy. (issue
ASTERISK-16115) Reported by: nik600 Patches:
jira_asterisk_16115_single_q_v1.8.patch (license #5621) patch
uploaded by rmudgett Modified
2013-01-04 22:54 +0000 [r378591] Jonathan Rose <jrose@digium.com>
* res/res_srtp.c: res_srtp: Prevent a crash from occurring due to
srtp_create failures in srtp_create Under some circumstances,
libsrtp's srtp_create function deallocates memory that it wasn't
initially responsible for allocating. Because we weren't
initially aware of this behavior, this memory was still used in
spite of being unallocated during the course of the
srtp_unprotect function. A while back I made a patch which would
set this value to NULL, but that exposed a possible condition
where we would then try to check a member of the struct which
would cause a segfault. In order to address these problems,
ast_srtp_unprotect will now set an error value when it ends
without a valid SRTP session which will result in the caller of
srtp_unprotect observing this error and hanging up the relevant
channel instead of trying to keep using the invalid session
address. (closes issue ASTERISK-20499) Reported by: Tootai
Review:
https://reviewboard.asterisk.org/r/2228/diff/#index_header
2013-01-04 21:12 +0000 [r378554] Michael L. Young <elgueromexicano@gmail.com>
* channels/chan_sip.c: Fix SIP Notify Messages To Have The Proper
IP Address In The FROM Field On a multihomed server when sending
a NOTIFY message, we were not figuring out which network should
be used to contact the peer. This patch fixes the problem by
calling ast_sip_ouraddrfor() and then build_via() so that our
NOTIFY message contains the correct IP address. Also, a debug
message is being added to help follow the call-id changes that
occur. This was helpful for confirming that the IP address was
set properly since the call-id contains the IP address. It also
will be helpful for troubleshooting purposes when following a
call in the debug logs. (closes issue ASTERISK-20805) Reported
by: Bryan Hunt Tested by: Bryan Hunt, Michael L. Young Patches:
asterisk-20805-notify-ip-v2.diff uploaded by Michael L. Young
(license 5026) Review: https://reviewboard.asterisk.org/r/2255/
2013-01-04 21:12 +0000 [r378553] Joshua Colp <jcolp@digium.com>
* res/res_rtp_asterisk.c: Don't pass STUN packets through the SRTP
unprotect function. (closes issue AST-1036) Reported by: jbigelow
2013-01-03 22:09 +0000 [r378514] Michael L. Young <elgueromexicano@gmail.com>
* apps/app_queue.c: Fix Queue Log Reporting Every Call
COMPLETECALLER With "h" Extension Present When the "h" extension
is present within the context of the queue, all calls are being
reported COMPLETECALLER even when the agent is hanging up the
call. This patch checks to see if the agent hung-up or not
instead of only relying on checking if the queue (caller) channel
hung-up or not. It would appear that having the h extension in
the mix, the pbx goes to the h extension, "hanging-up" the queue
channel and triggering the reporting of COMPLETECALLER. (closes
issue ASTERISK-20743) Reported by: call Tested by: call, Michael
L. Young Patches: asterisk-20743-q-cmplt-caller.diff uploaded by
Michael L. Young (license 5026) Review:
https://reviewboard.asterisk.org/r/2256/
2013-01-03 19:40 +0000 [r378456-378486] Richard Mudgett <rmudgett@digium.com>
* channels/chan_agent.c: chan_agent: Fix wrapup time wait response.
* Made agent_cont_sleep() and agent_ack_sleep() stop waiting if
the wrapup time expires. agent_cont_sleep() had tried but
returned the wrong value to stop waiting. * Made
agent_ack_sleep() take a struct agent_pvt pointer instead of a
void pointer for better type safety.
* channels/chan_agent.c: chan_agent: Misc code cleanup. * Fix
off-nominal path resource cleanup in agent_request(). * Create
agent_pvt_destroy() to eliminate inlined versions in many places.
* Pull invariant code out of loop in add_agent(). * Remove
redundant module user references in login_exec(). * Remove unused
struct agent_pvt logincallerid[] member. * Remove some redundant
code in agent_request().
2013-01-03 18:35 +0000 [r378455] Kinsey Moore <kmoore@digium.com>
* main/channel.c: Add missing test event This test event was
missing from channel.c causing the dial_LS_options test to fail
intermittently because of a race condition where most code paths
emitted the test event but this one did not. The dial_LS_options
test should stop bouncing now.
2013-01-03 17:41 +0000 [r378427] Richard Mudgett <rmudgett@digium.com>
* channels/chan_agent.c: chan_agent: Fix agent_indicate() locking.
Avoid deadlock potential with local channels and simplify the
locking.
2013-01-02 21:48 +0000 [r378375] Matthew Jordan <mjordan@digium.com>
* main/config.c, funcs/func_realtime.c: Prevent crashes from
occurring when reading from data sources with large values When
reading configuration data from an Asterisk .conf file or when
pulling data from an Asterisk RealTime backend, Asterisk was
copying the data on the stack for manipulation. Unfortunately, it
is possible to read configuration data or realtime data from some
data source that provides a large blob of characters. This could
potentially cause a crash via a stack overflow. This patch
prevents large sets of data from being read from an ARA backend
or from an Asterisk conf file. (issue ASTERISK-20658) Reported
by: wdoekes Tested by: wdoekes, mmichelson patches: *
issueA20658_dont_process_overlong_config_lines.patch uploaded by
wdoekes (license 5674) * issueA20658_func_realtime_limit.patch
uploaded by wdoekes (license 5674)
2013-01-02 21:08 +0000 [r378356] Richard Mudgett <rmudgett@digium.com>
* include/asterisk/channel.h, main/manager.c, main/features.c: Fix
AMI redirect action with two channels failing to redirect both
channels. The AMI redirect action can fail to redirect two
channels that are bridged together. There is a race between the
AMI thread redirecting the two channels and the bridge thread
noticing that a channel is hungup from the redirects. * Made the
bridge wait for both channels to be redirected before exiting. *
Made the AMI redirect check that all required headers are present
before proceeding with the redirection. * Made the AMI redirect
require that any supplied ExtraChannel exist before proceeding.
Previously the code fell back to a single channel redirect
operation. (closes issue ASTERISK-18975) Reported by: Ben Klang
(closes issue ASTERISK-19948) Reported by: Brent Dalgleish
Patches: jira_asterisk_19948_v11.patch (license #5621) patch
uploaded by rmudgett Tested by: rmudgett, Thomas Sevestre, Deepak
Lohani, Kayode Review: https://reviewboard.asterisk.org/r/2243/
2013-01-02 16:54 +0000 [r378269-378303] Matthew Jordan <mjordan@digium.com>
* main/devicestate.c, include/asterisk/channel.h,
channels/chan_iax2.c, res/res_jabber.c, main/channel.c,
channels/chan_dahdi.c, include/asterisk/event_defs.h,
channels/chan_skinny.c, main/features.c, main/event.c,
apps/app_confbridge.c, funcs/func_devstate.c, res/res_calendar.c,
include/asterisk/devicestate.h, channels/chan_local.c,
apps/app_meetme.c, channels/chan_sip.c, channels/chan_agent.c:
Prevent exhaustion of system resources through exploitation of
event cache Asterisk maintains an internal cache for devices in
the event subsystem. The device state cache holds the state of
each device known to Asterisk, such that consumers of device
state information can query for the last known state for a
particular device, even if it is not part of an active call. The
concept of a device in Asterisk can include entities that do not
have a physical representation. One way that this occurred was
when anonymous calls are allowed in Asterisk. A device was
automatically created and stored in the cache for each anonymous
call that occurred; this was possible in the SIP and IAX2 channel
drivers and through channel drivers that utilized the
res_jabber/res_xmpp resource modules (Gtalk, Jingle, and Motif).
These devices are never removed from the system, allowing
anonymous calls to potentially exhaust a system's resources. This
patch changes the event cache subsystem and device state
management to no longer cache devices that are not associated
with a physical entity. (issue ASTERISK-20175) Reported by:
Russell Bryant, Leif Madsen, Joshua Colp Tested by: kmoore
patches: event-cachability-3.diff uploaded by jcolp (license
5000)
* res/res_jabber.c, channels/sip/include/sip.h,
channels/chan_sip.c, main/http.c: Resolve crashes due to large
stack allocations when using TCP Asterisk had several places
where messages received over various network transports may be
copied in a single stack allocation. In the case of TCP, since
multiple packets in a stream may be concatenated together, this
can lead to large allocations that overflow the stack. This patch
modifies those portions of Asterisk using TCP to either favor
heap allocations or use an upper bound to ensure that the stack
will not overflow: * For SIP, the allocation now has an upper
limit * For HTTP, the allocation is now a heap allocation instead
of a stack allocation * For XMPP (in res_jabber), the allocation
has been eliminated since it was unnecesary. Note that the HTTP
portion of this issue was independently found by Brandon Edwards
of Exodus Intelligence. (issue ASTERISK-20658) Reported by:
wdoekes, Brandon Edwards Tested by: mmichelson, wdoekes patches:
ASTERISK-20658_res_jabber.c.patch uploaded by mmichelson (license
5049) issueA20658_http_postvars_use_malloc2.patch uploaded by
wdoekes (license 5674) issueA20658_limit_sip_packet_size3.patch
uploaded by wdoekes (license 5674)
2012-12-31 14:41 +0000 [r378217] Kinsey Moore <kmoore@digium.com>
* channels/chan_sip.c: Ensure chan_sip rejects encrypted streams
without crypto info This ensures that Asterisk rejects encrypted
media streams (RTP/SAVP audio and video) that are missing
cryptographic keys and ensures that the incoming SDP is
consistent with RFC4568 as far as having a crypto attribute
present for any SAVP streams. Review:
https://reviewboard.asterisk.org/r/2204/
2012-12-20 21:38 +0000 [r378164] Richard Mudgett <rmudgett@digium.com>
* main/channel.c: Give the causes[] a struct name.
2012-12-20 20:26 +0000 [r378147] Mark Michelson <mmichelson@digium.com>
* include/asterisk/rtp_engine.h: Adjust RTP instance's
available_formats callback to return the correct type. The RTP
engine public function that gets the available formats expects a
format_t to be returned; however when calling into an RTP
instance's callback to get the available formats, the callback
returned an int. This never was noticed in Asterisk because the
two RTP engines included do not provide an available_formats
callback. This introduces an API change, and the proposal for
this change was brought up on the Asterisk developers mailing
list [1]. There was no public objection to this change, so it is
now being put in. (closes AST-1054) reported by Doug Bailey [1]
http://lists.digium.com/pipermail/asterisk-dev/2012-December/058058.html
2012-12-18 17:35 +0000 [r378119] Kinsey Moore <kmoore@digium.com>
* main/channel.c: Add test events for time limit-related hangups
This patch adds hangup-related test events in order to support
testing of time-limited bridges. This aids in testing the S() and
L() bridge options. (issue SWP-4713)
2012-12-17 23:07 +0000 [r378088-378092] Richard Mudgett <rmudgett@digium.com>
* main/loader.c: Fix potential double free when unloading a module.
* channels/chan_local.c: Make chan_local module references tied to
local_pvt lifetime. The chan_local module references were
manually tied to the existence of the ;1 and ;2 channel links. *
Made chan_local module references tied to the existence of the
local_pvt structure as well as automatically take care of the
module references. * Tweaked the wording of the local_fixup()
failure warning message to make sense. Review:
https://reviewboard.asterisk.org/r/2181/
2012-12-14 21:23 +0000 [r378036] Richard Mudgett <rmudgett@digium.com>
* apps/app_queue.c: app_queue: Revert bad ringinuse=no patch. With
the option ringinuse=no set, the patch committed for
ASTERISK-16115 causes non-SIP queue members to never be called
because the device state is checked after a channel is created to
determine if the member is busy. These queue members always get
the "Member %s is busy, cannot dial" message. Most channel
drivers other than chan_sip use the default device state
handling. The default device-state state is considered in use or
unknown if the channel exists or not respectively. (closes issue
ASTERISK-20801) Reported by: rmudgett Patches:
jira_asterisk_16115_revert_r370418_v1.8.patch (license #5621)
patch uploaded by rmudgett
2012-12-13 13:43 +0000 [r377946] Kinsey Moore <kmoore@digium.com>
* channels/chan_sip.c: Ensure Min-SE is included in outbound
INVITEs Asterisk now includes Min-SE in outbound INVITEs when the
value is not 90 (the default) and session timers are not
disabled. This has the effect of Asterisk following RFC4028 more
closely with regard to 422 responses and preventing situations in
which Asterisk would be forced to temporarily accept a call to
tear it down based on a Session-Expires below the locally
configured Min-SE. (issue SWP-5051) Review:
https://reviewboard.asterisk.org/r/2222/ Reported-by: Kinsey
Moore Patch-by: Kinsey Moore
2012-12-12 22:39 +0000 [r377922] Rusty Newton <rnewton@digium.com>
* sounds/Makefile: Incremented EXTRA_SOUNDS_VERSION in
sounds/Makefile to 1.4.12 for new Extra Sounds releases See
CHANGES-* files in English extra 1.4.12 tarballs for new sound
prompts added. (closes ASTERISK-20328) Reported by: Matt Jordan
(closes AST-755) Reported by: John Bigelow
2012-12-11 21:54 +0000 [r377847-377881] Richard Mudgett <rmudgett@digium.com>
* main/aoc.c, main/image.c, main/cel.c, main/timing.c,
main/channel.c, main/data.c, main/stun.c, main/file.c,
main/http.c: Cleanup CLI commands on exit for several files.
(issue ASTERISK-20649) Reported by: Corey Farrell Patches:
unregister-cli-multiple-all.patch (license #5909) patch uploaded
by Corey Farrell
* main/udptl.c: Cleanup udptl on exit. * Cleanup CLI commands on
exit. (issue ASTERISK-20649) Reported by: Corey Farrell Patches:
udptl-shutdown-1_8-10.patch (license #5909) patch uploaded by
Corey Farrell udptl-shutdown-11-trunk.patch (license #5909) patch
uploaded by Corey Farrell Modified
2012-12-11 20:45 +0000 [r377840] Mark Michelson <mmichelson@digium.com>
* res/res_clialiases.c: Fix crash that can occur if CLI
registration fails for an aliased command. A recent memory leak
fix in main/cli.c causes an ast_cli_entry's command field to be
freed and NULLed if ast_cli_register() fails. res_clialiases was
ignoring the return value of ast_cli_register() and was then
passing the NULL command off to a a hash function. This resulted
in a crash. The fix is not to ignore the erroneous return value.
If ast_cli_register() fails, then we do not continue trying to
process the current alias.
2012-12-11 20:37 +0000 [r377688-377837] Richard Mudgett <rmudgett@digium.com>
* main/taskprocessor.c: Cleanup taskprocessor on exit. * Cleanup
CLI commands on exit. (issue ASTERISK-20649) Reported by: Corey
Farrell Patches: taskprocessor-cleanup-1_8-11-trunk.patch
(license #5909) patch uploaded by Corey Farrell
taskprocessor-cleanup-10-only.patch (license #5909) patch
uploaded by Corey Farrell Modified
* main/pbx.c: Cleanup pbx on exit. * Cleanup CLI commands on exit.
* Unreference hints and statecbs containers on exit. (issue
ASTERISK-20649) Reported by: Corey Farrell Patches:
pbx-cleanup-1_8.patch (license #5909) patch uploaded by Corey
Farrell pbx-cleanup-10.patch (license #5909) patch uploaded by
Corey Farrell pbx-cleanup-11-trunk.patch (license #5909) patch
uploaded by Corey Farrell Modified
* main/logger.c: Cleanup logger on exit. * Cleanup CLI commands,
destroy verbosers and logchannels lists on exit. (issue
ASTERISK-20649) Reported by: Corey Farrell Patches:
logger-cleanup-all.patch (license #5909) patch uploaded by Corey
Farrell Modified
* main/indications.c: Cleanup indications on exit. * Made
ast_unregister_indication_country() unlink the found tone zone
before selecting a new default_tone_zone to make it impossible to
select the tone zone being unregistered again. * Ringcadence is
no longer parsed twice in store_config_tone_zone(). * Cleanup CLI
commands and destroy default_tone_zone on exit. (issue
ASTERISK-20649) Reported by: Corey Farrell Patches:
indications-cleanup-all.patch (license #5909) patch uploaded by
Corey Farrell Modified
* main/frame.c: Cleanup frame on exit. * Cleanup CLI commands on
exit. (issue ASTERISK-20649) Reported by: Corey Farrell Patches:
frame-cleanup-1_8-only.patch (license #5909) patch uploaded by
Corey Farrell
* main/event.c: Cleanup event on exit. * Cleanup CLI commands on
exit. (issue ASTERISK-20649) Reported by: Corey Farrell Patches:
event_shutdown-10-only.patch (license #5909) patch uploaded by
Corey Farrell event_shutdown-1_8-11-trunk.patch (license #5909)
patch uploaded by Corey Farrell
* main/dnsmgr.c: Cleanup dnsmgr on exit. * Cleanup dnsmgr thread
and CLI commands on exit. (issue ASTERISK-20649) Reported by:
Corey Farrell Patches: dnsmgr-cleanup-1_8.patch (license #5909)
patch uploaded by Corey Farrell dnsmgr-cleanup-10-11-trunk.patch
(license #5909) patch uploaded by Corey Farrell Modified
* main/db.c: Cleanup astdb on exit. * Cleanup astdb thread and CLI
commands on exit. (issue ASTERISK-20649) Reported by: Corey
Farrell Patches: db-cleanup-1_8-only.patch (license #5909) patch
uploaded by Corey Farrell Modified
2012-12-10 16:51 +0000 [r377623-377655] Kinsey Moore <kmoore@digium.com>
* res/res_fax.c: Ensure ReceiveFax provides a CED tone via T.38
When using res_fax_digium, the T.38 CED tone was not being
provided properly which would cause some incoming faxes to fail.
This was not an issue with res_fax_spandsp since it does not
strictly honor the send_ced flag and sends the CED tone whenever
receiving a T.38 fax. (closes issue FAX-343) Reported-by:
Benjamin Tietz Patch-by: Kinsey Moore
* channels/chan_sip.c: Handle Session-Expires less than local
Min-SE in 200 OK Ensure that a call is immediately torn down if a
Session-Expires value received in a 200 OK is less than the local
Min-SE. This also prevents Asterisk from allowing calls with
Session-Expires below the RFC4028-mandated minimum (90s). (closes
issue ASTERISK-20653) Review:
https://reviewboard.asterisk.org/r/2237/ Patch-by: Kinsey Moore
2012-12-10 06:40 +0000 [r377557-377591] Igor Goncharovskiy <igor.goncharovsky@gmail.com>
* channels/chan_unistim.c: Fix codec mismatch Fix code to send in
both rx and tx open stream messages correct codecs. Found that on
phase 0/1 phones wrong codecs cause to no audio in some
situations. (issue ASTERISK-20183)
* channels/chan_unistim.c: Fix crash on transfer initiated from
insreeen menu on Unistim phones. Removed CDR-related code that
moved to do_masquarade before. (closes issue ASTERISK-20417)
Reported by: Rudolf Migalin
2013-01-14 Asterisk Development Team <asteriskteam@digium.com>
* Asterisk 1.8.20.0 Released.
2013-01-09 Asterisk Development Team <asteriskteam@digium.com>
* Asterisk 1.8.20.0-rc2 Released.
* AST-2012-014: Resolve crashes due to large stack allocations when
using TCP
Asterisk had several places where messages received over various
network transports may be copied in a single stack allocation. In
the case of TCP, since multiple packets in a stream may be
concatenated together, this can lead to large allocations that
overflow the stack.
This patch modifies those portions of Asterisk using TCP to either
favor heap allocations or use an upper bound to ensure that the
stack will not overflow:
* For SIP, the allocation now has an upper limit
* For HTTP, the allocation is now a heap allocation instead of a
stack allocation
* For XMPP, the allocation has been eliminated since it was
unnecessary.
* AST-2012-015: Prevent exhaustion of system resources through
exploitation of event cache
Asterisk maintains an internal cache for devices in the event
subsystem. The device state cache holds the state of each device
known to Asterisk, such that consumers of device state information
can query for the last known state for a particular device, even if
it is not part of an active call. The concept of a device in
Asterisk can include entities that do not have a physical
representation. One way that this occurred was when anonymous calls
are allowed in Asterisk. A device was automatically created and
stored in the cache for each anonymous call that occurred; this was
possible in the SIP and IAX2 channel drivers and through channel
drivers that utilized the res_jabber/res_xmpp resource modules (Gtalk,
Jingle, and Motif). These devices are never removed from the system,
allowing anonymous call to potentially exhaust a system's resources.
This patch changes the event cache subsystem and device state
management to no longer cache devices that are not associated with a
physical entity.
* Revert bad ringinuse=no patch.
With the option ringinuse=no set, the patch committed previous for
ASTERISK-16115 causes non-SIP queue members to never be called
because the device state is checked after a channel is created to
determine if the member is busy. These queue members always get the
"Member %s is busy, cannot dial" message.
Most channel drivers other than chan_sip use the default device
state handling. The default device state is considered in use or
unknown if the channel exists or not, respectively.
* Fix multiple calls to a queue member that is only in queue.
When ringinuse=no queue members can receive more than one call if
these calls happen at nearly the same time. This patch fixes it so a
queu member does not receive more than one call from a queue. note
that this fix does not prevent multiple calls to a member if hte
member is in more than one queue (see ASTERISK-16115).
2012-12-10 Asterisk Development Team <asteriskteam@digium.com>
* Asterisk 1.8.20.0-rc1 Released.
2012-12-10 01:38 +0000 [r377487-377509] Tilghman Lesher <tilghman@meg.abyt.es>
* main/xmldoc.c: Improve documentation by making all of the colors
used readable, no matter what the background color is. Dark blue
on a black background is unreadable, as is yellow on a light
background. This patch turns on the bright attribute for colors
when on a dark background and turns *off* the bright attribute
when the -W command line option is used (indicating a _light_
background). This ensures that text is readable in both cases.
Patch by: tilghman Review:
https://reviewboard.asterisk.org/r/2224
* addons/cdr_mysql.c: Remove some dead code and additionally handle
a case that wasn't handled.
2012-12-08 00:28 +0000 [r377398-377431] Richard Mudgett <rmudgett@digium.com>
* contrib/realtime/mysql/sippeers.sql: Fix order of SIP
allow/disallow in MySQL contrib script. Using the contrib
sippeers.sql script to create the sippeers MySQL table would
result in being unable to place calls if you set the disallow
value to all. (closes issue ASTERISK-20756) Reported by: Andre
Luis Patches: sippeers.patch patch uploaded by Andre Luis
* main/astmm.c: MALLOC_DEBUG: Only wait if we want atexit
allocation dumps.
2012-12-05 16:48 +0000 [r377257] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: Fix a SIP request memory leak with TLS
connections. During the TLS re-work in chan_sip some TLS specific
code was moved into a separate function. This function operates
on a copy of the incoming SIP request. This copy was never
deinitialized causing a memory leak for each request processed.
This function is now given a SIP request structure which it can
use to copy the incoming request into. This reduces the amount of
memory allocations done since the internal allocated components
are reused between packets and also ensures the SIP request
structure is deinitialized when the TLS connection is torn down.
(closes issue ASTERISK-20763) Reported by: deti
2012-12-05 16:46 +0000 [r377256] Jonathan Rose <jrose@digium.com>
* res/res_srtp.c: res_srtp: Fix a crash caused by srtp_dealloc on
an already dealloced session When srtp_create fails, the session
may be dealloced or just not alloced. At the same time though,
the session pointer might not be set to NULL in this process and
attempting to srtp_dealloc it again will cause a segfault. This
patch checks for failure of srtp_create and sets the session
pointer to NULL if it fails. (closes issue ASTERISK-20499)
Reported by: tootai Review:
https://reviewboard.asterisk.org/r/2228/
2012-12-03 22:51 +0000 [r377037-377165] Richard Mudgett <rmudgett@digium.com>
* main/asterisk.c: Cleanup ast_run_atexits() atexits list. *
Convert atexits list to a mutex instead of a rd/wr lock. The lock
is only write locked. * Move CLI verbose Asterisk ending message
to where AMI message is output in really_quit() to avoid further
surprises about using stuff already shutdown. (issue
ASTERISK-20649) Reported by: Corey Farrell
* include/asterisk/_private.h, main/stdtime/localtime.c,
main/asterisk.c: Cleanup core main on exit. * Cleanup time zones
on exit. * Make exit clean/unclean report consistent for AMI and
CLI in really_quit(). (issue ASTERISK-20649) Reported by: Corey
Farrell Patches: core-cleanup-1_8-10.patch (license #5909) patch
uploaded by Corey Farrell core-cleanup-11-trunk.patch (license
#5909) patch uploaded by Corey Farrell Modified
* main/config.c: Cleanup config cache on exit. (issue
ASTERISK-20649) Reported by: Corey Farrell Patches:
config-cleanup-all.patch (license #5909) patch uploaded by Corey
Farrell
* main/cli.c: Cleanup CLI resources on exit and CLI command
registration errors. (issue ASTERISK-20649) Reported by: Corey
Farrell Patches: cli-leaks-1_8-10.patch (license #5909) patch
uploaded by Corey Farrell cli-leaks-11-trunk.patch (license
#5909) patch uploaded by Corey Farrell Modified
* main/cdr.c: Cleanup CDR resources on exit. * Simplify do_reload()
return handling since it never returned anything other than 0.
(issue ASTERISK-20649) Reported by: Corey Farrell Patches:
cdr-cleanup-1_8.patch (license #5909) patch uploaded by Corey
Farrell cdr-cleanup-10-11-trunk.patch (license #5909) patch
uploaded by Corey Farrell Modified
* main/ccss.c: Fix CCSS CLI commands and logger level not
unregistered. (issue ASTERISK-20649) Reported by: Corey Farrell
Patches: ccss-cleanup-all.patch (license #5909) patch uploaded by
Corey Farrell
2012-11-30 21:30 +0000 [r376950] Richard Mudgett <rmudgett@digium.com>
* channels/misdn/isdn_lib.c: chan_misdn: Fix sending
RELEASE_COMPLETE in response to SETUP. Fix sending a
RELEASE_COMPLETE in response to a SETUP if chan_misdn does not
have a B channel available to assign to the call. (closes issue
ABE-2869) Reported by: Guenther Kelleter Patches:
setup-reject_2.diff (license #6372) patch uploaded by Guenther
Kelleter Modified ........ Merged revision 376949 from
https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
2012-11-30 17:04 +0000 [r376919] Sean Bright <sean@malleable.com>
* funcs/func_volume.c: Minor spelling fix to the VOLUME
documentation.
2012-11-30 16:12 +0000 [r376901] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Fix potential crashes during SIP attended
transfers. The principal behind this patch is simple. During a
transfer, we manipulate channels that are owned by a separate
thread than the one we currently are running in, so it makes
sense that we need to grab a reference to the channels so that
they cannot disappear out from under us. In the wild, crashes
were sometimes seen when the transferring party would hang up the
call before the transfer target answered the call. The most
common place to see the crash occur was when attempting to send a
connected line update to the transferer channel. (closes issue
ASTERISK-20226) Reported by Jared Smith Patches:
ASTERISK-20226.patch uploaded by Mark Michelson (License #5049)
Tested by: Jared Smith
2012-11-29 22:55 +0000 [r376864-376868] Richard Mudgett <rmudgett@digium.com>
* channels/chan_local.c: chan_local: Fix local_pvt ref leak in
local_devicestate(). Regression introduced by ASTERISK-20390 fix.
(closes issue ASTERISK-20769) Reported by: rmudgett Tested by:
rmudgett
* channels/chan_sip.c: Fix compile error. (issue ASTERISK-20724)
2012-11-29 21:49 +0000 [r376834] Michael L. Young <elgueromexicano@gmail.com>
* channels/chan_sip.c: Improve Code Readability And Fix Setting
natdetected Flag For 1.8, 10, 11 and trunk we are are improving
the code readability. For 11 and trunk, auto nat detection was
added. The natdetected flag was being set to 1 when the host
address in the VIA header did not specifiy a port. This patch
fixes this by setting the port on the temporary sock address used
to SIP_STANDARD_PORT in order for the sock address comparison to
work properly. (closes issue ASTERISK-20724) Reported by: Michael
L. Young Patches: asterisk-20724-set-port-v2.diff uploaded by
Michael L. Young (license 5026) Review:
https://reviewboard.asterisk.org/r/2206/
2012-11-29 00:42 +0000 [r376758-376788] Richard Mudgett <rmudgett@digium.com>
* main/astmm.c, main/asterisk.c: Add MALLOC_DEBUG atexit unreleased
malloc memory summary. * Adds the following CLI commands to
control MALLOC_DEBUG reporting of unreleased malloc memory when
Asterisk is shut down. memory atexit list on memory atexit list
off memory atexit summary byline memory atexit summary byfunc
memory atexit summary byfile memory atexit summary off * Made
check all remaining allocated region blocks atexit for fence
violations. * Increased the allocated region hash table size by
about three times. It still isn't large enough considering the
number of malloced blocks Asterisk uses. * Made CLI "memory show
allocations anomalies" use regions_check_all_fences(). Review:
https://reviewboard.asterisk.org/r/2196/
* main/astmm.c: Enhance MALLOC_DEBUG CLI commands. * Fixed CLI
"memory show allocations" misspelling of anomalies option. The
command will still accept the original misspelling. *
Miscellaneous tweaks to CLI "memory show allocations" command
output format. * Made CLI "memory show summary" summarize by line
number instead of by function if a filename is given. * Made CLI
"memory show summary" sort its output by filename or
function-name/line-number depending upon request. * Miscellaneous
tweaks to CLI "memory show summary" command output format.
2012-11-28 16:23 +0000 [r376725] Jonathan Rose <jrose@digium.com>
* main/manager.c: manager: Make challenge work with
allowmultiplelogin=no Prior to this patch, challenge would yield
a multiple logins error if used without providing the username
(which isn't really supposed to be an argument to challenge) if
allowmultiplelogin was set to no because allowmultiplelogin finds
a user with a zero length login name. This check is simply
disabled for the challenge action when the username is empty by
this patch. (closes issue ASTERISK-20677) Reported by: Vladimir
Patches: challenge_action_nomultiplelogin.diff uploaded by
Jonathan Rose (license 6182)
2012-11-27 23:47 +0000 [r376627-376688] Richard Mudgett <rmudgett@digium.com>
* UPGRADE.txt, main/pbx.c: Fix extension matching with the '-'
char. The '-' char is supposed to be ignored by the dialplan
extension matching. Unfortunately, it's treatment is not handled
consistently throughout the extension matching code. * Made the
old exten matching code consistently ignore '-' chars. * Made the
old exten matching code consistently handle case in the matching.
* Made ignore empty character sets. * Fixed ast_extension_cmp()
to return -1, 0, or 1 as documented. The only user of it in
pbx_lua.c was testing for -1. It was originally returning the
strcmp() value for less than which is not usually going to be -1.
* Fix character set sorting if the sets have the same number of
characters and start with the same character. Character set [0-9]
now sorts before [02-9a] as originally intended. * Updated some
extension label and priority already in use warnings to also
indicate if the extension is aliased. (closes issue
ASTERISK-19205) Reported by: Philippe Lindheimer, Birger "WIMPy"
Harzenetter Tested by: rmudgett Review:
https://reviewboard.asterisk.org/r/2201/
* pbx/pbx_dundi.c, addons/res_config_mysql.c,
apps/app_celgenuserevent.c: Remove unnecessary channel module
references. * Removed call to ast_module_user_hangup_all() in
res_config_mysql.c since it is effectively a noop. No channels
can attach a reference to that module. * Removed call to
ast_module_user_hangup_all() in app_celgenuserevent.c. The caller
of unload_module() has already called it. * Removed redundant
channel module references in pbx_dundi.c. The registered dialplan
function callback dispatchers for the read/read2/write callbacks
already reference the module before calling. * pbx_dundi: Moved
unregistering CLI commands, DUNDi switch, and dialplan functions
to the first thing the unload_module() does. This will reduce the
chance of new channels using DUNDi services while the module is
being torn down.
* include/asterisk/linkedlists.h: Made AST_LIST_REMOVE() simpler
and use better names. * Update doxygen of AST_LIST_REMOVE().
2012-11-22 23:51 +0000 [r376586] Matthew Jordan <mjordan@digium.com>
* include/asterisk/lock.h, main/lock.c, main/logger.c:
Re-initialize logmsgs mutex upon logger initialization to prevent
lock errors Similar to the patch that moved the fork earlier in
the startup sequence to prevent mutex errors in the recursive
mutex surrounding the read/write thread registration lock, this
patch re-initializes the logmsgs mutex. Part of the start up
sequence before forking the process into the background includes
reading asterisk.conf; this has to occur prior to the call to
daemon in order to read startup parameters. When reading in a
conf file, log statements can be generated. Since this can't be
avoided, the mutex instead is re-initialized to ensure a reset of
any thread tracking information. This patch also includes some
additional debugging to catch errors when locking or unlocking
the recursive mutex that surrounds locks when the DEBUG_THREADS
build option is enabled. DO_CRASH or THREAD_CRASH will cause an
abort() if a mutex error is detected. (issue ASTERISK-19463)
Reported by: mjordan Tesetd by: mjordan
2012-11-20 16:45 +0000 [r376521] Mark Michelson <mmichelson@digium.com>
* channels/sip/include/sip.h, channels/chan_sip.c: Add "Require:
timer" to 200 OK responses when appropriate. The method by which
the Require header is added to 200 responses is inspired by the
method that Olle Johansson uses in his darjeeling-prack branch.
(closes issue ASTERISK-20570) Reported by Matt Jordan, at the
behest of Olle Johansson Review:
https://reviewboard.asterisk.org/r/2172
2012-11-19 19:30 +0000 [r376469] Walter Doekes <walter+asterisk@wjd.nu>
* main/indications.c, channels/chan_sip.c, main/security_events.c:
Fix most leftover non-opaque ast_str uses. Instead of calling
str->str, one should use ast_str_buffer(str). Same goes for
str->used as ast_str_strlen(str) and str->len as
ast_str_size(str). Review:
https://reviewboard.asterisk.org/r/2198
2012-11-18 20:11 +0000 [r376428] Matthew Jordan <mjordan@digium.com>
* main/utils.c, main/stdtime/localtime.c, main/asterisk.c: Reorder
startup sequence to prevent lockups when process is sent to
background Although it is very rare and timing dependent, the
potential exists for the call to 'daemon' to cause what appears
to be a deadlock in Asterisk during startup. This can occur when
a recursive mutex is obtained prior to the daemon call executing.
Since daemon uses fork to send the process into the background,
any threading primitives are unsafe to re-use after the call.
Implementations of pthread recursive mutexes are highly likely to
store the thread identifier of the thread that previously
obtained the mutex. If the mutex was locked prior to the fork, a
subsequent unlock operation will potentially fail as the thread
identifier is no longer valid. Since the mutex is still locked,
all subsequent attempts to grab the mutex by other threads will
block. This behavior exhibited itself most often when
DEBUG_THREADS was enabled, as this compile time option surrounds
the mutexes in Asterisk with another recursive mutex that
protects the storage of thread related information. This made it
much more likely that a recursive mutex would be obtained prior
to daemon and unlocked after the call. This patch does the
following: a) It backports a patch from Asterisk 11 that prevents
the spawning of the localtime monitoring thread. This thread is
now spawned after Asterisk has fully booted. b) It re-orders the
startup sequence to call daemon earlier during Asterisk startup.
This limits the potential of threading primitives being accessed
by initialization calls before daemon is called. c) It removes
calls to ast_verbose/ast_log/etc. prior to daemon being called.
Developers should send error messages directly to stderr prior to
daemon, as calls to ast_log may access recursive mutexes that
store thread related information. d) It reorganizes when thread
local storage is created for storing lock information during the
creation of threads. Prior to this patch, the read/write lock
protecting the list of threads in ast_register_thread would
utilize the lock in the thread local storage prior to it being
initialized; this patch prevents that. On a very related note,
this patch will *greatly* improve the stability of the Asterisk
Test Suite. Review: https://reviewboard.asterisk.org/r/2197
(closes issue ASTERISK-19463) Reported by: mjordan Tested by:
mjordan
2012-11-16 19:31 +0000 [r376389] Jonathan Rose <jrose@digium.com>
* res/res_monitor.c: monitor: prevent attempts to move/remove
recordings skipped with 'i' and 'o'. The i and o options for
monitor skip the input and output sides of a recording
respectively. This patch addresses a problem in those options
when monitor is called without specifying a specific filename
where monitor will try to move the recording that was skipped.
Since this usually doesn't exist when these options are used, it
would produce a warning when it does this in most cases, but it
is conceivable that there are use cases where this could result
in moving/removing a file unintentionally. (closes issue
ASTERISK-20641) Reported by: Jonathan Rose Review:
https://reviewboard.asterisk.org/r/2190/
2012-11-15 23:58 +0000 [r376340] David M. Lee <dlee@digium.com>
* utils/extconf.c: Fixed extconf.c breakage introduced in r376306.
To quote wdoekes: > Note that I'm not confirming legitimacy of
having that file in tree at > all. Is anyone using
aelparse/conf2ael?
2012-11-15 22:40 +0000 [r376307] Jonathan Rose <jrose@digium.com>
* apps/app_meetme.c: app_meetme: Fix channels lingering when hung
up under certain conditions Channels would get stuck and MeetMe
would repeatedly display an Unable to write frame to channel
error in the conf_run function if hung up during certain sound
prompts such as during user count announcements. This patch fixes
that by reintroducing a hangup check in the meetme's main loop
(also in conf_run). (closes issue ASTERISK-20486) Reported by:
Michael Cargile Review: https://reviewboard.asterisk.org/r/2187/
Patches: meetme_hangup_patch_ASTERISK-20486_v3.diff uploaded by
Jonathan Rose (license 6182)
2012-11-15 22:27 +0000 [r376306] David M. Lee <dlee@digium.com>
* tests/test_hashtab_thrash.c (added), utils/hashtest2.c (removed),
include/asterisk/hashtab.h, utils/Makefile,
tests/test_astobj2_thrash.c (added), utils/utils.xml,
utils/hashtest.c (removed): Migrate hashtest/hashtest2 to be unit
tests. Both hashtest and hashtest2 are manual testing apps that
thrash hash tables (hashtab and ao2 containers, respectively), by
spinning up several threads that randomly insert, delete, lookup
and iterate over the hash table. If the app doesn't crash, the
hash table probably passes the test. Those utils are not a part
of the typical Asterisk build, so they do not usually get
compiled. This all makes them less that useful. This patch
removes those manual test programs and replaces them with
Asterisk unit test modules (test_{hashtab,astobj2}_thrash.so). It
also attempts to make the tests more deterministic. * Rather than
spinning up some number of threads that operate on the hash table
randomly, spin up four threads that concurrenly add, remove,
lookup and iterate over the hash table. * Each thread checks the
state of the hash table both during and after execution, and
indicates a test failure if things are not as expected. * Each
thread times out after 60 seconds to prevent deadlocking the unit
test run. (closes issue ASTERISK-20505) Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2189/
2012-11-15 01:43 +0000 [r376262] Rusty Newton <rnewton@digium.com>
* apps/app_voicemail.c: Patch to play correct sound file when a
voicemail's urgent status is removed We were attempting to play
"vm-urgent-removed", which didn't exist. Now we play
"vm-marked-nonurgent" which exists and is the correct sound file.
Previous behavior was silence and a warning on the CLI. (issue
ASTERISK-20280) (closes issue ASTERISK-20280) Reported by: Tomo
Takebe Tested by: Rusty Newton Patches: asterisk20280.patch
uploaded by Rusty Newton (license 5829)
2012-11-14 19:48 +0000 [r376232] Richard Mudgett <rmudgett@digium.com>
* pbx/pbx_spool.c: Fix call files when astspooldir is relative.
Future dated call files are ignored when astspooldir is relative
to the current directory. The queue_file() assumed that the qdir
needed to be prepended if the given filename did not start with a
'/'. If astspooldir is relative it is not going to start from the
root directory obviously so it will not start with a '/'. The
filename used in queue_file() ultimately results in qdir
prepended multiple times. * Made queue_file() not prepend qdir if
the filename contains a '/'. (closes issue ASTERISK-20593)
Reported by: James Le Cuirot Patches:
0004-Fix-future-call-files-from-relative-directories.patch
(license #6439) patch uploaded by James Le Cuirot
2012-11-13 18:10 +0000 [r376199] Brent Eagles <beagles@digium.com>
* main/channel.c: Patch to prevent stopping the active generator
when it is not the silence generator. This patch introduces an
internal helper function to safely check whether the current
generator is the one that is expected before deactivating it. The
current externally accessible ast_channel_stop_generator()
function has been modified to be implemented in terms of the new
function. (closes issue ASTERISK-19918) Reported by: Eduardo Abad
2012-11-12 20:44 +0000 [r376166] Joshua Colp <jcolp@digium.com>
* main/pbx.c: Properly check if the "Context" and "Extension"
headers are empty in a ShowDialPlan action. The code which
handles the ShowDialPlan action wrongly assumed that a non-NULL
return value from the function which retrieves headers from an
action indicates that the header has a value. This is incorrect
and the contents must be checked to see if they are blank.
(closes issue ASTERISK-20628) Reported by: jkroon Patches:
asterisk-showdialplan-incorrect-error.patch uploaded by jkroon
2012-11-12 20:13 +0000 [r376142] Michael L. Young <elgueromexicano@gmail.com>
* main/pbx.c: Fix Dynamic Hints Variable Substition - Underscore
Problem When adding a dynamic hint, if an extension contains an
underscore no variable subsitution is being performed. This patch
changes from checking if the extension contains an underscore to
checking if the extension begins with an underscore. (closes
issue ASTERISK-20639) Reported by: Steven T. Wheeler Tested by:
Steven T. Wheeler, Michael L. Young Patches:
asterisk-20639-dynamic-hint-underscore.diff uploaded by Michael
L. Young (license 5026) Review:
https://reviewboard.asterisk.org/r/2188/
2012-11-08 21:56 +0000 [r376087] Mark Michelson <mmichelson@digium.com>
* res/res_fax.c: Fix a "set but not used" warning on newer gccs.
Turns out the "helpful" setting of ms and res in this macro is
completely useless after the timeout antipattern fix. If you're a
new guy looking to write code, don't write a macro like this one.
2012-11-08 21:05 +0000 [r376029-376058] Richard Mudgett <rmudgett@digium.com>
* channels/sig_ss7.c: chan_dahdi/SS7: Made reject incoming call for
an in-alarm or blocked channel. If a SS7 call comes in requesting
a CIC that is in-alarm, the call is accepted and connects if the
extension exists in the dialplan. The call does not have any
audio. * Made release the call immediately with circuit
congestion cause. (closes issue ASTERISK-20204) Reported by: Tuan
Le Patches: jira_asterisk_20204_v1.8.patch (license #5621) patch
uploaded by rmudgett
* main/utils.c, main/astmm.c, main/asterisk.c,
include/asterisk/utils.h, include/asterisk/astmm.h: Add
MALLOC_DEBUG enhancements. * Makes malloc() behave like calloc().
It will return a memory block filled with 0x55. A nonzero value.
* Makes free() fill the released memory block and boundary
fence's with 0xdeaddead. Any pointer use after free is going to
have a pointer pointing to 0xdeaddead. The 0xdeaddead pointer is
usually an invalid memory address so a crash is expected. * Puts
the freed memory block into a circular array so it is not reused
immediately. * When the circular array rotates out a memory block
to the heap it checks that the memory has not been altered from
0xdeaddead. * Made the astmm_log message wording better. * Made
crash if the DO_CRASH menuselect option is enabled and something
is found. * Fixed a potential alignment issue on 64 bit systems.
struct ast_region.data[] should now be aligned correctly for all
platforms. * Extracted region_check_fences() from
__ast_free_region() and handle_memory_show(). * Updated
handle_memory_show() CLI usage help. Review:
https://reviewboard.asterisk.org/r/2182/
2012-11-07 17:08 +0000 [r375993-375994] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Remove some debugging that accidentally made
it in the last commit.
* main/utils.c, include/asterisk/channel.h, apps/app_queue.c,
channels/sig_pri.c, channels/chan_iax2.c, main/channel.c,
channels/chan_dahdi.c, channels/sig_analog.c,
apps/app_waitforring.c, include/asterisk/time.h, apps/app_jack.c,
apps/app_dial.c, main/pbx.c, main/rtp_engine.c,
channels/chan_sip.c, apps/app_meetme.c, res/res_fax.c,
apps/app_record.c, channels/chan_agent.c: Fix misuses of timeouts
throughout the code. Prior to this change, a common method for
determining if a timeout was reached was to call a function such
as ast_waitfor_n() and inspect the out parameter that told how
many milliseconds were left, then use that as the input to
ast_waitfor_n() on the next go-around. The problem with this is
that in some cases, submillisecond timeouts can occur, resulting
in the out parameter not decreasing any. When this happens
thousands of times, the result is that the timeout takes much
longer than intended to be reached. As an example, I had a
situation where a 3 second timeout took multiple days to finally
end since most wakeups from ast_waitfor_n() were under a
millisecond. This patch seeks to fix this pattern throughout the
code. Now we log the time when an operation began and find the
difference in wall clock time between now and when the event
started. This means that sub-millisecond timeouts now cannot play
havoc when trying to determine if something has timed out. Part
of this fix also includes changing the function ast_waitfor() so
that it is possible for it to return less than zero when a
negative timeout is given to it. This makes it actually possible
to detect errors in ast_waitfor() when there is no timeout.
(closes issue ASTERISK-20414) reported by David M. Lee Review:
https://reviewboard.asterisk.org/r/2135/
2012-11-06 18:18 +0000 [r375964] Richard Mudgett <rmudgett@digium.com>
* include/asterisk/channel.h, .cleancount,
include/asterisk/features.h, main/channel.c, main/features.c: Fix
stuck DTMF when bridge is broken. When a bridge is broken by an
AMI Redirect action or the ChannelRedirect application, an in
progress DTMF digit could be stuck sending forever. * Made
simulate a DTMF end event when a bridge is broken and a DTMF
digit was in progress. (closes issue ASTERISK-20492) Reported by:
Jeremiah Gowdy Patches: bridge_end_dtmf-v3.patch.txt (license
#6358) patch uploaded by Jeremiah Gowdy Modified to
jira_asterisk_20492_v1.8.patch jira_asterisk_20492_v1.8.patch
(license #5621) patch uploaded by rmudgett Tested by: rmudgett
Review: https://reviewboard.asterisk.org/r/2169/
2012-12-10 Asterisk Development Team <asteriskteam@digium.com>
* Asterisk 1.8.19.0 Released.
2012-12-06 Asterisk Development Team <asteriskteam@digium.com>
* Asterisk 1.8.19.0-rc3 Released.
* chan_local: Fix local_pvt ref leak in local_devicestate().
Regression introduced by ASTERISK-20390 fix.
(closes issue ASTERISK-20769)
Reported by: rmudgett
2012-12-05 Asterisk Development Team <asteriskteam@digium.com>
* Asterisk 1.8.19.0-rc2 Released.
* Fix a SIP request memory leak with TLS connections.
During the TLS re-work in chan_sip some TLS specific code was moved
into a separate function. This function operates on a copy of the
incoming SIP request. This copy was never deinitialized causing a
memory leak for each request processed.
This function is now given a SIP request structure which it can use
to copy the incoming request into. This reduces the amount of memory
allocations done since the internal allocated components are reused
between packets and also ensures the SIP request structure is
deinitialized when the TLS connection is torn down.
(closes issue ASTERISK-20763)
Reported by: deti
2012-11-06 Asterisk Development Team <asteriskteam@digium.com>
* Asterisk 1.8.19.0-rc1 Released.
2012-11-05 22:50 +0000 [r375893] Matthew Jordan <mjordan@digium.com>
* bridges/bridge_softmix.c, include/asterisk/timing.h,
res/res_musiconhold.c, channels/chan_iax2.c,
res/res_fax_spandsp.c, res/res_timing_kqueue.c, main/timing.c,
main/channel.c, res/res_timing_pthread.c, res/res_timing_dahdi.c,
res/res_timing_timerfd.c: Refactor ast_timer_ack to return an
error and handle the error in timer users Currently, if an
acknowledgement of a timer fails Asterisk will not realize that a
serious error occurred and will continue attempting to use the
timer's file descriptor. This can lead to situations where errors
stream to the CLI/log file. This consumes significant resources,
masks the actual problem that occurred (whatever caused the timer
to fail in the first place), and can leave channels in odd
states. This patch propagates the errors in the timing resource
modules up through the timer core, and makes users of these
timers handle acknowledgement failures. It also adds some
defensive coding around the use of timers to prevent using bad
file descriptors in off nominal code paths. Note that the patch
created by the issue reporter was modified slightly for this
commit and backported to 1.8, as it was originally written for
Asterisk 10. (issue ASTERISK-20032) Reported by: Jeremiah Gowdy
patches: jgowdy-timerfd-6-22-2012.diff uploaded by Jeremiah Gowdy
(license 6358)
2012-11-05 21:36 +0000 [r375862] Richard Mudgett <rmudgett@digium.com>
* main/loader.c: Add safety NULL pointer check in module user
references. Made __ast_module_user_remove() check for NULL
pointers. ........ Merged revision 375860 from C.3
2012-11-04 03:06 +0000 [r375727-375800] Matthew Jordan <mjordan@digium.com>
* main/manager.c: Don't attempt to purge sessions when no sessions
exist Manager's tcp/tls objects have a periodic function that
purge old manager sessions periodically. During shutdown, the
underlying container holding those sessions can be disposed of
and set to NULL before the tcp/tls periodic function is stopped.
If the periodic function fires, it will attempt to iterate over a
NULL container. This patch checks for whether or not the sessions
container exists before attempting to purge sessions out of it.
If the sessions container is NULL, we simply return. Note that
this error was also caught by the Asterisk Test Suite.
* main/manager.c: Properly clean up manager resources on exit This
patch does two things: 1) It properly unregisters the manager CLI
commands 2) It cleans up AMI users on exit. Prior to this patch,
the AMI users were not being disposed of properly, resulting in a
memory leak. (closes issue ASTERISK-20646) Reported by: Corey
Farrell patches: manager_shutdown.patch uploaded by Corey Farrell
(license 5909)
* main/xmldoc.c: Fix memory leak when unloading XML documentation
This patch is a modified version of a patch originally committed
for the Asterisk 11 branch in r375756. A portion of that patch,
that fixed the memory leak during unloading XML documentation,
applies to branches 1.8 and 10 as well. The patch for this issue
was modified for these two branches. (issue ASTERISK-20648)
Reported by: Corey Farrell Tested by: mjordan patches:
xmldoc-memory_leak.patch uploaded by Corey Farrell (license 5909)
* main/cdr.c: Prevent multiple CDR batches from conflicting when
scheduling the CDR write The Asterisk Test Suite caught an error
condition where a scheduled CDR batch write can be deleted twice
if two channels attempt to post their CDRs at the same time. The
batch CDR mutex is locked while the CDRs are appended to the
current batch list; however, it is unlocked prior to actually
scheduling the CDR write. As such, two threads can attempt to
remove the currently scheduled batch write at the same time,
resulting in an assertion error. This patch extends the time that
the mutex is locked to encompass actually scheduling the write.
This prevents two threads from unscheduling the currently
scheduled write at the same time.
2012-11-03 03:11 +0000 [r375698] Andrew Latham <lathama@gmail.com>
* README, include/asterisk/doxyref.h: Doxygen Updates Replace links
to missing text files removed in the 1.6.x series with links to
the wiki. Doxygen can handle URLs fine, don't atempt to quote
them. Also update the wiki link in the Readme to get everyone on
the same page. (issue ASTERISK-20259)
2012-11-02 20:48 +0000 [r375625-375658] Richard Mudgett <rmudgett@digium.com>
* main/channel.c, channels/chan_misdn.c, main/ccss.c: Things don't
need to be that const.
* channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h: Multiple
revisions 375519-375524 ........ r375519 | rmudgett | 2012-10-30
16:06:15 -0500 (Tue, 30 Oct 2012) | 11 lines chan_misdn: Timer
primitives must be handled first. The frm->addr is a different
"address space" than the stack/instance address of other Lx
primitives. The test for B channel instance address could fail.
Patches: patch01_timers.diff (license #6372) patch uploaded by
Guenther Kelleter JIRA ABE-2888 ........ r375520 | rmudgett |
2012-10-30 16:14:58 -0500 (Tue, 30 Oct 2012) | 10 lines
chan_misdn: Free memory in error paths and other memory leaks.
The one line commented with BUG is not easily fixable because
there is no de-init function one can call. Patches:
patch02_memory.diff (license #6372) patch uploaded by Guenther
Kelleter JIRA ABE-2888 ........ r375521 | rmudgett | 2012-10-30
16:38:41 -0500 (Tue, 30 Oct 2012) | 14 lines chan_misdn: ISDN NT
L2 de-establish/establish * An NT-PTMP cannot de/establish L2
since it doesn't know the TEIs. * On NT-PTP L2 is started when L1
is finally active in handle_l1. * L2 deactivation logging
cleanup. * L2 aggregate link status is unknown for NT-PTMP, show
as "UNKN". * Removed unused functions and code for L2 handling.
Patches: patch03_L2estab.diff (license #6372) patch uploaded by
Guenther Kelleter Modified JIRA ABE-2888 ........ r375522 |
rmudgett | 2012-10-30 16:56:14 -0500 (Tue, 30 Oct 2012) | 22
lines chan_misdn: Fix broken upper_id/lower_id usage. Sending PH
prim via lower_id layer (3 or 1) simply does not work. For TE (3)
it returns an error (len=-6) which is not evaluated by
handle_l1(), so the L1 layer status ends up wrong. Instead PH
must be sent via L4, only then does it reach L1 without an error
message. And NT PH prims only reach L1 when they are sent to
layer 2 id. --> use upper_id to send PH primitives. * Check for
errors in PH_(DE)ACTIVATE | CONFIRM. * Debug messages are
improved. * The lower_id is now not used for anything, except:
Why is lower_id layer deleted when it wasn't created? I removed
this code since it looks very wrong. Patches:
patch04_l1activation.diff (license #6372) patch uploaded by
Guenther Kelleter JIRA ABE-2888 ........ r375523 | rmudgett |
2012-10-30 17:29:15 -0500 (Tue, 30 Oct 2012) | 31 lines
chan_misdn: Fix loss of B channels if L1 is down. If you make 2
calls out an NT PTMP port which is not connected to any phone,
the B channel associated with that call becomes unusable until
Asterisk is restarted. The problem is the EVENT_SETUP is queued
when L1 is not up in misdn_lib_send_event(). If L1 cannot be
activated the event won't be dequeued. It gets even worse when
the call is hung up. The queued EVENT_SETUP will be overwritten
by an EVENT_DISCONNECT. The reserved B channel then will never be
freed. If later someone connects a phone to the port, L1 will
eventually activate and the queued EVENT_DISCONNECT is sent down
the stack. However, it is ignored because it is the wrong call
state. The real fix would be that activation and queueing for a
new SETUP is done by the NT stack. But since it doesn't, the
workaround must be removed because it doesn't always work. Fix:
The event is no longer queued but immediately sent to the stack.
If L1 cannot be activated, the L3 state machine that was started
by the EVENT_SETUP will do its work, i.e. a timeout will release
the B channel properly. The SETUP possibly cannot be sent the
first time but is resent by T303 in case L1 could be activated.
Patches: patch05_bchan-loss.diff (license #6372) patch uploaded
by Guenther Kelleter Modified JIRA ABE-2888 ........ r375524 |
rmudgett | 2012-10-30 18:26:05 -0500 (Tue, 30 Oct 2012) | 13
lines chan_misdn: Remove some calls to exit(). Try proper cleanup
when something goes wrong in misdn_lib_init(). Especially do not
call exit()! * Fix memory leak because stack_destroy() does not
free the stack struct. Patches: patch06_cleanup-init.diff
(license #6372) patch uploaded by Guenther Kelleter Modified JIRA
ABE-2888 ........ Merged revisions 375519-375524 from
https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
2012-11-02 16:58 +0000 [r375594] Michael L. Young <elgueromexicano@gmail.com>
* channels/chan_sip.c: Fix Wrong Result In Debug Message For SDP
Origin Processing While looking at some debug logs, I noticed
that it was being reported that the SDP origin line was
unsupported or failed. Upon looking into this on my local
machine, I found that I too was getting this debug message yet
everything seemed to be getting processed properly. What was
discovered is, that, the variable to determine what is displayed
in the debug message for the SDP line that was processed, was not
being set for the origin line when the result was successful.
This patch fixes this and was tested on local machine.
2012-10-31 14:23 +0000 [r375528] Matthew Jordan <mjordan@digium.com>
* res/res_calendar_ews.c: Properly extract the Body information of
an EWS calendar item Unlike all other calendar modules,
res_calendar_ews fails to extract the Body information for a
calendar item. This is due, in part, to a quirk in the schema in
the XML - not only does a CalendarItem contain a Body element,
but the CalendarItem exists as a descendant of a different Body
element. The neon parser was erroneously skipping all Body
elements. This patch fixes that by bypassing Body elements that
are not a child of CalendarItem, and parsing the Body element out
if it is a child. Note that the original patch by Terry Wilson
only needed slight modifications to make it properly pull the
Body information out; as such, while I've linked to the patch
that I uploaded for Dmitry, I've attributed the patch to Terry.
(closes issue ASTERISK-19738) Reported by: Dmitry Burilov Tested
by: Dmitry Burilov patches: calendar_ews_body_2012_10_29.diff
uploaded by Terry Wilson (license 6283)
2012-10-30 18:48 +0000 [r375484] Jonathan Rose <jrose@digium.com>
* apps/app_mixmonitor.c: mixmonitor: Add a test event This test
event is being used to fix the mixmonitor_audiohook_inherit test.
2012-10-30 02:07 +0000 [r375450] Matthew Jordan <mjordan@digium.com>
* apps/app_queue.c: Ensure that the Queue application tracks busy
members in off nominal situations There are a few code paths
where the Queue application fails to count a paused or in use
queue member as being 'busy'. This can cause callers to get stuck
in the Queue until a paused agent unpauses themselves. (closes
issue ASTERISK-20623) Reported by: Bryan Walters patches:
app_queue.patch uploaded by Bryan Walters (license 5851)
2012-10-29 21:01 +0000 [r375415] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Prevent resetting of NATted realtime peer
address on reload. If a "sip reload" is issued for a SIP peer,
then his IP address will be cleared, thus resulting in forgetting
the public IP address. Asterisk will then attempt to route SIP
traffic to the private IP address. The fix here is to make "sip
reload" ignore realtime peers when "host = dynamic" is spotted.
Realtime peers can now only have their IP address reset if they
have gone from being not dynamic to being dynamic. (closes issue
ASTERISK-18203) reported by daren ferreira (closes issue
ASTERISK-20572) reported by JoshE Patches: fix_nat_realtime.diff
uploaded by JoshE (license #6075)
2012-10-29 19:26 +0000 [r375361-375388] Richard Mudgett <rmudgett@digium.com>
* main/features.c: Fix the Park 'r' option when a channel parks
itself. When a channel uses the Park appliation to park itself
with the 'r' option, the channel hears music-on-hold instead of
the requested ringing. * Added a missing check for the 'r' option
when a channel parks itself. (closes issue ASTERISK-19382)
Reported by: James Stocks Patches by: dsessions Review:
https://reviewboard.asterisk.org/r/2148/
* channels/chan_dahdi.c: chan_dahdi: Fix segfault dereferencing a
NULL tech_pvt. The tech support customer was using the AMI
Redirect action shortly after a call was placed. While the
channel tried to do an ast_read(), the masquerade resulting from
the channel redirect took place. The masquerade in the middle of
the ast_read() resulted in the segfault. (closes issue AST-1025)
Reported by: Trey Blancher Patches: jira_ast_1025_v1.8_v2.patch
(license #5621) patch uploaded by rmudgett
2012-10-23 16:20 +0000 [r375272-375325] Jonathan Rose <jrose@digium.com>
* contrib/scripts/ast_tls_cert: ast_tls_cert script: Better
response for various exit conditions to openssl (closes issue
ASTERISK-20260) Reported by: Daniel O'Connor Patches:
ast_tls_cert-update.diff uploaded by Daniel O'Connor (license
6419)
* main/app.c: core: Fix a memory leak in app.c from an early return
ast_app_group_match_get_count allocates memory with the regcomp
function and we previously forgot to free it when bailing out due
to a regex compilation failure against category. (closes issue
AST-1018) Reported by: Guenther Kelleter Patches:
regcomp_memleak.diff uploaded by Guenther Kelleter (license 6372)
* codecs/gsm/src/code.c: GSM: Fix encoding problems with GSM
(closes issue ASTERISK-20457) Reported by: Richard Miller
Patches: code.patch uploaded by Richard Miller (license 5685)
2012-10-18 21:36 +0000 [r375216-375244] Jonathan Rose <jrose@digium.com>
* UPGRADE.txt: Correct version number in Upgrade.txt release notes
pertaining to queue order Showed 1.8.17 to 1.8.18, needs to be
1.8.18 to 1.8.19
* UPGRADE.txt: app_queue: add upgrade notes for 375216 Adds notes
describing behavioral changes to rrmemory strategy caused by
375216 (issue AST-989) Reported by: Thomas Arimont
* apps/app_queue.c: app_queue: Make ordering of rrmemory/rrordered
persist over add/remove members Prior to this patch, adding,
removing or reloading members to rrmemory would cause the order
to become completely jumbled. Now it behaves more or less like
rrordered other than the fact that it stores the members on a
hash table rather than a linked list. This patch also prevents
removal of members and member reloads from jumbling rrordered
queues. (issue AST-989) Reported by: Thomas Arimont Review:
https://reviewboard.asterisk.org/r/2164/
2012-10-18 19:42 +0000 [r375189] Richard Mudgett <rmudgett@digium.com>
* makeopts.in, Makefile, build_tools/make_version, configure,
configure.ac: build_tools: Allow Asterisk to report git SHAs in
version string. Make git more attractive for managing
work-in-progress. Especially convenient when a potential patch
set needs to be tested on multiple platforms since one can use
git to keep all the test environments in sync independent of a
subversion server. Now the Asterisk version will show the exact
git SHA5 that was used when building (still appended by "M" if
there are local modifications) from a git clone of the Asterisk
repository so the developer can more easily know what is actually
under test. You will now get this: $ asterisk -V Asterisk
GIT-1698298 Instead of this: $ asterisk -V Asterisk
UNKNOWN__and_probably_unsupported This has zero impact for those
not using git with the exception of an extra test in the
configure script to gather git's path. This is necessary to
prevent "sudo make install" from failing since git may not be in
the path in make's shell environment. (closes issue
ASTERISK-20483) Reported by: Shaun Ruffell Patches:
0001-build_tools-Allow-Asterisk-to-report-git-SHAs-in-ver.patch
(license #5417) patch uploaded by Shaun Ruffell Modified
2012-10-17 18:55 +0000 [r375146] Kinsey Moore <kmoore@digium.com>
* main/tcptls.c: Ensure Asterisk fails TCP/TLS SIP calls when
certificate checking fails When placing a call to a TCP/TLS SIP
endpoint whose certificate is not signed by a configured CA
certificate, Asterisk would issue a warning and continue to
process the call as if there was not an issue with the
certificate. Asterisk now properly fails the call if the
certificate fails verification or if the certificate does not
exist when certificate checking is enabled (the default
behavior). (closes issue ASTERISK-20559) Reported by: kmoore
Review: https://reviewboard.asterisk.org/r/2163/
2012-10-16 21:41 +0000 [r375074-375111] Walter Doekes <walter+asterisk@wjd.nu>
* channels/chan_sip.c: Fixes to the fd-oriented SIP TCP reads.
Don't crash on large user input. Allow SIP headers without space.
Optimize code a bit. Review:
https://reviewboard.asterisk.org/r/2162
* channels/chan_sip.c: Update sip_request_call SIP dial string
documentation. This was missed when merging review r1859.
2012-10-16 19:13 +0000 [r375059] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
* contrib/scripts/autosupport: autosupport: fix bashism '==' is
bashism (bashspecific, fails when dash is /bin/sh). Anyway, a
'case' works better there. (closes issue ASTERISK-20567) Reported
by: Tzafrir Cohen
2012-10-15 21:00 +0000 [r375025] Mark Michelson <mmichelson@digium.com>
* include/asterisk/strings.h, channels/chan_iax2.c,
apps/app_dial.c, main/ccss.c: Fix some potential misuses of
ast_str in the code. Passing an ast_str pointer by value that
then calls ast_str_set(), ast_str_set_va(), ast_str_append(), or
ast_str_append_va() can result in the pointer originally passed
by value being invalidated if the ast_str had to be reallocated.
This fixes places in the code that do this. Only the example in
ccss.c could result in pointer invalidation though since the
other cases use a stack-allocated ast_str and cannot be
reallocated. I've also updated the doxygen in strings.h to
include notes about potential misuse of the functions mentioned
previously. Review: https://reviewboard.asterisk.org/r/2161
2012-10-14 08:59 +0000 [r374977] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
* config.guess, config.sub: Update config.guess and config.sub:
2012-10-10 Update config.guess and config.sub to revision
fb456b34ef4aa02b95dc6be69aaa66fa94a844fb from the
savannah.gnu.org git repo. Adds support for e.g. aarch64 (ARM
64bit). config.guess:timestamp='2012-09-25'
config.sub:timestamp='2012-10-10'
2012-10-12 15:57 +0000 [r374905] Mark Michelson <mmichelson@digium.com>
* include/asterisk/tcptls.h, main/tcptls.c, channels/chan_sip.c: Do
not use a FILE handle when doing SIP TCP reads. This is used to
solve an issue where a poll on a file descriptor does not
necessarily correspond to the readiness of a FILE handle to be
read. This change makes it so that for TCP connections, we do a
recv() on the file descriptor instead. Because TCP does not
guarantee that an entire message or even just one single message
will arrive during a read, a loop has been introduced to ensure
that we only attempt to handle a single message at a time. The
tcptls_session_instance structure has also had an overflow buffer
added to it so that if more than one TCP message arrives in one
go, there is a place to throw the excess. Huge thanks goes out to
Walter Doekes for doing extensive review on this change and
finding edge cases where code could fail. (closes issue
ASTERISK-20212) reported by Phil Ciccone Review:
https://reviewboard.asterisk.org/r/2123
2012-10-11 15:42 +0000 [r374843] Matthew Jordan <mjordan@digium.com>
* main/cdr.c: Fix incorrect billing duration reported when batch
mode is enabled Similar to r369351, the billing duration can be
skewed when batch mode is enabled. This happened much more rarely
than the duration, as it only occured when the call was answered
(thereby indicating an actual answer time) and immediately hung
up on (indicating a billsec of 0). Since a billing time of '0'
can either mean that the call immediately ended or that the CDR
was improperly answered, we have to use additional information to
know whether or not we can trust the CDR billsec value. Prior to
this patch, we looked to see if we had a valid answer time. If we
did, and billsec was zero, we used the current time to calculate
what billsec value we could from the CDR being written. If batch
mode is enabled, this will incorrectly report a billsec value
being much greater than the actual duration of the call. Instead
of relying on the presence of an answer time to know whether or
not we can re-calculate the billsec for the CDR, we now also use
the presence of the CDR's end time to know if we need to
re-calculate or whether we can trust the billsec value that we
have. This prevents erroneous jumps in the billsec value, while
still making sure that in the worst case, some billing time will
be calculated. (closes issue AST-1016) Reported by: Thomas
Arimont Tested by: Thomas Arimont
2012-10-10 20:52 +0000 [r374686-374802] Richard Mudgett <rmudgett@digium.com>
* apps/app_queue.c: app_queue: Made pass connected line updates
from the caller to ringing queue members. Party A calls Party B
Party B puts Party A on hold. Party B calls a queue. Ringing
queue member D sees Party B identification. Party B transfers
Party A to the queue. Queue member D does not get a connected
line update for Party A. Queue member D answers the call and
still sees Party B information. However, if Party A later
transfers the call to Party C then queue member D gets a
connected line update for Party C. * Made pass connected line
updates from the caller to queue members while the queue members
are ringing. (closes issue AST-1017) Reported by: Thomas Arimont
(closes issue ABE-2886) Reported by: Thomas Arimont Tested by:
rmudgett ........ Merged revisions 374801 from
https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
* main/pbx.c: Fix execution of 'i' extension due to uninitialized
variable. The fix for ASTERISK-18243 added code that could
potentially use dst_exten[] uninitialized. As a result the 'i'
exten may not be executed when it should. (closes issue
ASTERISK-20455) Reported by: Richard Miller Patches:
pbx-1.8.16.0.diff (license #5685) patch uploaded by Richard
Miller Made some cosmetic modifications.
* configs/chan_dahdi.conf.sample: dahdi.conf.sample: Add
description for "buffers" setting. This contains an edited
version of the patch originally created by John Bigelow. (closes
issue ASTERISK-14435) Reported by: John Bigelow Patches:
buffers.patch (license #5091) patch uploaded by John Bigelow
0001-dahdi.conf.sample-Add-description-for-buffers-settin.patch
(license #5417) patch uploaded by Shaun Ruffell Modified
* pbx/pbx_spool.c: Fix deletion of unopenable spool files. If
scan_service() cannot open the spool file, it logs a message
saying that it will delete the file and calls remove_from_queue()
to do it. However, remove_from_queue() fails to delete the spool
file because struct outgoing has not yet been fully initialized.
* Merged allocating a new struct outgoing and init_outgoing()
into new_outgoing(). Allocation is initialization. * Made
apply_outgoing() not initialize the spool filename in struct
outgoing. * Made apply_outgoing() call ast_trim_blanks() and
ast_skip_blanks() rather than manually inlining them. * Reduced
indentation levels in apply_outgoing(). * Fixed a garbled comment
in remove_from_queue(). * Reworked scan_service() to simplify it.
(closes issue ASTERISK-17231) Reported by: David Chappell
Patches: spool_open_failure.diff (license #4997) patch uploaded
by David Chappell Started with this patch.
2012-11-06 Asterisk Development Team <asteriskteam@digium.com>
* Asterisk 1.8.18.0 Released.
2012-10-08 Asterisk Development Team <asteriskteam@digium.com>
* Asterisk 1.8.18.0-rc1 Released.
2012-10-05 20:20 +0000 [r374570-374581] dlee <dlee@localhost>:
* main/manager.c: I've committed too much. Reverting part of
r374570.
* main/manager.c: Improve AMI long line error handling In AMI's
parser, when it receives a long line (> 1024 characters), it
discards that line, but continues to process the message
normally. Typically, this is not a problem because a) who has
lines that long and b) usually a discarded line results in an
invalid message. But if that line is specifying an optional
field, then the message will be processed, you get a 'Response:
Success', but things don't work the way you expected them to.
This patch changes the behavior when a line-too-long parse error
occurs. * Changes the log message to avoid way-too-long (and
truncated anyways) log messages * Adds a 'parsing' status flag to
Response: Success * Sets parsing = MESSAGE_LINE_TOO_LONG if,
well, a line is too long * Responds with an appropriate error if
parsing != MESSAGE_OKAY (closes issue AST-961) Reported by: John
Bigelow Review: https://reviewboard.asterisk.org/r/2142/
2012-10-05 18:20 +0000 [r374536] Richard Mudgett <rmudgett@digium.com>
* channels/misdn/isdn_msg_parser.c, channels/misdn/isdn_lib.c,
channels/misdn/isdn_lib.h, channels/chan_misdn.c: Merged
revisions 374515-374535 from
https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
................ r374515 | rmudgett | 2012-10-04 17:52:36 -0500
(Thu, 04 Oct 2012) | 10 lines chan_misdn: Remove some deadcode *
Made setup_bc() static. Patches: patch1_unused-code.diff (license
#6372) patch uploaded by Guenther Kelleter Modified JIRA ABE-2882
................ r374516 | rmudgett | 2012-10-04 18:01:01 -0500
(Thu, 04 Oct 2012) | 7 lines chan_misdn: Remove unused bchan
states Patches: patch2_unused-states.diff (license #6372) patch
uploaded by Guenther Kelleter JIRA ABE-2882 ................
r374517 | rmudgett | 2012-10-04 18:17:51 -0500 (Thu, 04 Oct 2012)
| 16 lines chan_misdn: Remove unnecessary null pointer checks and
checks for stack->nt * cleanup_bc() is always called with valid
bc (or it would've crashed before). * Value of stack->nt is known
in advance at some places. * Rename handle_event() to
handle_event_te(), handle_frm() to handle_frm_te(). Patches:
patch3_checks.diff (license #6372) patch uploaded by Guenther
Kelleter Modified JIRA ABE-2882 ................ r374518 |
rmudgett | 2012-10-04 18:21:59 -0500 (Thu, 04 Oct 2012) | 7 lines
chan_misdn: Fix spelling in log messages Patches:
patch4_spelling.diff (license #6372) patch uploaded by Guenther
Kelleter JIRA ABE-2882 ................ r374519 | rmudgett |
2012-10-04 18:31:59 -0500 (Thu, 04 Oct 2012) | 15 lines
chan_misdn: Don't cleanup a bc twice. In handle_frm_te() after
calling misdn_lib_send_event(bc, EVENT_RELEASE_COMPLETE) bc is
emptied, cleaned and set not in use, although
misdn_lib_send_event() already did the same. This is bad. When
it's not in use we are not allowed to touch it. * Moved log
message in front of the resulting actions and fixed it to match
the case. Patches: patch5_bccleanup.diff (license #6372) patch
uploaded by Guenther Kelleter JIRA ABE-2882 ................
r374520 | rmudgett | 2012-10-04 18:43:56 -0500 (Thu, 04 Oct 2012)
| 12 lines chan_misdn: Fix memory leaks, bc, chan not cleaned up
etc., really bad stuff. * Fix return codes of cb_events() for
EVENT_SETUP to use caller's cleanup mechanisms. * Move
cl_queue_chan() call after bearer check. Patches:
patch6_leaks.diff (license #6372) patch uploaded by Guenther
Kelleter JIRA ABE-2882 ................ r374521 | rmudgett |
2012-10-04 18:48:38 -0500 (Thu, 04 Oct 2012) | 11 lines
chan_misdn: We must initialize cause on sending a DISCONNECT. We
must initialize cause on sending a DISCONNECT, so it is later
correctly indicated to ast_channel in case the answer
(RELEASE/RELEASE_COMPLETE) does not include one. Patches:
patch7_hangupcause.diff (license #6372) patch uploaded by
Guenther Kelleter JIRA ABE-2882 ................ r374522 |
rmudgett | 2012-10-04 19:03:56 -0500 (Thu, 04 Oct 2012) | 7 lines
chan_misdn: Remove unused code for upqueue Patches:
patch8_unused-upqueue.diff (license #6372) patch uploaded by
Guenther Kelleter JIRA ABE-2882 ................ r374523 |
rmudgett | 2012-10-04 19:11:50 -0500 (Thu, 04 Oct 2012) | 7 lines
chan_misdn: Improve debugging (port number, messages fixed, dups
removed) Patches: patch9_debug.diff (license #6372) patch
uploaded by Guenther Kelleter JIRA ABE-2882 ................
r374533 | rmudgett | 2012-10-05 12:17:18 -0500 (Fri, 05 Oct 2012)
| 8 lines chan_misdn: Better debug: we can print_bc_info even if
there's no ast leg. Patches: patch10_debug-bc-2.diff (license
#6372) patch uploaded by Guenther Kelleter Modified. JIRA
ABE-2882 ................ r374534 | rmudgett | 2012-10-05
12:34:10 -0500 (Fri, 05 Oct 2012) | 16 lines chan_misdn:
setup_bc() is called too early for an incoming SETUP on TE. This
prevents the B channel from being setup for HDLC mode when
requested by the bearer capability and config option hdlc=yes. It
violates ETS300102 Ch.5.2.3.2: "The user, in any case, must not
connect to the channel until a CONNECT ACKNOWLEDGE message has
been received." * Call setup_bc() on receipt of
CONNECT_ACKNOWLEGDE for PTMP, and on first response to SETUP for
PTP. Patches: abe-2881-2.diff (license #6372) patch uploaded by
Guenther Kelleter Modified. JIRA ABE-2881 ................
r374535 | rmudgett | 2012-10-05 12:41:05 -0500 (Fri, 05 Oct 2012)
| 2 lines chan_misdn: Remove some more deadcode. ................
2012-10-04 20:15 +0000 [r374475-374479] Alec L Davis <sivad.a@paradise.net.nz>
* CHANGES, main/dsp.c, configs/dsp.conf.sample: dsp.c User
Configurable DTMF_HITS_TO_BEGIN and DTMF_MISSES_TO_END Instead of
a recompile, allow values to be adjusted in dsp.conf For binary
distributions allows easy adjustment for wobbly GSM calls, and
other reasons. Defaults to DTMF_HITS_TO_BEGIN=2 and
DTMF_MISSES_TO_END=3 (closes issue ASTERISK-17493) Tested by:
alecdavis alecdavis (license 585) Review
https://reviewboard.asterisk.org/r/2144/
* main/dsp.c: dsp.c fix incorrect DTMF Digit_Duration. it's always
short by 'hits_to_begin*DTMF_GSIZE', or 25.5ms if hitstobegin=2
(issue ASTERISK-16003) Tested by: alecdavis alecdavis (license
585) Review https://reviewboard.asterisk.org/r/2145/
2012-10-04 17:39 +0000 [r374456] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: Fix a regression from direct media ACLs
where the directrtpsetup option no longer works. A check was
added for direct media ACLs that immediately forbid remote
bridging if there was no bridged channel. This caused
directrtpsetup to no longer function as it needs this information
before bridging actually occurs. Logic has now been adjusted so
if there is no bridged channel a remote bridge will still be
attempted. (closes issue ASTERISK-20511) Reported by: kristoff
Review: https://reviewboard.asterisk.org/r/2146/
2012-10-04 15:25 +0000 [r374426] dlee <dlee@localhost>:
* main/db.c, res/res_agi.c: Fix DBDelTree error codes for AMI, CLI
and AGI The AMI DBDelTree command will return Success/Key tree
deleted successfully even if the given key does not exist. The
CLI command 'database deltree' had a similar problem, but was
saved because it actually responded with '0 database entries
removed'. AGI had a slightly different error, where it would
return success if the database was unavailable. This came from
confusion about the ast_db_deltree retval, which is -1 in the
event of a database error, or number of entries deleted
(including 0 for deleting nothing). * Adds a Doxygen comment to
process_db_keys explaining its retval * Changed some poorly named
res variables to num_deleted * Specified specific errors when
calling ast_db_deltree (database unavailable vs. entry not found
vs. success) * Fixed similar bug in AGI database deltree, where
'Database unavailable' results in successful result (closes issue
AST-967) Reported by: John Bigelow Review:
https://reviewboard.asterisk.org/r/2138/
2012-10-04 04:39 +0000 [r374365-374384] Alec L Davis <sivad.a@paradise.net.nz>
* CHANGES, main/dsp.c, configs/dsp.conf.sample: dsp.c User
configuration of DTMF_NORMAL_TWIST and DTMF_REVERSE_TWIST values
Asterisk's DTMF Specifications are based on AT&T specs, which may
not be compatible in other countries. Various countries have
different specifications for the maximum power level differences
between the DTMF low group and high group of frequencies. Power
level difference between frequencies for different
Administrations/RPOAs NTT = Max. 5 dB AT&T = 4dB(reverse) to
8dB(normal) Danish = Max. 6 dB Australian = Max. 10 dB Brazilian
= Max. 9 dB ETSI = Max. 6 dB from ETSI ES 201 235-3 V1.3.1
(2006-03) Now allow 4 variables to be individually configured in
dsp.conf, with reasonable min/max of 2dB to 20dB. Default is AT&T
specifications Add's the following variables to dsp.conf
;dtmf_normal_twist=6.31 ;dtmf_reverse_twist=2.51
;relax_dtmf_normal_twist=6.31 ;relax_dtmf_reverse_twist=3.98
(closes issue ASTERISK-20442) Reported by: tbsky Tested by:
tbsky,alecdavis alecdavis (license 585) Review
https://reviewboard.asterisk.org/r/2141/
* main/dsp.c: _dsp_init: bring inline with trunk preparation for
clean merge of DTMF TWIST patch No functional changes, just
style. alecdavis (license 585) Reported by: Alec Davis Tested by:
alecdavis related https://reviewboard.asterisk.org/r/2141
2012-10-04 02:09 +0000 [r374177-374335] Matthew Jordan <mjordan@digium.com>
* res/res_jabber.c: Check for presence of buddy in info/dinfo
handlers The res_jabber resource module uses the ASTOBJ library
for managing its ref counted objects. After calling
ASTOBJ_CONTAINER_FIND to locate a buddy object, the pointer to
the object has to be checked to see if the buddy existed. Prior
to this patch, the buddy object was not checked for NULL; with
this patch in both aji_client_info_handler and aji_dinfo_handler
the pointer is checked before used and, if no buddy object was
found, the handlers return an error code. This patch does not
take the approach that our JID can be used to log in from another
resource. If that approach is desired, an improvement could be
made to this patch to create the buddy on the fly. This patch
seeks only to prevent Asterisk from crashing. Note that multiple
people have proposed patches for this issue; the patch being
committed here is based on those. (closes issue ASTERISK-19532)
Reported by: Karsten Wemheuer Tested by: Byron Clark patches:
fix-jabber uploaded by Karsten Wemheuer (license #5930)
xmpp_no_crash_with_ejabberd.patch uploaded by Byron Clark
(license #6157) (closes issue ASTERISK-19557) Reported by:
ulugutz
* main/ccss.c: Destroy the generic_monitors container after the
core_instances in ccss For each item in core_instances disposed
of in the shutdown of ccss, any generic monitor instances
referenced by the objects will be removed from generic_monitors
during their destruction. Hilarity ensues if generic_monitors no
longer exists. Thanks to the Asterisk Test Suite's generic_ccss
test for complaining loudly when it ran into this.
* main/asterisk.c: Ensure Shutdown AMI event is still fired during
Asterisk shutdown Richard pointed out that having the manager
dispose of itself gracefully during shutdown meant that the
Shutdown event will no longer get fired. This patch moves the AMI
event just prior to running the atexit callbacks.
* main/event.c, main/taskprocessor.c, res/res_musiconhold.c,
main/cel.c, main/indications.c, main/channel.c, main/data.c,
main/pbx.c, main/manager.c, main/ccss.c, main/features.c: Fix a
variety of ref counting issues This patch resolves a number of
ref leaks that occur primarily on Asterisk shutdown. It adds a
variety of shutdown routines to core portions of Asterisk such
that they can reclaim resources allocate duringd initialization.
Review: https://reviewboard.asterisk.org/r/2137
2012-10-01 16:45 +0000 [r374108] Sean Bright <sean@malleable.com>
* tests/test_db.c, apps/app_queue.c, main/db.c,
include/asterisk/astdb.h: app_queue: Support persisting and
loading of long member lists. Greenlight in #asterisk brought up
that he was receiving an error message "Could not create
persistent member string, out of space" when running app_queue in
Asterisk 10. dump_queue_members() made an assumption that 8K
would be enough to store the generated string, but with queues
that have large member lists this is not always the case. This
patch removes the limitation and uses ast_str instead of a fixed
sized buffer. The complicating factor comes from the fact that
ast_db_get requires a buffer and buffer size argument, which
doesn't let us pull back more than what we pass in, so I
introduced a new ast_db_get_allocated() which returns an
ast_strdup()'d copy of the value from astdb. As an aside, I did
some testing on the maximum size of data that we can store in the
BDB library we distribute and was able to store a 10MB string and
retrieve it with no problems, so I feel this is a safe patch.
Review: https://reviewboard.asterisk.org/r/2136/
2012-09-28 19:03 +0000 [r374032] Jonathan Rose <jrose@digium.com>
* res/res_jabber.c: res_jabber: Remove CLI command 'jabber test'
The opinion of development was that it is both improper to have
Matt's personal email address used in the source and that the
command wouldn't be useful without it. (closes issue AST-467)
Reported by: Malcolm Davenport
2012-09-28 12:14 +0000 [r373989] Joshua Colp <jcolp@digium.com>
* res/res_agi.c: Update documentation to make it explicit that
"stream file" will not restart musiconhold. (issue
ASTERISK-17367) Reported by: oej
2012-09-27 22:08 +0000 [r373945] Richard Mudgett <rmudgett@digium.com>
* apps/app_senddtmf.c: Fix SendDTMF crash and channel reference
leak using channel name parameter. The SendDTMF channel name
parameter has two issues. 1) Crashes if the channel name does not
exist. 2) Leaks a channel reference if the channel is the current
channel. Problem introduced by ASTERISK-15956. * Updated SendDTMF
documentation. * Renamed app to senddtmf_name and tweaked the
type.
2012-09-27 16:49 +0000 [r373878-373909] Joshua Colp <jcolp@digium.com>
* main/loader.c: loader: Ensure dependent modules are properly
initialized. If an Asterisk module specifies a dependency in
ast_module_info.nonoptreq, it is possible for Asterisk to skip
calling the modules's .load function. Asterisk was loading and
linking the module via load_dynamic_module() but was not adding
the module to the resource_heap. Therefore the module was not
initialized based on it's priority along with the other modules
in the heap. Now use load_resource() instead of
load_dynamic_module() for non-optional requirement. This will add
the module to the resource_heap so the module can be properly
initialized in the correct order. This is required if there are
any module global data structures initialized in the .load()
callback for the module on platforms which do not support weak
references. (issue ASTERISK-20439) Reported by: sruffell Patches:
0001-loader-Ensure-dependent-modules-are-properly-initial.patch
uploaded by sruffell (license 5417)
* channels/chan_local.c: Fix an issue where Local channels dialed
by app_queue are considered in use immediately. The chan_local
channel driver returns a device state of in use even if a created
Local channel has not yet been dialed. This fix changes the logic
to return a state of not in use until the channel itself has been
dialed. (closes issue ASTERISK-20390) Reported by: tim_ringenbach
Review: https://reviewboard.asterisk.org/r/2116/
2012-09-26 21:11 +0000 [r373848] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Move handling of 408 response so there is no
misleading warning message. (closes issue ASTERISK-20060)
Reported by: Walter Doekes
2012-09-26 18:04 +0000 [r373815] Richard Mudgett <rmudgett@digium.com>
* apps/app_meetme.c: Fixed meetme tab completion and command
documentation. * Removed unnecessary case sensitivity in meetme
list, lock, unlock, mute, unmute, and kick commands. * Separated
meetme lock/unlock, mute/unmute, and kick commands into their own
registered commands to simplify tab completion and parameter
checking. meetme_lock_cmd(), meetme_mute_cmd(), and
meetme_kick_cmd() * Simplified meetme_show_cmd() (closes issue
AST-1006) Reported by: John Bigelow Tested by: rmudgett
2012-09-25 23:07 +0000 [r373735-373773] Mark Michelson <mmichelson@digium.com>
* main/say.c: Fix saying of date in Dutch. The Dutch say the date
before the month. (closes issue ASTERISK-20353) Reported by: Teun
Ouwehand
* configs/agents.conf.sample, channels/chan_agent.c: Remove dead
code and documentation for nonexistent feature. multiplelogin was
removed from chan_agent back in 1.6.0 when AgentCallbackLogin()
was removed. (closes issue AST-948) reported by Steve Pitts
* apps/app_voicemail.c: Fix error where improper IMAP greetings
would be deleted. (closes issue ASTERISK-20435) Reported by:
fhackenberger Patches: asterisk-20435-imap-del-greeting.diff
uploaded by Michael L. Young (License #5026) (with suggested
modification made by me)
2012-09-25 20:10 +0000 [r373705] Joshua Colp <jcolp@digium.com>
* channels/chan_local.c: Fix T.38 support when used with chan_local
in between. Users of the T.38 API can indicate
AST_T38_REQUEST_PARMS on a channel to request that the channel
indicate a T.38 negotiation with the parameters present on the
channel. The return value of this indication is expected to be
AST_T38_REQUEST_PARMS upon success but with chan_local involved
this could never occur. This fix changes chan_local to always
return AST_T38_REQUEST_PARMS for this situation. If the
underlying channel technology on the other side does not support
T.38 this would have been determined ahead of time using
ast_channel_get_t38_state and an indication would not occur.
(closes issue ASTERISK-20229) Reported by: wdoekes Patches:
ASTERISK-20229.patch uploaded by wdoekes (license 5674) Review:
https://reviewboard.asterisk.org/r/2070/
2012-09-25 19:32 +0000 [r373666-373702] Kinsey Moore <kmoore@digium.com>
* res/res_rtp_asterisk.c: Fix an issue where media would not flow
for situations where the legacy STUN code is in use. The STUN
packets should *not* be blocked by strict RTP. (closes issue
ASTERISK-20415) Reported-by: Michele Cicciotti Patch-by: Josh
Colp (trunk r369817)
* apps/app_queue.c: "show" completion option for "queue" shouldn't
appear twice When tab-completing CLI commands starting with
"queue", "show" appeared twice in the list due to the way that
Asterisk's tab completion functions and the order in which the
commands were registered. The registration order has been altered
to resolve this issue. (closes issue AST-940) Reported-by: Steve
Pitts
2012-09-25 17:21 +0000 [r373652] Terry Wilson <twilson@digium.com>
* configs/sip.conf.sample, channels/sip/include/sip.h,
channels/chan_sip.c: Properly handle UAC/UAS roles for SIP
session timers The SIP session timer mechanism contains a
mandatory 'refresher' parameter (included in the Session-Expires
header) which is used in the session timer offer/answer signaling
within a SIP Invite dialog. It looks like asterisk is
interpreting the uac resp. uas role only as the initial role of
client and server (caller is uac, callee is uas). The standard
rfc 4028 however assigns the client role to the ((RE)-Invite)
requester, the server role to the ((RE)-Invite) responder. This
patch has Asterisk track the actual refresher as "us" or "them"
as opposed to relying on just the configured "uas" or "uac"
properties. (closes issue AST-922) Reported by: Thomas Airmont
Review: https://reviewboard.asterisk.org/r/2118/
2012-09-25 17:18 +0000 [r373618-373640] Richard Mudgett <rmudgett@digium.com>
* codecs/ilbc/iLBC_decode.c, codecs/ilbc/iLBC_encode.c: Fix
valgrind found memcpy issues in codec_ilbc. Valgrind found
codec_ilbc using memcpy instead of memmove for overlapping memory
blocks. (issue ASTERISK-19890) (closes issue ASTERISK-20231)
Reported by: Walter Doekes Patches: ASTERISK-20231.patch (license
#5674) patch uploaded by Walter Doekes
* codecs/Makefile: Make rebuild GSM, ilbc, or lpc10 codecs if the
respective sources change.
2012-09-25 16:15 +0000 [r373617] Jonathan Rose <jrose@digium.com>
* channels/chan_sip.c: chan_sip: Set Quality of Service for video
rtp instance (closes issue ASTERISK-20201) Reported by: ddkprog
Patches: chan_sip.c.diff uploaded by ddkprog (license 6008)
2012-09-25 13:27 +0000 [r373578] Kinsey Moore <kmoore@digium.com>
* configs/res_odbc.conf.sample: Fix documentation for default
username in res_odbc This was previously stated to be "root", but
is actually the name of the context if unspecified. (closes issue
ASTERISK-20258) Reported by: Stefan x
2012-09-25 11:58 +0000 [r373532-373550] Joshua Colp <jcolp@digium.com>
* res/res_rtp_multicast.c: Fix an issue where a caller to ast_write
on a MulticastRTP channel would determine it failed when in
reality it did not. When sending RTP packets via multicast the
amount of data sent is stored in a variable and returned from the
write function. This is incorrect as any non-zero value returned
is considered a failure while a return value of 0 is success. For
callers (such as ast_streamfile) that checked the return value
they would have considered it a failure when in reality nothing
went wrong and it was actually a success. The write function for
the multicast RTP engine now returns -1 on failure and 0 on
success, as it should. (closes issue ASTERISK-17254) Reported by:
wybecom
* channels/chan_sip.c: Add missing checks that I neglected. The SIP
technology and SIP info technology should be considered equal.
(closes issue ASTERISK-20409) Reported by: michele cicciotti
privatewave
2012-09-24 22:15 +0000 [r373504] Matthew Jordan <mjordan@digium.com>
* res/res_rtp_asterisk.c: Revert change to res_rtp_asterisk
committed in r373236 (1.8) The change committed in r373236
attempted to account for endpoints that increased their RTP
timestamp in DTMF end of event re-transmissions. This change
attempted to make Asterisk continue to work with endpoints that
failed to follow the RFC while maintaining the fix that allowed
for out of order DTMF to be handled. Unfortunately, there is no
free lunch, and this patch broke any system that sent DTMF
immediately after an RTP session was established or when an SSRC
is updated. As such, that patch is being reverted for the
previous behavior. Endpoints that erroneously increase the RTP
timestamp in DTMF end of event packets will not work properly
with Asterisk. (issue ASTERISK-20424)
2012-09-24 22:09 +0000 [r373500] Richard Mudgett <rmudgett@digium.com>
* channels/chan_sip.c: Be consistent, send From: "Anonymous"
<sip:anonymous@anonymous.invalid> When setting
CALLERID(pres)=unavailable in the dialplan, the From header in
the SIP message contains "Anonymous"
<sip:Anonymous@anonymous.invalid>. For consistency, Asterisk
should use a lowercase a in the userpart of the URI. * Make the
From header use a lowercase A in the userpart of the anonymous
URI. (closes issue ASTERISK-19838) Reported by: Antti Yrjola
Patches: chan_sip_patch_ASTERISK-19838.patch (license #6383)
patch uploaded by Antti Yrjola
2012-09-24 20:57 +0000 [r373467] Jonathan Rose <jrose@digium.com>
* apps/app_mixmonitor.c, funcs/func_audiohookinherit.c:
func_audiohookinherit: Document some missed sources. This patch
also mentions that AUDIOHOOK_INHERIT can be used to transfer
MixMonitor audiohooks. There is also wiki that addresses
audiohooks and the use of AUDIOHOOK_INHERIT at the following
link: https://wiki.asterisk.org/wiki/display/AST/Audiohooks
(closes issue ASTERISK-18220) Reported by: Ishfaq Malik
2012-09-24 19:15 +0000 [r373438] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: Fix a deadlock caused by a race condition
between removing a hint and reloading the dialplan and
subscribing to the removed hint. If conditions were right it was
possible for both the PBX core and chan_sip to deadlock by both
having a lock that the other wants. In the case of the PBX core
it had the contexts lock and wanted a SIP dialog lock, while in
the case of chan_sip it had the SIP dialog lock and wanted the
contexts lock. This fix unlocks the SIP dialog before getting the
extension state so that the other thread will not block on trying
to lock it. Once the extension state is retrieved the SIP dialog
is locked again and life carries on. As the SIP dialog is
reference counted it is not possible for it to go away after
unlocking. (closes issue ASTERISK-20437) Reported by: jhutchins
2012-09-24 15:40 +0000 [r373424] Richard Mudgett <rmudgett@digium.com>
* channels/chan_sip.c: Fix potential reentrancy problems in
chan_sip. Asterisk v1.8 and later was not as vulnerable to this
issue. * Made find_call() lock each private as it processes the
found dialogs. (Primary cause of ABE-2876) * Made the other
functions that traverse the dialogs container lock each private
as it examines them. * Fix race condition in sip_call() if the
thread that sent the INVITE is held up long enough for a response
to be processed. The p->initid for the INVITE retransmission
could be added after it was canceled by the response processing.
* Made __sip_destroy() clean up resource pointers after freeing.
This is primarily defensive in case someone has a stale private
pointer. * Removed redundant memset() in reqprep(). The call to
init_req() already does the memset() and is the first reference
to req in reqprep(). * Removed useless set of req.method in
transmit_invite(). The calls to initreqprep() and reqprep() have
to do this because they memset() the req. JIRA ABE-2876
.......... Merged -r373423 from
https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
2012-09-21 19:00 +0000 [r373298-373342] Jonathan Rose <jrose@digium.com>
* channels/iax2-provision.c: iax2-provision: Fix improper return on
failed cache retrieval (closes issue ASTERISK-20337) reported by:
John Covert Patches: iax2-provision.c.patch uploaded by John
Covert (license 5512)
* apps/app_queue.c: app_queue: Make queue reload members and
variants of that work Prior to this patch, 'queue reload members'
cli command did not work at all. This also affects the manager
function 'QueueReload' when supplied with the 'members: yes'
field. (closes issue AST-956) Reported by: John Bigelow
2012-09-20 19:12 +0000 [r373242] Joshua Colp <jcolp@digium.com>
* apps/app_meetme.c: Fix incorrect MeetME conference bridge
reference count decrementing and sometimes premature destruction.
When using the 'e' or 'E' option to MeetMe the configured
conference bridges are loaded and examined to see if any are
empty. If no conference bridges are empty the caller is prompted
to enter the number of one. This operation left around a pointer
to the last created conference bridge still containing
participants. When the caller that was not able to find any empty
conference bridge hung up this pointer was disposed of and the
reference count of the conference bridge decremented. If there
was only a single participant in the conference bridge it was
ultimately destroyed prematurely. (closes issue AST-994) Reported
by: John Bigelow
2012-09-20 18:41 +0000 [r373236] Matthew Jordan <mjordan@digium.com>
* res/res_rtp_asterisk.c: When processing RFC 2833 DTMF, accomodate
increasing timestamps in End events While endpoints should not be
changing the source timestamp between DTMF event packets, the
fact is there exists those endpoints that do exactly that. To
work around this, we absorb timestamps within the expected
re-transmit period. Note that this period only affects End of
Event packets, so it should not prevent the detection of new DTMF
digits that happen to arrive right on top of each other. (closes
issue ASTERISK-20424) Reported by: Vladimir Mikhelson Tested by:
mjordan, Vladimir Mikhelson Review:
https://reviewboard.asterisk.org/r/2124
2012-09-19 16:02 +0000 [r373165] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: Fix a regression where direct media was not
permitted for calls using SIP INFO DTMF. A change was committed
to fix direct media ACL support. This change wrongly assumed that
only a single channel technology structure exists for chan_sip.
This is in fact false as a second exists for calls using SIP INFO
DTMF. The code which performs direct media ACL checking now
checks for both the non-INFO DTMF and INFO DTMF channel
technology structures. (closes issue ASTERISK-20409) Reported by:
michele cicciotti privatewave
2012-09-18 20:12 +0000 [r373131] Sean Bright <sean@malleable.com>
* main/manager.c: Don't crash when passing a NULL message to
__astman_get_header. Before this commit, __astman_get_header
would blindly dereference the passed in 'struct message *' to
traverse the header list. There are cases, however, such as
'*CLI> sip qualify peer foo' where the message pointer is NULL,
so we need to check for that.
2012-09-15 00:13 +0000 [r373090] Richard Mudgett <rmudgett@digium.com>
* channels/sig_ss7.c: Made companding law for SS7 calls only
determined by SS7 signaling type. For SS7, the companding law for
a call was chosen inconsistently depending upon ss7type (ITU vs
ANSI) and the DAHDI companding default (T1 vs E1). For incoming
calls, the companding law was determined by ss7type. For outgoing
calls, the companding law was determined by the DAHDI default.
With the wrong combination you would get A-law/u-law conflicts.
An A-law/u-law conflict sounds like bad static on the line. SS7
ITU signaling with E1 line: ok SS7 ITU signaling with T1 line:
noise SS7 ANSI signaling with E1 line: noise SS7 ANSI signaling
with T1 line: ok * Fix the companding law used to be determined
by the SS7 signaling type only.
2012-09-14 19:07 +0000 [r373061] Matthew Jordan <mjordan@digium.com>
* main/ssl.c, main/tcptls.c, channels/chan_sip.c: Resolve memory
leaks in TLS initialization and TLS client connections This patch
resolves two sources of memory leaks when using TLS in Asterisk:
1) It removes improper initialization (and multiple
re-initializations) of portions of the SSL library. Asterisk
calls SSL_library_init and SSL_load_error_strings during SSL
initialization; collectively this obviates the need for calling
any of the following during initialization or client connection
handling: * ERR_load_crypto_strings (handled by
SSL_load_error_strings) * OpenSSL_add_all_algorithms (synonym for
SSL_library_init) * SSLeay_add_ssl_algorithms (synonym for
SSL_library_init) 2) Failure to completely clean up all memory
allocated by Asterisk and by the SSL library for TLS clients.
This included not freeing the SSL_CTX object in the SIP channel
driver, as well as not clearing the error stack when the TLS
client exited. Note that these memory leaks were found by Thomas
Arimont, and this patch was essentially written by him with some
minor tweaks. (closes issue AST-889) Reported by: Thomas Arimont
Tested by: Thomas Arimont patches: (bugAST-889.patch) by Thomas
Arimont (license 5525) Review:
https://reviewboard.asterisk.org/r/2105
2012-09-13 18:39 +0000 [r373024] dlee <dlee@localhost>:
* include/asterisk/channel.h, main/channel.c: Fix timeouts for
ast_waitfordigit[_full]. ast_waitfordigit_full would simply pass
its timeout to ast_waitfor_nandfds, expecting it to decrement the
timeout by however many milliseconds were waited. This is a
problem if it consistently waits less than 1ms. The timeout will
never be decremented, and we wait... FOREVER! This patch makes
ast_waitfordigit_full manage the timeout itself. It maintains the
previously undocumented behavior that negative timeouts wait
forever. (closes issue ASTERISK-20375) Reported by: Mark
Michelson Tested by: Mark Michelson Review:
https://reviewboard.asterisk.org/r/2109/
2012-09-13 Asterisk Development Team <asteriskteam@digium.com>
* Asterisk 1.8.17.0-rc1 Released.
2012-09-12 15:42 +0000 [r372959] Matthew Jordan <mjordan@digium.com>
* main/astobj2.c, include/asterisk/astobj2.h: Constify
__ao2_ref_debug in astobj2 When REF_DEBUG is enabled in certain
files - most notably ccss.c - the 'tag' parameter passed to
__ao2_ref_debug will be a const char *. The function currently
expects that parameter to not be const. This causes a warning
when compiling, as the const qualifier is being discarded. With
dev-mode enabled, this prevents compiling Asterisk. This patch
makes __ao2_ref_debug's tag and file parameters const. (closes
issue ASTERISK-20408) Reported by: mjordan
2012-09-12 14:51 +0000 [r372932] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Add channel name to a warning to make
debugging easier. The "autodestruct with owner in place" message
is typically indicative of a channel reference leak. Printing out
the name of the channel in the message may be helpful when trying
to debug the issue.
2012-09-11 22:11 +0000 [r372902] Jonathan Rose <jrose@digium.com>
* channels/chan_local.c: chan_local: Switch from using a random 4
digit hex identifier to unique id Changes chan_local channels to
use an 8 digit hex identifier generated atomically and
sequentially in order to eliminate the chance of having multiple
channels with the same name during high call volume situations.
(issue ASTERISK-20318) Reported by: Dan Cropp Review:
https://reviewboard.asterisk.org/r/2104/
2012-09-11 15:26 +0000 [r372840] Mark Michelson <mmichelson@digium.com>
* main/features.c: Fix bad channel application data reference. When
channels get bridged due to an AMI bridge action or a DTMF
attended transfer, the two channels that get bridged have their
application data pointing to the other channel's name. This means
that if one channel is hung up but the other moves on, it means
that the channel that moves on will have its application data
pointing at freed memory. (issue ASTERISK-20335) Reported by:
aragon
2012-09-10 20:53 +0000 [r372804] Kinsey Moore <kmoore@digium.com>
* channels/chan_iax2.c: Ensure iax2 debug output is displayed when
expected When IAX2 debug was changed from iax_showframe to
iax_outputframe, some instances were missed (or added afterward).
This was causing debug output to not be displayed when expected.
(closes issue ASTERISK-20338) Reported-by: John Covert Patch-by:
John Covert
2012-09-10 18:35 +0000 [r372765] Jonathan Rose <jrose@digium.com>
* apps/app_meetme.c: app_meetme: Document that 'p' option will
continue in dialplan. (closes issue AST-991) Reported by John
Bigelow
2012-09-10 18:31 +0000 [r372763] Kinsey Moore <kmoore@digium.com>
* channels/chan_sip.c: Warn on CLI when UDPTL init fails This adds
a CLI warning when a SDP offer is rejected due to UDPTL
initialization failure. Previously, there was no indication of
the reason for offer rejection in this case. (closes issue
ASTERISK-20357) Reported-by: Francesco Usseglio Gaudi
2012-09-10 17:07 +0000 [r372736] Jonathan Rose <jrose@digium.com>
* main/channel.c: Masquerade: Retain parkinglot settings made by
CHANNEL function. Prior to this patch, the user would have a
parkinglot set on a channel that was parked and when the channel
was retrieved, any attempt by that channel to park would simply
use the default. This patch makes parkinglot values set in this
way be retained through the masquerade. (closes issue AST-990)
Reported by: Nick Huskinson Patches:
masquerade_parkinglot_patch.diff Uploaded by Jonathan Rose
(license 6182)
2012-09-09 01:19 +0000 [r372709] Matthew Jordan <mjordan@digium.com>
* channels/sip/sdp_crypto.c: Only re-create an SRTP session when
needed; respond with correct crypto policy In r356604, SRTP
handling was fixed to accomodate multiple crypto keys in an SDP
offer and the ability to re-create an SRTP session when the
crypto keys changed. In certain circumstances - most notably when
a phone is put on hold after having been bridged for a
significant amount of time - the act of re-creating the SRTP
session causes problems for certain models of phones. The patch
committed in r356604 always re-created the SRTP session
regardless of whether or not the cryptographic keys changed.
Since this is technically not necessary, this patch modifies the
behavior to only re-create the SRTP session if Asterisk detects
that the remote key has changed. This allows models of phones
that do not handle the SRTP session changing to continue to work,
while also providing the behavior needed for those phones that do
re-negotiate cryptographic keys. In addition, in Asterisk 1.8
only, it was found that phones that offer AES_CM_128_HMAC_SHA1_32
will end up with no audio if the phone is the initiator of the
call. The phone will send an INVITE request specifying that
AES_CM_128_HMAC_SHA1_32 be used for the cryptographic policy;
Asterisk will set its policy to that value. Unfortunately, when
the call is Answered and a 200 OK is sent back to the UA, the
policy sent in the response's SDP will be the hard coded value
AES_CM_128_HMAC_SHA1_80. This potentially results in Asterisk
using the INVITE request's policy of AES_CM_128_HMAC_SHA1_32,
while the phone uses Asterisk's response of
AES_CM_128_HMAC_SHA1_80. Hilarity ensues as both endpoints think
the other is crazy. This patch fixes that by caching the policy
from the request and responding with it. Note that this is not a
problem in Asterisk 10 and later, as the ability to configure the
policy was added in that version. (issue ASTERISK-20194) Reported
by: Nicolo Mazzon Tested by: Nicolo Mazzon Review:
https://reviewboard.asterisk.org/r/2099
2012-09-08 03:54 +0000 [r372682] dlee <dlee@localhost>:
* main/Makefile: Add OPENSSL_INCLUDE to the CFLAGS for ssl.c and
tcptls.c. Without this flag, those files will compile with the
system installed OpenSSL headers (if they exist). This is a real
bummer if a different path was specified using --with-ssl=
(closes issue ASTERISK-20392)
2012-09-07 23:05 +0000 [r372620-372655] Richard Mudgett <rmudgett@digium.com>
* main/astmm.c: Fix MALLOC_DEBUG version of ast_strndup(). (closes
issue ASTERISK-20349) Reported by: Brent Eagles
* funcs/func_math.c: Remove annoying unconditional debug message
from INC/DEC functions. (closes issue AST-1001) Reported by:
Guenther Kelleter
* apps/app_queue.c: Fix exception path typo in app_queue.c
try_calling(). (closes issue ASTERISK-20380) Reported by: Jeremy
Pepper Patches: fix-local-channel-locking.patch (license #6350)
patch uploaded by Jeremy Pepper
* apps/app_voicemail.c: Fix VoicemailUserEntry event headers
ServerEmail and MailCommand reported values. The AMI action
VoicemailUsersList VoicemailUserEntry event headers ServerEmail
and MailCommand did not report the global values if they were not
overridden. The VoicemailUserEntry event header ServerEmail was
not populated with the global value if the voicemail user did not
override it. The VoicemailUserEntry event header MailCommand was
never populated with a value. * Removed unused struct ast_vm_user
member mailcmd[]. (closes issue AST-973) Reported by: John
Bigelow Tested by: rmudgett
2012-09-07 02:24 +0000 [r372554-372581] Matthew Jordan <mjordan@digium.com>
* apps/app_minivm.c: Free ast_str objects when temp file fails to
be created in MiniVM The previous commit (r372554) was from a
patch that was written before r366880, which ensured that ast_str
objects allocated in the sendmail routine were free'd in off
nominal paths. This commit frees the string objects in the off
nominal path introduced in r372554. (issue ASTERISK-17133)
Reported by: Tzafrir Cohen
* apps/app_minivm.c: Fix file descriptor leak and pointer scope
issue in MiniVM when sending mail When MiniVM sends an e-mail and
it has the volgain option set, it will spawn sox in a separate
process to handle the manipulation of the sound file. In doing
so, it creates a temporary file. There are two problems here: 1)
The file descriptor returned from mkstemp is leaked 2) The
finalfilename character pointer points to a buffer that loses
scope once volgain processing is finished. Note that in r316265,
Russell fixed some gcc warnings by using the return value of the
mkstemp call. A warning was placed in minivm that the file
descriptor was going to be leaked. This patch reverts that
change, as it handles the leak and 'uses' the file descriptor
returned from mkstemp. (closes issue ASTERISK-17133) Reported by:
Tzafrir Cohen patches: minivm_18501_demo.diff uploaded by Tzafrir
Cohen (license #5035)
2012-09-06 21:38 +0000 [r372517] Kinsey Moore <kmoore@digium.com>
* apps/app_queue.c: Ensure listed queues are not offered for
completion When using tab-completion for the list of queues on
"queue reset stats" or "queue reload
{all|members|parameters|rules}", the tab-completion listing for
further queues erroneously listed queues that had already been
added to the list. The tab-completion listing now only displays
queues that are not already in the list. (closes issue AST-963)
Reported-by: John Bigelow
2012-09-06 18:54 +0000 [r372498] dsessions <dsessions@localhost>:
* configs/res_ldap.conf.sample, channels/chan_sip.c: LDAP Realtime
Peers Cannot Register Prior to 1.8, it was not necessary for an
explicit "type" to be set for an asterisk LDAP realtime peer. Now
the routine find_peer actually checks the type field during
registration and fails to find the peer if it is not set. The
attached patches make the realtime type equal whatever type is
being searched for if the type is 0 upon return from routine
build_peer. (closes issue ASTERISK-17222) Reported by: John
Covert Patch by: David Vossel Tested by: Darren Sessions Review:
https://reviewboard.asterisk.org/r/2095/
2012-09-06 15:52 +0000 [r372471] Jonathan Rose <jrose@digium.com>
* UPGRADE.txt: chan_sip: Note change in behavior to how
directmediapermit/deny ACL works r366547 introduced a change to
the directmedia ACL for chan_sip which modified the behavior
significantly. Prior to the patch, this option would bridge peers
with directmedia if a peer's IP address matched its own
directmedia ACL. After that patch, the peer would check the
bridged peer's ACL instead. This change has been present since
1.8.14.0. That patched failed to document the change in
Upgrade.txt, so this patch adds mention of that change to
UPGRADE.txt (UPGRADE-1.8.txt in newer branches) (issue AST-876)
2012-09-06 14:28 +0000 [r372444] Kinsey Moore <kmoore@digium.com>
* apps/app_queue.c: Ensure "rules" is tab-completable for "queue
show" Previously, tabbing at the end of "queue show" produced a
list of available queues about which information could be shown,
but did not include an alternative command, "rules", to access
information about queue rules. The "rules" item should now be
shown in the list of tab-completable items. (closes issue
AST-958) Reported-by: John Bigelow
2012-09-06 02:48 +0000 [r372390-372417] Matthew Jordan <mjordan@digium.com>
* pbx/pbx_dundi.c: Fix DUNDi message routing bug when neighboring
peer is unreachable Consider a scenario where DUNDi peer PBX1 has
two peers that are its neighbors, PBX2 and PBX3, and where PBX2
and PBX3 are also neighbors. If the connection is temporarily
broken between PBX1 and PBX3, PBX1 should not include PBX3 in the
list of peers it sends to PBX2 in a DPDISCOVER message, as it
cannot send messages to PBX3. If it does, PBX2 will assume that
PBX3 already received the message and fail to forward the message
on to PBX3 itself. This patch fixes this by only including peers
in a DPDISCOVER message that are reachable by the sending node.
This includes all peers with an empty address (00:00:00:00:00:00)
and that are have been reached by a qualify message. This patch
also prevents attempting to qualify a dynamic peer with an empty
address until that peer registers. (closes issue ASTERISK-19309)
Reported by: Peter Racz patches: dundi_routing.patch uploaded by
Peter Racz (license 6290) The patch uploaded by Peter was
modified slightly for this commit.
* apps/app_followme.c: Allow configured numbers for FollowMe to be
greater than 90 characters When parsing a 'number' defined in
followme.conf, FollowMe previously parsed the number in the
configuration file into a buffer with a length of 90 characters.
This can artificially limit some parallel dial scenarios. This
patch allows for numbers of any length to be defined in the
configuration file. Note that Clod Patry originally wrote a patch
to fix this problem and received a Ship It! on the JIRA issue.
The patch originally expanded the buffer to 256 characters.
Instead, the patch being committed duplicates the string in the
config file on the stack before parsing it for consumption by the
application. (closes issue ASTERISK-16879) Reported by: Clod
Patry Tested by: mjordan patches: followme_no_limit.diff uploaded
by Clod Patry (license #5138) Slightly modified for this commit.
2012-09-05 19:20 +0000 [r372354] Kinsey Moore <kmoore@digium.com>
* main/manager.c: Correct documentation for ModuleLoad AMI action
The documentation incorrectly listed 'rtp' as a reloadable
subsystem and left out many other reloadable subsystems. It is
now also documented that subsystems may only be reloaded, not
loaded or unloaded. (closes issue AST-977) Reported-by: John
Bigelow
2012-09-05 18:34 +0000 [r372339] Alec L Davis <sivad.a@paradise.net.nz>
* main/dsp.c: dsp.c: in ast_mf_detect_init incorrectly sets
goertzel samples to 160, should be MF_GSIZE Related
https://reviewboard.asterisk.org/r/2097/
2012-09-05 18:29 +0000 [r372337] Kinsey Moore <kmoore@digium.com>
* main/pbx.c: Ensure counts generated in
manager_show_dialplan_helper are correct When
manager_show_dialplan_helper was written, the counter increment
for the total number of contexts was placed with the extensions
increment instead of in the enclosing loop. This function should
now generate correct context counts. (closes issue AST-970)
Reported-by: John Bigelow
2012-09-05 13:13 +0000 [r372268] Matthew Jordan <mjordan@digium.com>
* apps/app_voicemail.c: Fix memory leaks in app_voicemail when
using IMAP storage or realtime config This patch fixes two memory
leaks: 1. When find_user is called with NULL as its first
parameter, the voicemail user returned is allocated on the heap.
The inboxcount2 function uses find_user in such a fashion when
counting new messages, and fails to free the resulting voicemail
user object. 2. When populate_defaults is called on a voicemail
user, it wipes whatever flags have been set on the object by
copying over the global flags object. If the VM_ALLOCED flag was
ste on the voicemail user prior to doing so, that flag is
removed. This leaks the voicemail user when free_user is later
called. (closes issue ASTERISK-19155) Reported by: Filip Jenicek
patches: asterisk.patch2 uploaded by Filip Jenicek (license 6277)
Patch slightly modified for this commit. Review:
https://reviewboard.asterisk.org/r/2096
2012-09-05 07:35 +0000 [r372212-372239] Alec L Davis <sivad.a@paradise.net.nz>
* main/dsp.c: dsp.c: Fix multiple issues when no-interdigit delay
is present, and fast DTMF 50ms/50ms Revert DTMF hit/miss detector
to original -r349249 method with some changes, remove
unnecessary; 1. reseting of hits=0, when no signal, only need to
set it once. 2. incrementing of hits, when the hit is the same as
the current hit. 3. setting of lasthit, when it's the same as
before. Change HITS_TO_BEGIN to 2, MISSES_TO_END to 3 & 3
spelling mistakes (closes issue ASTERISK-19610) alecdavis
(license 585) Reported by: Jean-Philippe Lord Tested by:
alecdavis Review: https://reviewboard.asterisk.org/r/2085/
* main/dsp.c: dsp.c: optimize goerztzel sample loops, in
dtmf_detect, mf_detect and tone_detect use a temporary short int
when repeatedly used to call goertzel_sample. alecdavis (license
585) Reported by: alecdavis Tested by: alecdavis Review:
https://reviewboard.asterisk.org/r/2093/
2012-09-05 03:45 +0000 [r372185] Michael L. Young <elgueromexicano@gmail.com>
* res/res_rtp_asterisk.c: Fix Incrementing Sequence Number For
Retransmitted DTMF End Packets In Asterisk 1.4+, a fix was put in
place to increment the sequence number for retransmitted DTMF end
packets. With the introduction of the RTP engine API in 1.8, the
sequence number was no longer being incremented. This patch fixes
this regression as well as cleans up a few lines that were not
doing anything. (closes issue ASTERISK-20295) Reported by: Nitesh
Bansal Tested by: Michael L. Young Patches:
01_rtp_event_seq_num.patch uploaded by Nitesh Bansal (license
6418) asterisk-20295-dtmf-fix-cleanup.diff uploaded by Michael L.
Young (license 5026) Review:
https://reviewboard.asterisk.org/r/2083/
2012-09-05 02:16 +0000 [r372158] Matthew Jordan <mjordan@digium.com>
* cel/cel_pgsql.c: Fix memory leak when CEL is successfully written
to PostgreSQL database PQClear is not called when the result
object of a call to PQExec has a status of PGRES_COMMAND_OK.
Interestingly enough, the off nominal case was handled properly,
so this memory leak only occurred when CEL records were
successfully written. This patch properly clears the result in
the nominal code path. (closes issue ASTERISK-19991) Reported by:
Etienne Lessard Tested by: Etienne Lessard patches:
mem_leak_cel_pgsql.patch uploaded by Etienne Lessard (license
#6394)
2012-08-30 20:51 +0000 [r372048-372089] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c: Prevent crash on shutdown due to refcount error
on queues container. When app_queue is unloaded, the queues
container has its refcount decremented, potentially to 0. Then
the taskprocessor responsible for handling device state changes
is unreferenced. If the taskprocessor happens to be just about to
run its task, then it will create and destroy an iterator on the
queues container. This can cause the refcount on the queues
container to increase to 1 and then back to 0. Going back to 0 a
second time results in double frees. This failure was seen
periodically in the testsuite when Asterisk would shut down.
* apps/app_queue.c: Help prevent ringing queue members from being
rung when ringinuse set to no. Queue member status would not
always get updated properly when the member was called, thus
resulting in the member getting multiple calls. With this change,
we update the member's status at the time of calling, and we also
check to make sure the member is still available to take the call
before placing an outbound call. (closes issue ASTERISK-16115)
reported by nik600 Patches: app_queue.c-svn-r370418.patch
uploaded by Italo Rossi (license #6409)
2012-08-30 16:21 +0000 [r371961-372015] Matthew Jordan <mjordan@digium.com>
* channels/chan_iax2.c: AST-2012-013: Resolve ACL rules being
ignored during calls by some IAX2 peers When an IAX2 call is made
using the credentials of a peer defined in a dynamic Asterisk
Realtime Architecture (ARA) backend, the ACL rules for that peer
are not applied to the call attempt. This allows for a remote
attacker who is aware of a peer's credentials to bypass the ACL
rules set for that peer. This patch ensures that the ACLs are
applied for all peers, regardless of their storage mechanism.
(closes issue ASTERISK-20186) Reported by: Alan Frisch Tested by:
mjordan, Alan Frisch
* main/manager.c, README-SERIOUSLY.bestpractices.txt: AST-2012-012:
Resolve AMI User Unauthorized Shell Access through ExternalIVR
The AMI Originate action can allow a remote user to specify
information that can be used to execute shell commands on the
system hosting Asterisk. This can result in an unwanted
escalation of permissions, as the Originate action, which
requires the "originate" class authorization, can be used to
perform actions that would typically require the "system" class
authorization. Previous attempts to prevent this permission
escalation (AST-2011-006, AST-2012-004) have sought to do so by
inspecting the names of applications and functions passed in with
the Originate action and, if those applications/functions matched
a predefined set of values, rejecting the command if the user
lacked the "system" class authorization. As noted by IBM X-Force
Research, the "ExternalIVR" application is not listed in the
predefined set of values. The solution for this particular
vulnerability is to include the "ExternalIVR" application in the
set of defined applications/functions that require "system" class
authorization. Unfortunately, the approach of inspecting fields
in the Originate action against known applications/functions has
a significant flaw. The predefined set of values can be bypassed
by creative use of the Originate action or by certain dialplan
configurations, which is beyond the ability of Asterisk to
analyze at run-time. Attempting to work around these scenarios
would result in severely restricting the applications or
functions and prevent their usage for legitimate means. As such,
any additional security vulnerabilities, where an
application/function that would normally require the "system"
class authorization can be executed by users with the "originate"
class authorization, will not be addressed. Instead, the
README-SERIOUSLY.bestpractices.txt file has been updated to
reflect that the AMI Originate action can result in commands
requiring the "system" class authorization to be executed. Proper
system configuration can limit the impact of such scenarios.
(closes issue ASTERISK-20132) Reported by: Zubair Ashraf of IBM
X-Force Research
* doc/CODING-GUIDELINES (added): Restore CODING-GUIDELINES to doc
folder In r294740, the CODING-GUIDELINES was removed from the doc
folder in favor of the content on the Asterisk wiki. Some folks
still look in the doc folder initially for coding guideline
suggestions; as such, this patch adds a CODING-GUIDELINES file
back into the doc folder. The content of the file merely points
to the correct page on the Asterisk wiki where the coding
guidelines currently live. (closes issue ASTERISK-20279) Reported
by: Andrew Latham Patches: CODING-GUIDELINES.diff uploaded by
Andrew Latham (license 5985)
2012-08-29 20:42 +0000 [r371919] Jonathan Rose <jrose@digium.com>
* apps/app_meetme.c: app_meetme: Adding test events for following
activity in MeetMe.
2012-08-29 19:38 +0000 [r371860-371888] Richard Mudgett <rmudgett@digium.com>
* main/channel.c: Initialize file descriptors for dummy channels to
-1. Dummy channels usually aren't read from, but functions like
SHELL and CURL use autoservice on the channel. (closes issue
ASTERISK-20283) Reported by: Gareth Palmer Patches:
svn-371580.patch (license #5169) patch uploaded by Gareth Palmer
(modified)
* apps/app_dial.c: Fix hangup cause passthrough regression. The
v1.8 -r369258 change to fix the F and F(x) action logic
introduced a regression in passing the hangup cause from the
called channel to the caller channel. (closes issue
ASTERISK-20287) Reported by: Konstantin Suvorov Patches:
app_dial_hangupcause.patch (license #6421) patch uploaded by
Konstantin Suvorov (modified) Tested by: rmudgett
2012-08-29 16:59 +0000 [r371824] Jonathan Rose <jrose@digium.com>
* channels/chan_sip.c: chan_sip: Send 408 on retransmit timeout
instead of 603 (closes issue ASTERISK-20124) Reported by: Walter
Doekes
2012-08-27 21:47 +0000 [r371747-371787] Mark Michelson <mmichelson@digium.com>
* configs/agents.conf.sample: Fix misleading documentation in
agents.conf.sample regarding ackcall usage. The documentation
made it sound as if the DTMF acknowledgment was needed at the
time the agent logs in, rather than when the agent is called.
This is likely a relic from the days when there were multiple
ways of logging in agents. (closes issue AST-962) reported by
Steve Pitts
* main/manager.c: Fix incorrect documentation of the MailboxStatus
manager command. The "Waiting" field was misdocumented as
reporting the number of messages waiting. In reality, it simply
indicated the presence or absence of waiting messages. (closes
issue AST-975) reported by John Bigelow
* configs/queues.conf.sample: Fix incorrectly documented option in
queues.conf sharedlastcall defaults to "no" not "yes" (closes
issue AST-979) reported by Steve Pitts
2012-08-27 16:40 +0000 [r371718] dlee <dlee@localhost>:
* main/lock.c: Fixes ast_rwlock_timed[rd|wr]lock for BSD and
variants. The original implementations simply wrap pthread
functions, which take absolute time as an argument. The spinlock
version for systems without those functions treated the argument
as a delta. This patch fixes the spinlock version to be
consistent with the pthread version. (closes issue
ASTERISK-20240) Reported by: Egor Gorlin Patches: lock.c.patch
uploaded by Egor Gorlin (license 6416)
2012-08-27 13:43 +0000 [r371690] Kinsey Moore <kmoore@digium.com>
* main/utils.c: Implement workaround for BETTER_BACKTRACES crash
When compiling with BETTER_BACKTRACES enabled, Asterisk will
sometimes crash when "core show locks" is run. This happens
regularly in the testsuite since several tests run "core show
locks" to help with debugging. This seems to be a fault with
libraries on certain operating systems (notably CentOS 6.2/6.3)
running on virtual machines and utilizing gcc 4.4.6. (closes
issue ASTERISK-20090)
2012-08-26 23:03 +0000 [r371662] Alec L Davis <sivad.a@paradise.net.nz>
* main/dsp.c: mf_detect: incorrectly used DTMF_GSIZE instead of
MF_GSIZE
2012-08-21 20:35 +0000 [r371590] Mark Michelson <mmichelson@digium.com>
* main/utils.c, apps/app_queue.c, pbx/pbx_config.c,
res/res_jabber.c, apps/app_stack.c, channels/chan_oss.c,
res/res_config_sqlite.c, cdr/cdr_tds.c, main/xmldoc.c,
apps/app_dial.c, channels/chan_dahdi.c, channels/chan_sip.c,
funcs/func_odbc.c, main/file.c: Fix misuses of asprintf
throughout the code. This fixes three main issues * Change
asprintf() uses to ast_asprintf() so that it pairs properly with
ast_free() and no longer causes MALLOC_DEBUG to freak out. * When
ast_asprintf() fails, set the pointer NULL if it will be
referenced later. * Fix some memory leaks that were spotted while
taking care of the first two points. (Closes issue
ASTERISK-20135) reported by Richard Mudgett Review:
https://reviewboard.asterisk.org/r/2071
2012-08-20 15:25 +0000 [r371544] Kinsey Moore <kmoore@digium.com>
* main/udptl.c: Ignore recovered zero-length secondary UDPTL
packets In some cases, recovering lost packets using the
secondary packet recovery mechanism with UDPTL/T.38 can result in
the recovery of zero-length packets. These must be ignored or the
frame generated from them can cause segfaults and allocation
failures. (closes issue ASTERISK-19762) (closes issue
ASTERISK-19373) Reported-by: Benjamin (bulkorok) Reported-by: Rob
Gagnon (rgagnon)
2012-08-17 18:51 +0000 [r371469] Matthew Jordan <mjordan@digium.com>
* main/xmldoc.c: Fix memory leak in XML documentation When
formatting documentation fields, the XML documentation parser
calls xmldoc_get_formatted. This function allocates a string
buffer at the beginning of its routine. Unfortunately, on certain
code paths, it also calls xmldoc_string_cleanup, which assumes
that it will create the string buffer. The previously allocated
string buffer is then leaked by the xmldoc_string_cleanup
routine. Now: we don't do that. (closes issue AST-932) Reported
by: Alexander Homig
2012-08-17 15:49 +0000 [r371393-371436] Kinsey Moore <kmoore@digium.com>
* main/loader.c: Add instrumentation to subsystem reloads When
Asterisk is built with TEST_FRAMEWORK defined, Asterisk will now
generate TestEvent AMI events on subsystem reloads such as cdr,
dnsmgr, extconfig, etc. (issue PQ-1126)
* main/loader.c: Add module reload instrumentation for
TEST_FRAMEWORK This adds AMI events for module reloads when
Asterisk is built with TEST_FRAMEWORK enabled and corrects
generation of the module load AMI event. (issue PQ-1126)
2012-08-16 22:30 +0000 [r371392] Terry Wilson <twilson@digium.com>
* main/config.c: Handle integer over/under-flow in ast_parse_args
The strtol family of functions will return *_MIN/*_MAX on
overflow. To detect when an overflow has happened, errno must be
set to 0 before calling the function, then checked afterward.
(closes issue ASTERISK-20120) Reported by: Matt Jordan Review:
https://reviewboard.asterisk.org/r/2073/
2012-08-16 18:57 +0000 [r371337-371357] Jonathan Rose <jrose@digium.com>
* channels/chan_sip.c: chan_sip: Use pvt outgoing_call variable to
set Remote-Party-ID Header Previously the pvt SIP_OUTGOING flag
was used instead, which will frequently flip during reinvites.
(closes issue AST-897) Reported by: Thomas Arimont
* channels/chan_sip.c: chan_sip: Trigger reinvite if the SDP answer
is included in the SIP ACK Under certain conditions, a SIP
transaction involving directmedia wouldn't trigger a re-invite
because the SDP answer was included in an ACK instead of in a
message that we would have triggered the invite with. This patch
just queues a source change control frame if the dialog is using
directmedia when we find sdp for an ACK. (closes issue AST-913)
Reported by: Thomas Arimont
2012-08-15 23:10 +0000 [r371306] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c: Fix bug where final queue member would not be
removed from memory. If a static queue had realtime members, then
there could be a potential for those realtime members not to be
properly deleted from memory. If the queue's members were loaded
from realtime and then all the members were deleted from the
backend, then the queue would still think these members existed.
The reason was that there was a short- circuit in code such that
if there were no members found in the backend, then the queue
would not be updated to reflect this. Note that this only
affected static queues with realtime members. Realtime queues
with realtime members were unaffected by this issue. (closes
issue ASTERISK-19793) reported by Marcus Haas
2012-08-15 20:14 +0000 [r371270] Kinsey Moore <kmoore@digium.com>
* channels/chan_sip.c: Avoid unconditional NULLing of mwipvt on
relatedpeer on SIP dialog destruction The other instance of this
bug was fixed by jcolp/file in r121496. If we are destroying a
dialog only set the MWI dialog pointer on the related peer to
NULL if it is the dialog currently being destroyed. (closes issue
ASTERISK-20119) Patch-by: Misha Vodsedalek
2012-08-13 20:00 +0000 [r371201] Kinsey Moore <kmoore@digium.com>
* main/loader.c, apps/app_meetme.c: Add test instrumentation This
adds test instrumentation for loading and unloading of modules
and for certain actions in MeetMe to be used in the testsuite or
any other consumer of AMI events. These will only be generated
when Asterisk is built with TEST_FRAMEWORK enabled. (issue
PQ-1131) (issue PQ-1133)
2012-08-13 19:49 +0000 [r371198] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Fix problem where incorrect pointer was
checked for nullity.
2012-08-10 21:21 +0000 [r371141] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c: Fix a couple of documentation problems in
app_queue.c * The RemoveQueueMember app made mention of options
that could be passed in, but no options are supported. I have
removed the listing of options from the documentation. * The
RQMSTATUS variable did not list "NOTDYNAMIC" as a possible value
that could be set. (closes issue AST-949) reported by Steve Pitts
(closes issue AST-954) reported by Steve Pitts
2012-08-10 16:40 +0000 [r371060-371089] Alexandr Anikin <may@telecom-service.ru>
* addons/chan_ooh323.c: remove ALREADYGONE flag on ooh323 call data
by ooh323_indicate (CONGESTION/BUSY) due to call hasn't gone
there really. This indication arrive from asterisk core not h.323
stack (closes issue ASTERISK-19308) Reported by: Dmitry Melekhov
Patches: ASTERISK-19308.patch
* addons/ooh323c/src/ooGkClient.c: Send re-register packets by GRQ
(gatekeeper request) interval (close issue ASTERISK-20094)
Patches: ASTERISK-20094-2.patch
2012-08-09 18:58 +0000 [r371012] Richard Mudgett <rmudgett@digium.com>
* channels/sig_pri.c, channels/sig_ss7.c, channels/chan_dahdi.c,
configure, include/asterisk/autoconfig.h.in, configure.ac: Use
better libss7 detection test and move libpri compile test.
2012-08-09 18:58 +0000 [r370988-371011] Alexandr Anikin <may@telecom-service.ru>
* addons/ooh323c/src/ooGkClient.c: Fix to resend GRQ/RRQ if RRJ
(registration reject) is received (close issue ASTERISK-20094)
Patches: ASTERISK-20094.patch
* addons/ooh323c/src/ooh323ep.c: change opening h323 logfile with
append mode instead of overwrite
2012-08-09 17:39 +0000 [r370985] Kinsey Moore <kmoore@digium.com>
* apps/app_meetme.c: Correct documentation for the MeetMe x flag
The documentation for the x flag for MeetMe incorrectly described
its function as closing down the conference when the last marked
user left. It actually causes the users with that flag to leave
the conference when the last marked user exits. The functionality
of this flag is not changing.
2012-08-08 22:40 +0000 [r370952] Michael L. Young <elgueromexicano@gmail.com>
* apps/app_chanspy.c: Fix Not Unreferencing A Spied Channel When a
channel hangs up while being spied upon and the option to exit
the ChanSpy application when the spied on channel hangs up is
set, ast_autochan_destroy is not being called and therefore a
reference to the spied upon channel is not removed. The symptom
being reported was that when using func_group in the dialplan and
calling "group show channels" at the cli, the spied upon channel
was still being shown while "core show channels" showed that the
channel was not up. This patch calls ast_autochan_destroy when a
spied upon channel hangs up and the option to exit the ChanSpy
application is set, removing the reference to the channel
allowing the count for the group that the spied channel was part
of to be decremented. (closes issue ASTERISK-17515) Reported by:
Arkadiusz Malka Tested by: Alexandr Gordeev, Michael L. Young
Patches: asterisk-17515-destroy-autochan.diff uploaded by Michael
L. Young (license 5026)
2012-08-08 20:28 +0000 [r370923] Kinsey Moore <kmoore@digium.com>
* main/channel.c: Do not define a cause that doesn't actually exist
AST_CAUSE_NOTDEFINED is a placeholder for usage when there is no
cause information. As such, it should not be defined and
translatable as a cause.
2012-08-08 19:58 +0000 [r370900] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c, channels/sig_analog.c,
channels/sig_analog.h: Fix the analog dial *0 flash-hook of
bridged peer feature. The flash-hook the bridged peer feature now
correctly determines if the bridged peer is another chan_dahdi
channel, that it is an analog channel, and that it has the
correct signaling for an FXO port. It now also flash-hooks the
correct channel.
2012-08-07 19:19 +0000 [r370856] Kinsey Moore <kmoore@digium.com>
* main/channel.c: Add missing AST_CAUSE_* -> text translations
2012-08-06 15:00 +0000 [r370797] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Improve debug message for temporary outbound
proxies. Thanks to Paul Belanger for pointing this out.
2012-08-03 21:43 +0000 [r370769-370771] Mark Michelson <mmichelson@digium.com>
* channels/sip/config_parser.c: Seriously? Another compilation
error fixed. Somebody beat me.
* channels/chan_sip.c: Remove unused variable.
* channels/sip/config_parser.c, channels/sip/include/sip.h,
channels/chan_sip.c: Fix error in the "IPorHost" section of a SIP
dialstring. This is based on the review request posted by Walter
Doekes (referenced lower in the commit message) The main fix here
is to treat the IPorHost portion of the dial string as a
temporary outbound proxy. This ensures requests get sent to the
proper location. Due to the age of the request, some parts were
no longer relevant. For instance, the request moved outbound
proxy parsing code into a single method. This is done in a
previous commit, so it was not necessary to do again. Also, the
review request fixed some errors with regards to request routing
for CANCEL and ACK requests. This has also been fixed in more
recent commits. (closes issue ASTERISK-19677) reported by Walter
Doekes Review https://reviewboard.asterisk.org/r/1859
2012-08-01 02:25 +0000 [r370697] Kinsey Moore <kmoore@digium.com>
* utils/extconf.c: Revert alloca changes for utils These changes
were a tad overzealous in the utils directory. Unfortunately,
these don't compile with a "make".
2012-07-31 20:54 +0000 [r370666] Matthew Jordan <mjordan@digium.com>
* channels/chan_sip.c: Schedule pokes of registered SIP peers
within a given timespan after SIP reload With a large number of
SIP peers registered, performing a SIP reload causes a flood of
SIP OPTIONS request packets. These are immediately sent out, and,
as responses come back, can cause peers to be flagged as 'lagged'
due to handling of the many response messages. This fix prevents
this "packet storm" and schedules the pokes for a random time.
That time varies between 1 ms and the peer's qualify time, or, if
the qualify time is unknown, the global qualifyfreq setting. The
committed patch has some very small modifications to the patch
schmidts wrote for the review. (closes issue ASTERISK-19154)
Reported by: Nicolo Mazzon patches: issue19154.patch license
#6034 uploaded by schmidts Review:
https://reviewboard.asterisk.org/r/1652
2012-07-31 19:31 +0000 [r370642] Kinsey Moore <kmoore@digium.com>
* main/utils.c, funcs/func_logic.c, channels/chan_gtalk.c,
cdr/cdr_pgsql.c, channels/chan_iax2.c, res/res_jabber.c,
main/config.c, main/channel.c, res/ael/pval.c,
apps/app_osplookup.c, main/manager.c, pbx/pbx_spool.c,
main/strcompat.c, apps/app_minivm.c, main/features.c,
res/res_agi.c, main/http.c, main/logger.c, pbx/pbx_ael.c,
main/app.c, channels/chan_alsa.c, pbx/pbx_realtime.c,
addons/chan_mobile.c, apps/app_while.c, include/asterisk/utils.h,
main/pbx.c, res/res_config_pgsql.c, channels/chan_sip.c,
apps/app_festival.c, pbx/pbx_lua.c, funcs/func_cut.c,
tests/test_linkedlists.c, apps/app_getcpeid.c,
funcs/func_global.c, channels/chan_jingle.c, main/tcptls.c,
funcs/func_channel.c, apps/app_directed_pickup.c,
main/callerid.c, main/file.c, apps/app_macro.c, main/astmm.c,
apps/app_sms.c, main/event.c, pbx/pbx_dundi.c,
include/asterisk/strings.h, utils/extconf.c,
apps/app_mixmonitor.c, main/asterisk.c, main/dsp.c,
addons/res_config_mysql.c, apps/app_voicemail.c,
addons/app_mysql.c, apps/app_meetme.c, apps/app_dictate.c,
main/say.c, main/threadstorage.c, funcs/func_strings.c: Clean up
and ensure proper usage of alloca() This replaces all calls to
alloca() with ast_alloca() which calls gcc's __builtin_alloca()
to avoid BSD semantics and removes all NULL checks on memory
allocated via ast_alloca() and ast_strdupa(). (closes issue
ASTERISK-20125) Review: https://reviewboard.asterisk.org/r/2032/
Patch-by: Walter Doekes (wdoekes)
2012-07-31 15:26 +0000 [r370618] Mark Michelson <mmichelson@digium.com>
* configs/sip.conf.sample, channels/sip/include/sip.h,
channels/chan_sip.c: Help mitigate potential reinvite glare
scenarios. When Asterisk servers are set up back-to-back, and
direct media is to be used betweeen endpoints, it is fairly
common for the two Asterisk servers to send direct media
reinvites to each other simultaneously. This results in 491s and
ACKs being exchanged between the servers. While the media
eventually gets set up properly, the problem is that there can be
a noticeable delay for the streams to stabilize. This patch adds
a new directmedia option called "outgoing". With this set, an
immediate direct media reinvite will only be sent if the call
direction is outgoing. For incoming dialogs, an immediate direct
media reinvite will not be sent, but further "reactionary" direct
media reinvites may be sent. For those who are having some deja
vu, that's because this patch was originally committed to trunk
since there is a new configuration option added. After seeing a
bug report about audio being slow to set up on SIP calls, it
became apparent that this patch would be the best solution for
resolving the issue. The patch is unintrusive and will have no
effect unless the option is explicitly enabled. (closes issue
AST-896) reported by Thomas Arimont (closes issue ASTERISK-19857)
reported by Matt Jordan
2012-09-13 Asterisk Development Team <asteriskteam@digium.com>
* Asterisk 1.8.16.0 Released.
2012-09-11 Asterisk Development Team <asteriskteam@digium.com>
* Asterisk 1.8.16.0-rc2 Released.
* AST-2012-013: Resolve ACL rules being ignored during calls by some
IAX2 peers
* AST-2012-012: Resolve AMI User Unauthorized Shell Access through
ExternalIVR
* r371860: Fix hangup cause passthrough regression.
The v1.8 -r369258 change to fix the F and F(x) action logic
introduced a regression in passing the hangup cause from the called
channel to the caller channel.
(closes issue ASTERISK-20287)
Reported by: Konstantin Suvorov
Patches:
app_dial_hangupcause.patch (license #6421) patch uploaded by
Konstantin Suvorov (modified)
Tested by: rmudgett
* r372709: Only re-create an SRTP session when needed; respond with
correct crypto policy
In r356604, SRTP handling was fixed to accomodate multiple crypto
keys in an SDP offer and the ability to re-create an SRTP session
when the crypto keys changed. In certain circumstances - most
notably when a phone is put on hold after having been bridged for a
significant amount of time - the act of re-creating the SRTP session
causes problems for certain models of phones. The patch committed in
r356604 always re-created the SRTP session regardless of whether or
not the cryptographic keys changed. Since this is technically
not necessary, this patch modifies the behavior to only re-create the
SRTP session if Asterisk detects that the remote key has changed.
This allows models of phones that do not handle the SRTP session
changing to continue to work, while also providing the behavior
needed for those phones that do re-negotiate cryptographic keys.
In addition, in Asterisk 1.8 only, it was found that phones that
offer AES_CM_128_HMAC_SHA1_32 will end up with no audio if the phone
is the initiator of the call. The phone will send an INVITE request
specifying that AES_CM_128_HMAC_SHA1_32 be used for the cryptographic
policy; Asterisk will set its policy to that value. Unfortunately,
when the call is Answered and a 200 OK is sent back to the UA, the
policy sent in the response's SDP will be the hard coded value
AES_CM_128_HMAC_SHA1_80. This potentially results in Asterisk using
the INVITE request's policy of AES_CM_128_HMAC_SHA1_32, while the
phone uses Asterisk's response of AES_CM_128_HMAC_SHA1_80. Hilarity
ensues as both endpoints think the other is crazy.
This patch fixes that by caching the policy from the request and
responding with it. Note that this is not a problem in Asterisk 10
and later, as the ability to configure the policy was added in that
version.
(issue ASTERISK-20194)
Reported by: Nicolo Mazzon
Tested by: Nicolo Mazzon
Review: https://reviewboard.asterisk.org/r/2099
* r372840: Fix bad channel application data reference.
When channels get bridged due to an AMI bridge action
or a DTMF attended transfer, the two channels that
get bridged have their application data pointing to
the other channel's name. This means that if one channel
is hung up but the other moves on, it means that the
channel that moves on will have its application data
pointing at freed memory.
(issue ASTERISK-20335)
2012-07-31 Asterisk Development Team <asteriskteam@digium.com>
* Asterisk 1.8.16.0-rc1 Released.
2012-07-30 16:47 +0000 [r370563] Richard Mudgett <rmudgett@digium.com>
* channels/chan_misdn.c: Release B channel allocation on error path
in chan_misdn.
2012-07-25 21:00 +0000 [r370494] Jonathan Rose <jrose@digium.com>
* res/res_agi.c: res_agi: Add message indicating need for \n
character in verbose message The while loop responsible for
reading AGI messages from a fastAGI service can end up looping
indefinitely when an AGI script fails to indicate the end of a
message with a \n character. This patch adds an indication that
we are expecting a \n character to end the message to make it
more clear to users that this is necessary if they are receiving
this warning over and over. (issue ASTERISK-20061) Reported by:
Eike Kuiper
2012-07-24 16:53 +0000 [r370429] Kevin P. Fleming <kpfleming@digium.com>
* main/frame.c: Rewrite a comment that didn't adequately explain
the code it was documenting.
2012-07-24 16:49 +0000 [r370428] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
* channels/chan_oss.c: chan_oss: fix "sample rate" error message
2012-07-23 21:09 +0000 [r370360-370383] Kevin P. Fleming <kpfleming@digium.com>
* funcs/func_shell.c: Improve documentation for the SHELL()
dialplan function.
* main/channel.c: Free any datastores attached to dummy channels.
Revision 370205 added the use of a datastore attached to a dummy
channel to resolve a memory leak, but
ast_dummy_channel_destructor() in this branch did not free
datastores, resulting in a continued (but slightly smaller)
memory leak. This patch backports the change to free said
datastores from the Asterisk trunk. (related to issue AST-916)
2012-07-19 22:07 +0000 [r370275] Richard Mudgett <rmudgett@digium.com>
* main/cel.c: Fix compiler warnings. gcc (GCC) 4.2.4 has problems
casting away constness.
2012-07-19 22:00 +0000 [r370252-370273] Matthew Jordan <mjordan@digium.com>
* main/cel.c: Fix compilation error when MALLOC_DEBUG is enabled To
fix a memory leak in CEL, a channel datastore was introduced
whose destruction function pointer was pointed to the ast_free
macro. Without MALLOC_DEBUG enabled this compiles as fine, as
ast_free is defined as free. With MALLOC_DEBUG enabled, however,
ast_free takes on a definition from a different place then
utils.h, and became undefined. This patch resolves this by using
a reference to ast_free_ptr. When MALLOC_DEBUG is enabled, this
calls ast_free; when MALLOC_DEBUG is not enabled, this is defined
to be ast_free, which is defined to be free. (issue AST-916)
Reported by: Thomas Arimont
* res/res_rtp_asterisk.c: Handle extremely out of order RFC 2833
DTMF The current implementation of RFC 2833 DTMF handling in
res_rtp_asterisk will, if a packet arrives out of order, drop the
packet. This is to prevent duplicate ton generation in the
Asterisk core. Since the RTP layer does not buffer data itself,
this is the only option the RTP layer currently has for handling
packets that arrive out of order. For the most part, this doesn't
matter. For a particular digit, so long as a BEGIN packet arrives
before the first END packet, the digit will be produced. If
subsequent BEGIN packets arrive interleaved with the ENDs, they
will be dropped; likewise, if the BEGIN or END packets themselves
are out of order, those packets are dropped but sufficient
information is conveyed to the Asterisk core to produce the
appropriate digit. For certain sequences of DTMF packets - most
notably when, for a particular digit, an END packet arrives
before any BEGIN packet for that digit - this is a real problem.
When an END arrives before any BEGINs, the END packet is dropped
- but at the same time, it causes subsequent BEGIN packets for
that digit to be ignored. When the next in order END packet
arrives, it too is dropped - Asterisk believes that there was no
initial BEGIN. The solution this patch provides is to trust the
END packet to convey the information needed for the Asterisk core
to produce the DTMF digit. If we receive an END packet, and it: *
Has a timestamp greater then the last timestamp received from an
END packet * Does not have the same sequence number as the last
received sequence number (and is thus not an END packet
retransmission) Then we send the END frame up to the Asterisk
core. It contains enough DTMF information for Asterisk to produce
the digit. On the other hand, if we receive a BEGIN or
continuation packet that occurs with a timestamp equal to or less
then the last END timestamp, then we've received something out of
order - but we already have received enough information to
produce the digit. These packets are dropped. Much thanks goes to
Olle Johansson (oej) for providing the idea for this solution.
Review: https://reviewboard.asterisk.org/r/2033/ (issue
ASTERISK-18404) Reporter: Stephane Chazelas Tested by: Matt
Jordan
2012-07-18 19:12 +0000 [r370183-370205] Kevin P. Fleming <kpfleming@digium.com>
* main/cel.c: Resolve severe memory leak in CEL logging modules. A
customer reported a significant memory leak using Asterisk 1.8.
They have tracked it down to
ast_cel_fabricate_channel_from_event() in main/cel.c, which is
called by both in-tree CEL logging modules (cel_custom.c and
cel_sqlite3_custom.c) for each and every CEL event that they log.
The cause was an incorrect assumption about how data attached to
an ast_channel would be handled when the channel is destroyed;
the data is now stored in a datastore attached to the channel,
which is destroyed along with the channel at the proper time.
(closes issue AST-916) Review:
https://reviewboard.asterisk.org/r/2053/
* apps/app_macro.c, channels/chan_iax2.c, apps/app_mixmonitor.c,
apps/app_stack.c, funcs/func_global.c, res/res_odbc.c,
main/channel.c, addons/app_mysql.c, main/pbx.c,
funcs/func_curl.c, main/ccss.c, funcs/func_odbc.c,
funcs/func_lock.c: Ensure that all ast_datastore_info structures
are 'const'. While addressing a bug, I came across a instance of
'struct ast_datastore_info' that was not declared 'const'. Since
the API already expects them to be 'const', this patch changes
the declarations of all existing instances that were not already
declared that way.
2012-07-16 19:50 +0000 [r370131] Walter Doekes <walter+asterisk@wjd.nu>
* channels/chan_sip.c: Code cleanup and bugfix in chan_sip
outboundproxy parsing. The bug was clearing the global
outboundproxy when a peer-specific outboundproxy was bad. The
cleanup reduces duplicate code. Review:
https://reviewboard.asterisk.org/r/2034/ Reviewed by: Mark
Michelson
2012-07-16 13:44 +0000 [r370081] Kinsey Moore <kmoore@digium.com>
* UPGRADE.txt, CHANGES: Add comments about the BUILD_NATIVE change
This is a significant change and mention of it should have gone
into UPGRADE.txt and CHANGES.
2012-07-12 20:15 +0000 [r370017] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c, channels/sig_analog.c: Add missing
ast_hangup() calls on some analog exception paths. Make starting
analog_ss_thread() or __analog_ss_thread() failure paths hangup
the channel.
2012-07-12 20:05 +0000 [r369993-370014] Kinsey Moore <kmoore@digium.com>
* channels/chan_sip.c: Include Expires header for SIP PUBLISH
requests RFC3903 requres SIP PUBLISH requests to have Expires
headers, so add them. Review:
https://reviewboard.asterisk.org/r/2003/ Patch-by: gareth
* channels/chan_sip.c: Prevent double uri_escaping in chan_sip when
pedantic is enabled If pedantic mode is enabled, outbound invites
will have double-escaped contacts. This avoids setting an
already-escaped string into a field where it is expected to be
unescaped. (closes issue ASTERISK-20023) Reported-by: Walter
Doekes
2012-07-12 14:23 +0000 [r369970] Michael L. Young <elgueromexicano@gmail.com>
* funcs/func_math.c: Correct Documentation For DEC Function The
documentation for DEC in func_math.c was incorrect. Looks like a
copy and paste error. (Closes issue ASTERISK-20095) Reported by:
Billy Chia Tested by: Michael L. Young Patches: func_math.patch
uploaded by Billy Chia (license 6381)
2012-07-11 17:08 +0000 [r369937] Tilghman Lesher <tilghman@meg.abyt.es>
* funcs/func_realtime.c: Allow the REALTIME() function to report
errors back to the caller. Also, do more error checking on the
arguments specified to the REALTIME() function and clarify the
documentation. While I was editing the file, a few coding
guidelines fixups, as well. Review:
https://reviewboard.asterisk.org/r/2031/
2012-07-30 Asterisk Development Team <asteriskteam@digium.com>
* Asterisk 1.8.15.0 Released.
2012-07-11 Asterisk Development Team <asteriskteam@digium.com>
* Asterisk 1.8.15.0-rc1 Released.
2012-07-10 13:33 +0000 [r369869] Kinsey Moore <kmoore@digium.com>
* apps/app_stack.c, main/pbx.c: Improve Goto and GotoIf related
documentation Correct documentation on labeliftrue and
labeliffalse parameters of GotoIf() and update several other
locations that use the same syntax. (closes issue ASTERISK-20007)
Patch-by: Leif Madsen Reported-by: WIMPy
2012-07-09 17:05 +0000 [r369818] Jason Parker <jparker@digium.com>
* configs/sip_notify.conf.sample: Add Digium phones context to
sip_notify sample config. This makes it so that they can be
reconfigured remotely. (closes issue ASTERISK-19910)
2012-07-09 14:38 +0000 [r369792] Jonathan Rose <jrose@digium.com>
* channels/chan_sip.c: chan_sip: Fix small behavioral change
accidentally introduced in r369750 When removing the warning for
AST_CONTROL_FLASH from sip_indicate, I also inadvertently changed
the return value, which would likely make the indication not be
sent in audio. This fixes that while still removing the warning
message.
2012-07-06 20:54 +0000 [r369750] Jonathan Rose <jrose@digium.com>
* channels/chan_sip.c: chan_sip: Add case for FLASH control frames
so that we don't display a warning. chan_sip channels can receive
flash control frames when connected to analog phones and possibly
for other reasons. There really isn't a reason to warn when these
frames are received, we can safely ignore them. Patches:
dahdi_sip_flash.diff uploaded by Jonathan Rose (license 6182)
2012-07-06 18:40 +0000 [r369708-369731] Mark Michelson <mmichelson@digium.com>
* main/tcptls.c: Remove a superfluous and dangerous freeing of an
SSL_CTX. The problem here is that multiple server sessions share
a SSL_CTX. When one session ended, the SSL_CTX would be freed and
set NULL, leaving the other sessions unable to function. The code
being removed is superfluous because the SSL_CTX structures for
servers will be properly freed when ast_ssl_teardown is called.
(closes issue ASTERISK-20074) Reported by Trevor Helmsley
Patches: ASTERISK-20074.diff uploaded by Mark Michelson (license
#5049) Testers: Trevor Helmsley
* main/bridging.c: Fix bridging thread leak. The bridge thread was
exiting but was never being reaped using pthread_join(). This has
been fixed now by calling pthread_join() in ast_bridge_destroy().
(closes issue ASTERISK-19834) Reported by Marcus Hunger Review:
https://reviewboard.asterisk.org/r/2012
2012-07-05 19:01 +0000 [r369652] Kinsey Moore <kmoore@digium.com>
* apps/app_voicemail.c: AST-2012-011: Resolve heap corruption issue
with voicemail The heard and deleted arrays in the voicemail
state structure were not handled properly following the memory
leak fix in r354890 and a fix for an invalid free in r356797.
This could result in accessing and writing into freed memory. The
allocation for these arrays has been reworked to avoid the
possibility of invalid frees, access of freed memory, and crashes
that were occurring as a result of this. Locking around accesses
and modifications of the voicemail state structure members
dh_arraysize, heard, and deleted has been added to prevent
simultaneous modification and access when IMAP storage is in use.
If IMAP storage is not in use, this locking is not compiled in.
Review: https://reviewboard.asterisk.org/r/1994/ (closes issue
ASTERISK-19923) Reported by: Dan Delaney Tested by: Dan Delaney,
Julian Yap Patches: vm_alloc_fix.diff uploaded by kmoore (license
6273)
2012-07-05 17:01 +0000 [r369626] Matthew Jordan <mjordan@digium.com>
* channels/chan_sip.c: Do not send a BYE when a provisional
response arrives during a re-INVITE Commits r369557 and r369579
were done to improve handling of re-INVITEs when the UA that was
supposed to receive the re-INVITE fails to respond. A limitation
of those patches occurred when a UA sent a provisional response
to the re-INVITE. This triggered a sending of a BYE in
check_pending. This patch tweaks the handling of the re-INVITE
such that a BYE is not sent in response to those messages. (issue
ASTERISK-19992) Reported by: Steve Davies Tested by: Steve Davies
patches: (reinvite_tweak.diff license #5012 by Steve Davies)
2012-07-03 16:58 +0000 [r369557-369579] Terry Wilson <twilson@digium.com>
* channels/chan_sip.c: More improvements to re-INVITEs timing out
after a provisional response There is no need to call
check_pendings() on a final response to an INVITE when destroying
the scheduler entry as it will be done later during normal
processing. (issue ASTERISK-19992)
* channels/sip/include/sip.h, channels/chan_sip.c: Better handle
re-INVITEs with provisional but no final repsonses A previous
attempt at fixing this issue had negative side effects related to
attended transfers which this patch should resolve. Many thanks
to Steve Davies for all of the good suggestions and testing.
(closes issue ASTERISK-19992) Reported by: Steve Davies Tested
by: Steve Davies, Terry Wilson Review:
https://reviewboard.asterisk.org/r/2009/
2012-06-29 16:52 +0000 [r369471-369490] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: With some configurations a transport is not
actually specified so assume UDP in these cases.
* channels/chan_sip.c: Make the address family filter specific to
the transport. (closes issue ASTERISK-16618) Reported by: Leif
Madsen Review: https://reviewboard.asterisk.org/r/1667/
2012-06-27 20:58 +0000 [r369436] Terry Wilson <twilson@digium.com>
* channels/sip/include/sip.h, channels/chan_sip.c: AST-2012-010:
Clean up after a reinvite that never gets a final response The
basic problem is that if a re-INVITE is sent by Asterisk and it
receives a provisional response, but no final response, then the
dialog is never torn down. In addition to leaking memory, this
also leaks file descriptors and will eventually lead to Asterisk
no longer being able to process calls. This patch just keeps
track of whether there is an outstanding re-INVITE, and if there
is goes ahead and cleans up everything as though there was no
outstanding reinvite. Review:
https://reviewboard.asterisk.org/r/2009/ (closes issue
ASTERISK-19992) Reported by: Steve Davies Tested by: Steve
Davies, Terry Wilson
2012-06-26 13:21 +0000 [r369366-369390] Matthew Jordan <mjordan@digium.com>
* main/adsi.c: Fix crash in unloading of res_adsi module When
res_adsi is unloaded, it removes the ADSI functions that it
previously installed by passing a NULL adsi_funcs pointer to
ast_adsi_install_funcs. This function was not checking whether or
not the adsi_funcs pointer passed in was NULL before
dereferencing it to check whether or not the version of the
functions matches what the core was expecting it. This patch
makes it so that the version is only checked if a potentially
valid adsi_funcs pointer was passed in. Passing in NULL removes
the installed functions, bypassing the version check.
* main/cdr.c: Tweak CDR change in r369351 As Tilghman pointed out
on review 1996, the check to see if a CDR end time has been set
is sufficient to know whether or not the duration value can be
used. The check-in done for r369351 forgot to include this
change.
2012-06-25 19:13 +0000 [r369352] Mark Michelson <mmichelson@digium.com>
* channels/sip/include/sip.h, channels/chan_sip.c: Re-fix how local
tag is generated when sending a 481 to an INVITE. Match our local
tag to whatever to-tag was sent in the initial INVITE. Because
the size of the to-tag may not fit in the buffer in the sip_pvt,
it has been changed to a string field. (closes issue
ASTERISK-19892) reported by Walter Doekes Review:
https://reviewboard.asterisk.org/r/1977
2012-06-25 19:12 +0000 [r369351] Matthew Jordan <mjordan@digium.com>
* main/cdr.c: Fix incorrect duration reporting in CDRs created in
batch mode Certain places in core/cdr.c would, if the duration
value were 0, calculate the duration as being the delta between
the current time and the time at which the CDR record was
started. While this does not typically cause a problem in
non-batch mode, this can cause an issue in batch mode where CDR
records are gathered and written long after those calls have
ended. In particular, this affects calls that were never
answered, as those are expected to have a duration of 0. Often,
this would result in CDR logs with a significant number of calls
with lengthy durations, but dispositions of "BUSY". Note that
this does not affect cdr_csv, as that backend does not use
ast_cdr_getvar and instead directly reports the duration value.
The affected core backends include cdr_apative_odbc and
cdr_custom; other extended or deprecated CDR backends may
potentially still directly manipulate the duration values. (issue
ASTERISK-19860) Reported by: Thomas Arimont (issue AST-883)
Reported by: Thomas Arimont Tested by: Matt Jordan Review:
https://reviewboard.asterisk.org/r/1996/
2012-06-25 15:57 +0000 [r369327] Richard Mudgett <rmudgett@digium.com>
* main/features.c: Fix Bridge application occasionally returning to
the wrong location. * Fix do_bridge_masquerade() getting the
resume location from the zombie channel. The code must not touch
a clone channel after it has masqueraded it. The clone channel
has become a zombie and is starting to hangup. (closes issue
ASTERISK-19985) Reported by: jamicque Patches:
jira_asterisk_19985_v1.8.patch (license #5621) patch uploaded by
rmudgett Tested by: jamicque
2012-06-25 15:50 +0000 [r369302-369324] Mark Michelson <mmichelson@digium.com>
* main/adsi.c (added): Forgot to svn add this file in my last
commit.
* res/res_adsi.exports.in (removed), include/asterisk/adsi.h,
main/Makefile, res/res_adsi.c: Eliminate embedding of res_adsi.so
module. The way this is done is to stop using the optional API.
Instead, res_adsi.so, when loaded fills in a table of function
pointers. Review: https://reviewboard.asterisk.org/r/1991
* channels/chan_sip.c: Be more consistent with the return code for
requests received from invalid domain. When Asterisk receives an
INVITE from an external domain when allowexternaldomains=no send
a 403 instead of a 404. This is consistent with Asterisk's
behavior when receiving a REGISTER in this situation. (Closes
issue ASTERISK-19601) Reported by Matthew Jordan Patches:
ASTERISK-19601-no401.patch uploaded by Mark Michelson (License
#5049)
2012-06-23 00:04 +0000 [r369235-369282] Richard Mudgett <rmudgett@digium.com>
* main/features.c: Fix Bridge application and AMI Bridge action
error handling. * Fix AMI Bridge action disconnecting the AMI
link on error. * Fix AMI Bridge action and Bridge application not
checking if their masquerades were successful. * Fix Bridge
application running the h-exten when it should not. * Made
do_bridge_masquerade() return if the masquerade was successful so
the Bridge application and AMI Bridge action could deal with it
correctly. * Made bridge_call_thread_launch() hangup the passed
in channels if the bridge_call_thread fails to start. Those
channels would have been orphaned. * Made builtin_atxfer() check
the success of the transfer masquerade setup.
* apps/app_queue.c: Explicitly check caller hangup in app Queue
rather than a polluted res2 value.
* apps/app_dial.c: Check if PBX was started and fix F and F(x)
action logic in Dial application.
* main/ccss.c: Check if PBX was started for generic CCSS recall.
* channels/chan_sip.c: Change incorrect chan_sip zombie hangup
debug message. They are all zombies now.
2012-06-22 19:28 +0000 [r369214] Terry Wilson <twilson@digium.com>
* channels/chan_sip.c: Don't crash on a guest directmedia call A
sip_pvt may not have relatedpeer set if a call doesn't match up
with a peer. If there is no relatedpeer, there is no direct media
ACL to apply, so just return that it is allowed. (closes issue
ASTERISK-20040) Reported by: Terry Wilson
2012-06-22 17:14 +0000 [r369195] Kinsey Moore <kmoore@digium.com>
* channels/chan_sip.c: Don't parse media stream state for SIP video
streams The sendonly/recvonly/sendrecv/inactive media stream
attributes were parsed for video, but nothing was ever done with
them. With this code removed, an UNSUPPORTED message is produced
when these attributes are used in conjunction with a video stream
which is the better behavior since they were never really
supported in the first place.
2012-06-20 17:33 +0000 [r369130-369146] Alexandr Anikin <may@telecom-service.ru>
* addons/ooh323c/src/ooq931.c, addons/ooh323c/src/ooCalls.c: fix
locking issue on empty callList (issue ASTERISK-19298) Reported
by: Dmitry Melekhov Patches: ASTERISK-18322-2.patch
* addons/chan_ooh323.c: fix compile error (1.8 don't have
ast_channel_name macro)
2012-06-20 02:03 +0000 [r369108] Michael L. Young <elgueromexicano@gmail.com>
* include/asterisk/netsock2.h, main/netsock2.c: Fix NULL pointer
segfault in ast_sockaddr_parse() While working with
ast_parse_arg() to perform a validity check, a segfault occurred.
The segfault occurred due to passing a NULL pointer to
ast_sockaddr_parse() from ast_parse_arg(). According to the
documentation in config.h, "result pointer to the result. NULL is
valid here, and can be used to perform only the validity checks."
This patch fixes the segfault by checking for a NULL pointer.
This patch also adds documentation to netsock2.h about why it is
necessary to check for a NULL pointer. (Closes issue
ASTERISK-20006) Reported by: Michael L. Young Tested by: Michael
L. Young Patches: asterisk-20006-netsock-null-ptr.diff uploaded
by Michael L. Young (license 5026) Review:
https://reviewboard.asterisk.org/r/1990/
2012-06-19 23:28 +0000 [r369090] Alexandr Anikin <may@telecom-service.ru>
* addons/chan_ooh323.c: check rtptimeouts in ooh323 channels as per
config file (rtp voice, video, udptl except rtcp) (closes issue
ASTERISK-19179) Reported by: TSAREGORODTSEV Yury Patches:
19179-ooh323-2.patch
2012-06-19 15:30 +0000 [r369066] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Fix request routing issue when outboundproxy
is used. Asterisk was incorrectly setting the destination of
CANCELs and ACKs for error responses to the URI of the initial
INVITE. This resulted in further requests, such as INVITEs with
authentication credentials, to be routed incorrectly. Instead,
when these CANCEL or ACKs are to be sent, we should simply keep
the destination the same as what it previously was. There is no
need to alter it any. (closes issue ASTERISK-20008) Reported by
Marcus Hunger Patches: ASTERISK-20008.patch uploaded by Mark
Michelson (license #5049)
2012-06-18 18:07 +0000 [r369043] Richard Mudgett <rmudgett@digium.com>
* main/features.c: Fix monitoring calls put in a parking lot. * Fix
a regression that was introduced by -r366167 which effectively
disabled monitoring parked calls. (closes issue ASTERISK-20012)
Reported by: sdolloff Tested by: rmudgett
2012-06-15 15:57 +0000 [r369001-369002] Kevin P. Fleming <kpfleming@digium.com>
* build_tools/find_missing_support_level (added): Add a script to
enable finding source files without support-levels defined.
* main/devicestate.c, main/astfd.c, main/ssl.c,
main/taskprocessor.c, main/astobj2.c, main/indications.c,
main/config.c, main/loader.c, main/term.c, main/cli.c,
channels/sig_analog.c, main/framehook.c, main/strcompat.c,
main/plc.c, res/ais/evt.c, main/fskmodem_int.c, main/syslog.c,
main/stdtime/localtime.c, main/db.c, main/bridging.c,
channels/sig_ss7.c, main/datastore.c, main/sched.c,
channels/sip/sdp_crypto.c, main/pbx.c, main/strings.c,
channels/vcodecs.c, channels/iax2-provision.c, main/aoc.c,
pbx/dundi-parser.c, main/cel.c, channels/iax2-parser.c,
main/chanvars.c, main/netsock.c, main/data.c, main/srv.c,
channels/chan_misdn.c, main/privacy.c,
channels/sip/dialplan_functions.c, main/fixedjitterbuf.c,
main/test.c, main/audiohook.c, main/alaw.c, main/asterisk.c,
main/timing.c, main/global_datastores.c, main/fskmodem_float.c,
main/ccss.c, channels/sip/reqresp_parser.c,
channels/misdn/isdn_msg_parser.c, main/utils.c, main/xml.c,
main/autochan.c, main/enum.c, channels/misdn/isdn_lib.c,
main/fskmodem.c, channels/misdn_config.c, main/io.c,
res/ael/pval.c, main/channel.c, main/cdr.c, main/ulaw.c,
main/dial.c, main/tdd.c, main/heap.c, channels/console_gui.c,
channels/misdn/ie.c, main/logger.c, channels/console_board.c,
main/app.c, main/image.c, main/dns.c, main/lock.c, main/stun.c,
main/dnsmgr.c, channels/sip/srtp.c, main/translate.c,
main/slinfactory.c, main/jitterbuf.c, main/acl.c,
channels/sig_pri.c, main/tcptls.c, main/hashtab.c,
main/abstract_jb.c, main/callerid.c, main/file.c,
res/snmp/agent.c, main/astmm.c, channels/misdn/portinfo.c,
main/event.c, channels/sip/config_parser.c, channels/vgrabbers.c,
main/xmldoc.c, main/dsp.c, main/udptl.c, main/netsock2.c,
main/autoservice.c, main/rtp_engine.c, main/frame.c,
main/security_events.c, res/ais/clm.c, main/threadstorage.c,
main/say.c, channels/console_video.c: Add support-level
indications to many more source files. Since we now have tools
that scan through the source tree looking for files with specific
support levels, we need to ensure that every file that is a
component of a 'core' or 'extended' module (or the main Asterisk
binary) is explicitly marked with its support level. This patch
adds support-level indications to many more source files in tree,
but avoids adding them to third-party libraries that are included
in the tree and to source files that don't end up involved in
Asterisk itself.
2012-06-14 15:23 +0000 [r368898-368927] Mark Michelson <mmichelson@digium.com>
* main/Makefile: Revert Makefile change to remove embedding
res_adsi.so The change has resulted in a linking error for
certain versions of GCC. This is much worse than the original
issue, so for now, temporarily revert the change. A more thorough
change will be sought out.
* funcs/func_volume.c: Fix a deadlock that occurs when func_volume
is used on a local channel. This was discovered by trying to
perform a call forward to an extension that makes use of
func_volume. When the local channel is optimized away, the
datastore on the local;2 channel would have its audiohook
destroyed rather than detaching the audiohook from the channel
and then destroying it. With this patch, func_volume's datastore
destructor takes the proper route of detaching the audiohook and
then destroying it. (closes issue ASTERISK-19611) reported by
Volker Sauer Patches: ASTERISK-19611.patch uploaded by Mark
Michelson (license #5049)
2012-06-13 20:26 +0000 [r368894] Matthew Jordan <mjordan@digium.com>
* res/res_smdi.c, res/res_adsi.c: Mark res_smdi/res_adsi as 'core'
supported modules Recently, various issues surrounding weak
symbols have caused problems with modules that rely on that
feature to be enabled in menuselect. This includes app_voicemail
and chan_dahdi, as they both rely upon res_smdi and res_adsi,
which, in certain circumstances, may not be enabled by default in
menuselect. Because res_smdi/res_adsi are dependencies for
chan_dahdi/app_voicemail, this patch marks both as 'core'
supported modules. This will allow both app_voicemail and
chan_dahdi to be enabled as well, regardless of whether or not
that system supports weak symbols. (issue AST-900) Reported by:
Thomas Arimont (issue AST-885) Reported by: Denis Alberto
Martinez
2012-06-13 19:00 +0000 [r368873] Mark Michelson <mmichelson@digium.com>
* main/Makefile: Remove forced linking of res_adsi.o In GCC 4.5+
the result is that Asterisk has a phantom module loaded at
startup, claiming to be res_adsi. (closes issue ASTERISK-19920)
reported by Leif Madsen
2012-06-13 14:27 +0000 [r368830-368852] Matthew Jordan <mjordan@digium.com>
* Makefile: Do not install empty directories; add ASTLIBDIR r368830
modified the installation script to only create a directory if
that directory does not exist. If some directory variable was
empty, it would attempt to create the empty location. It also
failed to create the ASTLIBDIR directory. This patch fixes it
such that the correct directories are made and only created if a
value specifying them actually exists.
* Makefile: Do not perform install on existing directories If a
directory already exists, performing a 'make install' will remove
the permissions associated with the current directory and replace
them with the permissions of the user executing the install. This
patch changes this behavior to only perform an install on the
directory if the directory does not exist. Thus, if a user later
changes the permissions on that directory, those permissions will
be preserved in subsequent installs. Review:
https://reviewboard.asterisk.org/r/1986 Review:
https://reviewboard.asterisk.org/r/1864 (closes issue
ASTERISK-19492) Reported by: Karl Fife Tested by: Paul Belanger,
Tilghman Lesher patches: ASTERISK-19492 by pabelanger (uploaded
by mjordan)
2012-06-12 15:36 +0000 [r368807] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Set the Caller ID "tag" on peers even if
remote party information is present. On incoming calls, we were
setting the cid_tag on the dialog only if there was no remote
party information (Remote-Party-ID or P-Asserted-Identity)
present. The Caller ID tag is an invented parameter, though, and
should be set no matter the circumstance. (closes issue
ASTERISK-19859) Reported by Thomas Arimont (closes issue AST-884)
Reported by Trey Blancher
2012-06-11 17:03 +0000 [r368759] Richard Mudgett <rmudgett@digium.com>
* include/asterisk/channel.h, channels/chan_iax2.c, main/channel.c,
channels/chan_dahdi.c, channels/sig_analog.c,
channels/chan_sip.c: Fix deadlock potential with
ast_set_hangupsource() calls. Calling ast_set_hangupsource() with
the channel lock held can result in a deadlock because the
function also locks the bridged channel. (issue ASTERISK-19537)
(closes issue AST-891) Reported by: Guenther Kelleter Tested by:
Guenther Kelleter (closes issue ASTERISK-19801) Reported by: Alec
Davis
2012-06-11 15:13 +0000 [r368719-368738] Kinsey Moore <kmoore@digium.com>
* apps/app_queue.c, main/loader.c, channels/chan_dahdi.c,
res/res_config_odbc.c, channels/sip/dialplan_functions.c,
pbx/pbx_config.c, apps/app_directory.c, res/res_odbc.c,
res/res_speech.c, apps/app_voicemail.c, main/udptl.c,
channels/sip/sdp_crypto.c, channels/chan_sip.c, res/res_fax.c,
main/say.c, funcs/func_strings.c, channels/sip/reqresp_parser.c:
Fix coverity UNUSED_VALUE findings in core support level files
Most of these were just saving returned values without using them
and in some cases the variable being saved to could be removed as
well. (issue ASTERISK-19672)
* main/md5.c: Fix compilation in dev-mode Backport a compilation
fix in md5.c from trunk that only showed up in dev-mode under
certain compiler versions.
2012-07-10 Asterisk Development Team <asteriskteam@digium.com>
* Asterisk 1.8.14.0 Released.
2012-07-06 Asterisk Development Team <asteriskteam@digium.com>
* Asterisk 1.8.14.0-rc2 Released.
* AST-2012-010: Possible Resource Leak on Uncompleted Re-INVITE
transactions
* AST-2012-011: Remote Crash Vulnerability in VoiceMail Application
* Fix crash on a guest directmedia call
A sip_pvt may not have relatedpeer set if a call doesn't match up
with a peer. If there is no relatedpeer, there is no direct media
ACL to apply, so just return that is is allowed.
(closes issue ASTERISK-20040)
* Fix request routing issue when outboundproxy is used
Asterisk was incorrectly setting the destination of CANCELs and ACKs
for error responses to the URI of the initial INVITE. This resulted
in further requests, such as INVITEs with authentication
credentials, to be routed incorrectly. Instead when these CANCEL or
ACKs are to be esnt, we should simply keep the destination the same
as what it previously was. There is no need to alter it any.
(closes issue ASTERISK-20008)
* Fix monitoring calls put in a parking lot
Fix a regression that was introduced by r366167 which effectively
disabled monitoring parked calls.
(closes issue ASTERISK-20012)
2012-06-08 Asterisk Development Team <asteriskteam@digium.com>
* Asterisk 1.8.14.0-rc1 Released.
2012-06-06 21:27 +0000 [r368644] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c, channels/sig_analog.c: Fix POTS flash hook
to orignate a second call deadlock. A deadlock can occur when a
POTS phone tries to flash hook to originate a second call for
3-way or transfer. If another process is scanning the channels
container when the POTS line flash hooks then a deadlock will
occur. * Release the channel and private locks when creating a
new channel as a result of a flash hook. (closes issue
ASTERISK-19842) Reported by: rmudgett Tested by: rmudgett
2012-06-06 19:13 +0000 [r368625] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Fix a specific scenario where ACKs are not
matched. If a dialog-starting INVITE contains a to-tag, then
Asterisk will respond with a 481. In this case, the resulting
incoming ACK would not be matched, so Asterisk would continue
retransmitting the 481 until the transaction times out. There
were two issues. Asterisk, upon creating a sip_pvt would generate
a local tag. However, when the time came to transmit the 481,
since there was a to-tag in the INVITE, Asterisk would place this
original to-tag in the 481 response. When the ACK came in,
Asterisk would attempt to match the to-tag in the ACK to the
generated local tag. Unfortunately, Asterisk never actually
transmitted a response with the generated local tag, so the
to-tag in the ACK would not match. The other problem was that
when the 481 was sent, nothing was set on the sip_pvt to indicate
what CSeq is expected in the ACK. To fix the first problem, we
zero out the to-tag seen in the incoming INVITE. This way,
Asterisk, when time to send a response, will send its generated
local tag instead. To fix the second problem, we set the
sip_pvt's pendinginvite to the CSeq of the INVITE when we send a
481. (closes issue ASTERISK-19892) Reported by Mark Michelson
2012-06-06 17:20 +0000 [r368604] Matthew Jordan <mjordan@digium.com>
* build_tools/make_version: Add feature modifier to versions
produced from branches Certain branches, such as Certified
Asterisk, may have a modifier added to them that specifies the
features available in that branch. For branches, this modifier is
expected to be reflected in the location of the branch in
subversion. For example, a subversion of URL of
/certified/branches/1.8.11 would have a feature modifier of
'certified'. This is slightly different then how features are
determined for tags, where the feature is part of the actual tag
name, e.g., "10.5.0-digiumphones". In keeping with the
nomenclature used for tags, the feature specifier for branches is
translated and placed after the revision numbers. For the example
given previously, this would result in a branch version of
"Asterisk SVN-branch-1.8.11-cert-rXXXXXX".
2012-06-06 16:07 +0000 [r368586] Kinsey Moore <kmoore@digium.com>
* channels/chan_sip.c: Ensure overlapping hold flags do not
conflict When changing between different modes of hold, the flags
were not being cleared out properly causing a failure to change
hold states. (closes issue ASTERISK-19919) Patch-by: Morten
Tryfoss Reported-by: Morten Tryfoss
2012-06-06 01:08 +0000 [r368567] Richard Mudgett <rmudgett@digium.com>
* main/features.c: Fix parked call performing a DTMF blind transfer
after being retrieved. When a parked call was retrieved from the
parking lot, it could not do a blind transfer because it caused
the involved calls to be hung up unconditionally. * Made the
ParkedCall application return the ast_bridge_call() return value.
(closes issue ABE-2862) Reported by: Vlad Povorozniuc
2012-06-05 15:26 +0000 [r368520-368533] Kinsey Moore <kmoore@digium.com>
* apps/app_minivm.c: Resolve some build warnings My newly upgraded
compiler caught these usages of uninitialized values. They
weren't actually used.
* apps/app_voicemail.c: Ensure that pages and emails are sent using
RFC822-compliant date format When localization was added to
app_voicemail, these headers were altered when they should have
remained in en_US format for RFC compliance. This reverts the
changes to those two lines. (closes issue ASTERISK-19876)
2012-06-04 21:56 +0000 [r368498] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Relay proper SIP responses on calling side.
Revision 351130 broke corect HANGUPCAUSE setting for the 404 case
in chan_sip. Other cases were also potentially broken. This patch
fixes the relaying of causes to be what they used to be. (closes
issue ASTERISK-19914) Reported by Pavel Troller Tested by Walter
Doekes (via a reviewboard test to be committed later) Patches:
chan_sip.diff uploaded by Pavel Troller (license #6302)
2012-06-04 21:10 +0000 [r368405-368469] Richard Mudgett <rmudgett@digium.com>
* UPGRADE.txt: Document BLINDTRANSFER behavior change. (issue
ASTERISK-19322) (closes issue ASTERISK-19875) Reported by: call
* main/channel.c: Fix potential deadlock between masquerade and
chan_local. * Restructure ast_do_masquerade() to not hold channel
locks while it calls ast_indicate(). * Simplify many calls to
ast_do_masquerade() since it will never return a failure now. If
it does fail internally because a channel driver callback
operation failed, the only thing ast_do_masquerade() can do is
generate a warning message about strange things may happen and
press on. * Fixed the call to ast_bridged_channel() in
ast_do_masquerade(). This change fixes half of the deadlock
reported in ASTERISK-19801 between masquerades and chan_iax.
(closes issue ASTERISK-19537) Reported by: rmudgett Tested by:
rmudgett Review: https://reviewboard.asterisk.org/r/1915/
2012-06-01 23:21 +0000 [r368308] Richard Mudgett <rmudgett@digium.com>
* apps/app_stack.c: Fix deadlock when Gosub used with alternate
dialplan switches. Attempting to remove a channel from
autoservice with the channel lock held will result in deadlock. *
Restructured gosub_exec() to not call ast_parseable_goto() and
ast_exists_extension() with the channel lock held. (closes issue
ASTERISK-19764) Reported by: rmudgett Tested by: rmudgett
2012-06-01 18:18 +0000 [r368218] Kevin P. Fleming <kpfleming@digium.com>
* channels/chan_sip.c: Improve SDP parsing warning messages *
'Unsupported media type' is only reported when that is in fact
the case, not when a supported media type is included in an 'm'
line that has an invalid format. * All warning messages related
to parsing 'm' lines now include the 'm' line contents. * (minor
bugfix) newline added to port-number-zero warning messages. *
Warning messages improved to use RFC-specified terminology for
various items. * Warnings for offers that include more than one
port for a single media type now include the media type. Review:
https://reviewboard.asterisk.org/r/1811/
2012-06-01 03:25 +0000 [r368092] Michael L. Young <elgueromexicano@gmail.com>
* funcs/func_channel.c: Add documentation to function CHANNEL for
options echocan_mode and buffers The ability to set
"echocan_mode" and "buffers" through the dialplan was added to
chan_dahdi some time ago. This patch adds some documentation to
func_channel. (Closes issue ASTERISK-19911) Reported by: Dale
Noll Tested by: Michael L. Young Patches:
asterisk-19911-branch18.diff uploaded by Michael L. Young
(license 5026) Review: https://reviewboard.asterisk.org/r/1949/
2012-05-31 18:00 +0000 [r367906-368039] Richard Mudgett <rmudgett@digium.com>
* main/db1-ast/btree/bt_open.c, apps/app_queue.c,
channels/chan_iax2.c, pbx/pbx_config.c, res/ael/pval.c,
main/tcptls.c, main/manager.c, res/res_config_odbc.c,
channels/chan_sip.c, channels/chan_agent.c, funcs/func_math.c,
main/features.c: Coverity Report: Fix issues for error type
REVERSE_INULL (core modules) * Fixes findings:
0-2,5,7-15,24-26,28-31 (issue ASTERISK-19648) Reported by: Matt
Jordan
* channels/sig_pri.c, channels/sig_ss7.c: Use the
DEADLOCK_AVOIDANCE() macro instead. (issue ASTERISK-19854)
* channels/sig_pri.c, channels/sig_ss7.c: Fix deadlock when
executing CLI "pri show channels" and "ss7 show channels"
commands. * Fix sig_pri_lock_owner() to avoid deadlock properly.
* Code pri_grab() better. * Fix sig_ss7_lock_owner() to avoid
deadlock properly. * Code ss7_grab() better. (closes issue
ASTERISK-19854) Reported by: Jaxon Patches:
jira_asterisk_19854_v1.8.6.patch (license #5621) patch uploaded
by rmudgett (Modified to do the same thing to sig_ss7) Tested by:
Jaxon
* apps/app_meetme.c: Coverity Report: Fix issues for error type
REVERSE_INULL (deprecated modules) * Fix only issue pointed out
by deprecated_REVERSE_INULL.txt for app_meetme.c in find_user().
* Change use of %i to %d in sscanf() in find_user(). The use of
%i gives unexpected parsing because it can accept hex, octal, and
decimal integer formats. * Changed other uses of %i in
app_meetme() to use %d for consistency. (issue ASTERISK-19648)
Reported by: Matt Jordan
2012-05-29 18:30 +0000 [r367843] Matthew Jordan <mjordan@digium.com>
* channels/chan_skinny.c: AST-2012-008: Fix remote crash
vulnerability in chan_skinny When a skinny session is
unregistered, the corresponding device pointer is set to NULL in
the channel private data. If the client was not in the on-hook
state at the time the connection was closed, the device pointer
can later be dereferenced if a message or channel event attempts
to use a line's pointer to said device. The patches prevent this
from occurring by checking the line's pointer in message handlers
and channel callbacks that can fire after an unregistration
attempt. (closes issue ASTERISK-19905) Reported by: Christoph
Hebeisen Tested by: mjordan, Damien Wedhorn Patches:
AST-2012-008-1.8.diff uploaded by mjordan (license 6283)
AST-2012-008-10.diff uploaded by mjordan (license 6283)
2012-05-25 16:28 +0000 [r367781] Richard Mudgett <rmudgett@digium.com>
* channels/chan_iax2.c: AST-2012-007: Fix IAX receiving HOLD
without suggested MOH class crash. * Made schedule_delivery() set
the received frame f->data.ptr to NULL if the datalen is zero. *
Fix queue_signalling() memcpy() size error. * Made
queue_signalling() not use C++ keyword variable names. (closes
issue ASTERISK-19597) Reported by: mgrobecker Patches:
jira_asterisk_19597_v1.8.patch (license #5621) patch uploaded by
rmudgett Tested by: rmudgett, Michael L. Young
2012-05-25 02:27 +0000 [r367730] Michael L. Young <elgueromexicano@gmail.com>
* channels/chan_sip.c: Fix pvt_sip for inbound call to use peer's
allowtransfer setting The pvt_sip allowtransfer was not being set
to that of the peer's setting. Therefore, the global
allowtransfer setting was being used instead which would lead to
calls not being transfered if the global setting was set to 'no'
despite the setting on the peer being 'yes' and vice versa, calls
would be allowed to transfer even if the peer's setting was 'no'
but the global setting was 'yes'. (Closes issue ASTERISK-19856)
Reported by: Jacek Tested by: Michael L. Young, Jacek Patches:
issue-asterisk-19856-branch10-v3.diff uploaded by Michael L.
Young (license 5026) Review:
https://reviewboard.asterisk.org/r/1923/
2012-05-24 22:21 +0000 [r367469-367678] Richard Mudgett <rmudgett@digium.com>
* apps/app_queue.c, apps/app_dial.c: Fix Dial I option ignored if
dial forked and one fork redirects. The Dial and Queue I option
is intended to block connected line updates and redirecting
updates. However, it is a feature that when a call is locally
redirected, the I option is disabled if the redirected call runs
as a local channel so the administrator can have an opportunity
to setup new connected line information. Unfortunately, the Dial
and Queue I option is disabled for *all* forked calls if one of
those calls is redirected. * Make the Dial and Queue I option
apply to each outgoing call leg independently. Now if one
outgoing call leg is locally redirected, the other outgoing calls
are not affected. * Made Dial not pass any redirecting updates
when forking calls. Redirecting updates do not make sense for
this scenario. * Made Queue not pass any redirecting updates when
using the ringall strategy. Redirecting updates do not make sense
for this scenario. * Fixed deadlock potential with chan_local
when Dial and Queue send redirecting updates for a local
redirect. * Converted the Queue stillgoing flag to a boolean
bitfield. (closes issue ASTERISK-19511) Reported by: rmudgett
Tested by: rmudgett Review:
https://reviewboard.asterisk.org/r/1920/
* main/pbx.c: Fix WaitExten(x,m(musicclass)) string termination.
The AST_CONTROL_HOLD MOH class from the WaitExten application can
now be queued onto a channel, passed over local channels with the
/m option, and passed over IAX channels.
2012-05-23 20:27 +0000 [r367416] Mark Michelson <mmichelson@digium.com>
* main/tcptls.c: Only call SSL_CTX_free if DO_SSL is defined.
Thanks to Paul Belanger for pointing out this error.
2012-05-23 13:06 +0000 [r367362] Matthew Jordan <mjordan@digium.com>
* channels/chan_sip.c: Update a peer's LastMsgsSent when the peer
is notified of waiting messages Previously, MWI logic utilized a
counter called 'lastmsgssent' to know whether or not MWI NOTIFY
requests had been sent to a specific peer. When MWI notifications
were changed to use the internal event framework, this value was
no longer needed for its original purpose. Hence, it was no
longer updated with the new/old message counts for a peer.
However, the value was still presented when, either by AMI or
CLI, a 'sip show peer [peer]' command was executed. The output of
the command would always display the erroneous value of
32767/65535 for 'LastMsgsSent'. This patch makes it so that the
value of lastmsgssent is updated appropriately. The value should
now display the new/old message counts for a particular peer.
(closes issue ASTERISK-17866) Reported by: Steve Davies patches
by: ast-17866-rb1272.patch (License #5041 by irroot) Modified
slightly for this commit Review:
https://reviewboard.asterisk.org/r/1939
2012-05-22 17:14 +0000 [r367266-367292] Terry Wilson <twilson@digium.com>
* include/asterisk/channel.h, main/cel.c, main/asterisk.c,
main/channel.c, include/asterisk/cel.h: Fix race condition for
CEL LINKEDID_END event This patch fixes to situations that could
cause the CEL LINKEDID_END event to be missed. 1) During a core
stop gracefully, modules are unloaded when ast_active_channels ==
0. The LINKDEDID_END event fires during the channel destructor.
This means that occasionally, the cel_* module will be unloaded
before the channel is destroyed. It seemed generally useful to
wait until the refcount of all channels == 0 before unloading, so
I added a channel counter and used it in the shutdown code. 2)
During a masquerade, ast_channel_change_linkedid is called. It
calls ast_cel_check_retire_linkedid which unrefs the linkedid in
the linkedids container in cel.c. It didn't ref the new linkedid.
Now it does. Review: https://reviewboard.asterisk.org/r/1900/
* channels/chan_sip.c: Resolve crash in subscribing for MWI
notifications ASTOBJ_UNREF sets the variable to NULL after
unreffing it, so the variable should definitely not be used after
that. To solve this in the two cases that affect subscribing for
MWI notifications, we instead save the ref locally, and unref
them in the error conditions. (closes issue ASTERISK-19827)
Reported by: B. R Review:
https://reviewboard.asterisk.org/r/1940/
2012-05-18 17:47 +0000 [r367002-367027] Mark Michelson <mmichelson@digium.com>
* channels/chan_dahdi.c, main/say.c: Address MISSING_BREAK static
analysis reports some more. This addresses core findings 4 and 6.
Moises Silva helped me by stating that a break could be safely
added to the case where it is added in chan_dahdi.c In say.c, I
have added a comment indicating that static analysis complains
but that it is currently unknown if this is correct. This fixes
all core findings of this type. (closes issue ASTERISK-19662)
reported by Matthew Jordan
* include/asterisk/tcptls.h, main/tcptls.c, channels/chan_sip.c:
Fix memory leak of SSL_CTX structures in TLS core. SSL_CTX
structures were allocated but never freed. This was a bigger
issue for clients than servers since new SSL_CTX structures could
be allocated for each connection. Servers, on the other hand,
typically set up a single SSL_CTX for their lifetime. This is
solved in two ways: 1. In __ssl_setup(), if a tcptls_cfg has an
ssl_ctx on it, it is freed so that a new one can take its place.
2. A companion to ast_ssl_setup() called ast_ssl_teardown() has
been added so that servers can properly free their SSL_CTXs.
(issue ASTERISK-19278)
2012-05-18 15:42 +0000 [r366944] Matthew Jordan <mjordan@digium.com>
* main/cli.c, channels/chan_sip.c, funcs/func_odbc.c: Fix more
memory leaks This patch adds to what was fixed in r366880.
Specifically, it addresses the following: * chan_sip: dispose of
an allocated frame in off nominal code paths in sip_rtp_read *
func_odbc: when disposing of an allocated resultset, ensure that
any rows that were appended to that resultset are also disposed
of * cli: free the created return string buffer in another off
nominal code path (issue ASTERISK-19665) Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/1922/
2012-05-18 14:16 +0000 [r366882] Kinsey Moore <kmoore@digium.com>
* channels/sip/config_parser.c: Reorder and renumber tests
appropriately It appears that a patch did not apply properly when
adding tests 12 and 13 and test 11 was duplicated. These tests
have been reordered and renumbered such that they make sense.
2012-05-18 13:58 +0000 [r366880] Matthew Jordan <mjordan@digium.com>
* res/res_calendar_caldav.c, res/res_musiconhold.c,
res/res_jabber.c, apps/app_queue.c, channels/chan_iax2.c,
main/enum.c, main/editline/term.c, main/config.c, res/res_srtp.c,
main/editline/tokenizer.c, main/cli.c, channels/chan_dahdi.c,
main/data.c, funcs/func_odbc.c, apps/app_minivm.c,
main/features.c, main/editline/readline.c,
channels/sip/config_parser.c, main/xmldoc.c, res/res_calendar.c,
apps/app_voicemail.c, res/res_rtp_asterisk.c, main/netsock2.c,
res/res_calendar_icalendar.c, res/res_calendar_exchange.c,
main/pbx.c, apps/app_page.c, channels/chan_sip.c,
funcs/func_dialgroup.c, apps/app_record.c: Fix a variety of
memory leaks This patch addresses a number of memory leaks in a
variety of modules that were found by a static analysis tool. A
brief summary of the changes: * app_minivm: free ast_str objects
on off nominal paths * app_page: free the ast_dial object if the
requested channel technology cannot be appended to the dialing
structure * app_queue: if a penalty rule failed to match any
existing rule list names, the created rule would not be inserted
and its memory would be leaked * app_read: dispose of the created
silence detector in the presence of off nominal circumstances *
app_voicemail: dispose of an allocated unique ID field for MWI
event un-subscribe requests in off nominal paths; dispose of
configuration objects when using the secret.conf option *
chan_dahdi: dispose of the allocated frame produced by
ast_dsp_process * chan_iax2: properly unref peer in CLI command
"iax2 unregister" * chan_sip: dispose of the allocated frame
produced by sip_rtp_read's call of ast_dsp_process; free memory
in parse unit tests * func_dialgroup: properly deref ao2 object
grhead in nominal path of dialgroup_read * func_odbc: free
resultset in off nominal paths of odbc_read * cli: free
match_list in off nominal paths of CLI match completion * config:
free comment_buffer/list_buffer when configuration file load is
unchanged; free the same buffers any time they were created and
config files were processed * data: free XML nodes in various
places * enum: free context buffer in off nominal paths *
features: free ast_call_feature in off nominal paths of
applicationmap config processing * netsock2: users of
ast_sockaddr_resolve pass in an ast_sockaddr struct that is
allocated by the method. Failures in ast_sockaddr_resolve could
result in the users of the method not knowing whether or not the
buffer was allocated. The method will now not allocate the
ast_sockaddr struct if it will return failure. * pbx: cleanup
hash table traversals in off nominal paths; free ignore pattern
buffer if it already exists for the specified context * xmldoc:
cleanup various nodes when we no longer need them *
main/editline: various cleanup of pointers not being freed before
being assigned to other memory, cleanup along off nominal paths *
menuselect/mxml: cleanup of value buffer for an attribute when
that attribute did not specify a value * res_calendar*: responses
are allocated via the various *_request method returns and should
not be allocated in the various write_event methods; ensure
attendee buffer is freed if no data exists in the parsed node;
ensure that calendar objects are de-ref'd appropriately *
res_jabber: free buffer in off nominal path * res_musiconhold:
close the DIR* object in off nominal paths * res_rtp_asterisk: if
we run out of ports, close the rtp socket object and free the rtp
object * res_srtp: if we fail to create the session in libsrtp,
destroy the temporary ast_srtp object (issue ASTERISK-19665)
Reported by: Matt Jordan Review:
https://reviewboard.asterisk.org/r/1922
2012-05-17 14:40 +0000 [r366791] Jonathan Rose <jrose@digium.com>
* channels/chan_sip.c: chan_sip: Fix missed locking of opposing pvt
for directmedia acl from r366547 It also required deadlock
avoidance since two sip_pvts structs needed to be locked
simultaneously. Trunk handles it differently, so this is a 1.8
and 10 patch only. (issue AST-876)
2012-05-17 12:51 +0000 [r366740] Matthew Jordan <mjordan@digium.com>
* res/res_calendar_ews.c, channels/chan_dahdi.c: Fix checking
bounds of array index after using it; improper sizeof This patch
fixes two problems pointed out by a static analysis tool. * In
chan_dahdi, when an event is handled the index of the sub channel
is first obtained. In very off nominal cases, the method that
determines the index can return a negative value. In the event
handling code, whether or not the index returned is valid was
being checked after that value was used to index into an array.
This patch makes it so the value is checked before any indexing
is done. * In res_calendar_ews, sizeof was being passed a pointer
instead of the struct to determine the amount of memory to
allocate. (issue ASTERISK-19651) Reported by: Matt Jordan (closes
issue ASTERISK-19671) Reported by: Matt Jordan
2012-05-16 15:52 +0000 [r366597-366650] Mark Michelson <mmichelson@digium.com>
* main/http.c: Fix incorrect default port number for HTTP server.
Thanks to Tzafrir Cohen for bringing this up on the Asterisk
developers mailing list.
* channels/chan_sip.c: Correct misuse of ast_strip_quoted() when
getting a Diversion header's reason parameter. The use here was
assuming that the pointer would be updated, but the updated
string is actually returned by ast_strip_quoted() instead.
2012-05-15 20:14 +0000 [r366547] Jonathan Rose <jrose@digium.com>
* channels/chan_sip.c: chan_sip: Check the right channel's host
address for directmediapermit/deny Prior to this patch, when
checking the addresses for directmediapermit and directmediadeny,
Asterisk would check the host address of the channel permit/deny
was specified, which differs from the expectations of both our
users and the development team. Instead, directmediapermit/deny
now checks against the address of the channel that the peer with
the ACL is connected to. (issue AST-876) Review:
https://reviewboard.asterisk.org/r/1899/
2012-05-14 19:57 +0000 [r366389-366409] Mark Michelson <mmichelson@digium.com>
* pbx/dundi-parser.c: Fix two more coverity constant expression
result findings. These correspond to findings 0 and 1 in the core
findings of ASTERISK-19649. After contacting Mark Spencer, he was
unsure of what the intent behind these lines of code were, so
they are being axed. For Asterisk 1.8 and 10, the output of
debugging DUNDi frames will not be changed, but for trunk the
"Retry" portion will be omitted since it does not properly
distinguish retransmissions from initial frames. (closes issue
ASTERISK-19649) Reported by Matthew Jordan
* channels/chan_sip.c: Fix broken reinvite glare scenario. To make
a long story short, reinvite glares were broken because Asterisk
would invert the To and From headers when ACKing a 491 response.
The reason was because the initreq of the dialog was being
changed to the incoming glared reinvite instead of being set to
the outgoing glared reinvite. This change has three parts * In
handle_incoming, we never will reject an ACK because it has a
to-tag present, even if we think the request may be out of
dialog. * In handle_request_invite, we do not change the initreq
when receiving a reinvite to which we will respond with a 491. *
In handle_request_invite, several superflous settings up
pendinginvite have been removed since this is dones automatically
by transmit_response_reliable Review:
https://reviewboard.asterisk.org/r/1911
2012-05-11 23:53 +0000 [r366296] Russell Bryant <russell@russellbryant.com>
* addons/format_mp3.c: format_mp3: Fix a possible crash mp3_read().
This patch fixes a potential crash in mp3_read() by not assuming
that dbuf has enough data to finish filling up the output buffer.
The patch also makes sure that the dbuf state gets reset after we
know we read everything out of it already. In passing, this patch
includes some other cleanups of this module, including stripping
trailing whitespace, formatting fixes based on coding guidelines,
and removing a number of unused members from the private state
struct. (closes issue ASTERISK-19761) Reported by: Chris
Maciejewsk Tested by: Chris Maciejewsk
2012-05-10 23:38 +0000 [r366240] Richard Mudgett <rmudgett@digium.com>
* main/channel.c: * Made ast_change_name() hold the channels
container lock while changing the channel name. * Eliminate
redundant list not empty check in clone_variables().
2012-05-10 20:50 +0000 [r366167] Kinsey Moore <kmoore@digium.com>
* main/devicestate.c, pbx/dundi-parser.c, channels/chan_iax2.c,
channels/iax2-parser.c, main/config.c, res/res_monitor.c,
main/channel.c, main/cdr.c, res/ael/pval.c, main/data.c,
channels/chan_dahdi.c, main/tcptls.c, main/manager.c,
main/features.c, main/app.c, main/event.c, pbx/pbx_dundi.c,
res/res_odbc.c, main/xmldoc.c, apps/app_voicemail.c,
funcs/func_speex.c, main/pbx.c, res/res_calendar_icalendar.c,
channels/chan_sip.c, funcs/func_lock.c, channels/chan_agent.c,
channels/sip/reqresp_parser.c: Resolve FORWARD_NULL static
analysis warnings This resolves core findings from ASTERISK-19650
numbers 0-2, 6, 7, 9-11, 14-20, 22-24, 28, 30-32, 34-36, 42-56,
82-84, 87, 89-90, 93-102, 104, 105, 109-111, and 115. Finding
numbers 26, 33, and 29 were already resolved. Those skipped were
either extended/deprecated or in areas of code that shouldn't be
disturbed. (Closes issue ASTERISK-19650)
2012-05-10 16:47 +0000 [r366094] Jonathan Rose <jrose@digium.com>
* channels/iax2-provision.c, apps/app_queue.c,
channels/chan_iax2.c, res/ael/ael.flex, funcs/func_devstate.c,
main/asterisk.c, main/db.c, main/xmldoc.c, apps/app_voicemail.c,
main/pbx.c, channels/sig_analog.c, channels/chan_sip.c,
funcs/func_lock.c, main/features.c, main/acl.c: Coverity Report:
Fix issues for error type CHECKED_RETURN for core (issue
ASTERISK-19658) Reported by: Matt Jordan Review:
https://reviewboard.asterisk.org/r/1905/
2012-05-10 16:10 +0000 [r366052] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Close the proper tcptls_session when session
creation fails. (issue AST-998) Reported by: Thomas Arimont
Tested by: Thomas Arimont
2012-05-10 15:35 +0000 [r365989-366048] Jonathan Rose <jrose@digium.com>
* apps/app_chanspy.c, apps/app_page.c, funcs/func_cdr.c,
main/features.c, apps/app_disa.c: Coverity Report: Fix issues for
error type UNINIT in Core supported modules (issue
ASTERISK-19652) Reported by: Matt Jordan Review:
https://reviewboard.asterisk.org/r/1909/
* codecs/codec_dahdi.c: Block on frameout if the hardware has
enough samples to complete a frame. Fixes some problems with
skipping audio in elaborate scenarios involving multiple codecs
by making codec_dahdi operate in a more synchronous fashion
similar to codec_g729. This change also fixes the use of file
conversion tools from Asterisk's CLI. This change may cause the
thread responsible for transcoding audio to block briefly (Shaun
Ruffell describes this as 'several milliseconds') while waiting
for the hardware transcoder. (closes issue ASTERISK-19643)
reported by: Shaun Ruffell Patches:
0001-codec_dahdi-Block-on-frameout-the-hardware-has-enoug.patch
uploaded by Shaun Ruffell (license 5417)
2012-05-09 16:11 +0000 [r365896] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Prevent sip_pvt refleak when an ast_channel
outlasts its corresponding sip_pvt. chan_sip was coded under the
assumption that a SIP dialog with an owner channel will always be
destroyed after the owner channel has been hung up. However,
there are situations where the SIP dialog can time out and auto
destruct before the corresponding channel has hung up. A typical
example of this would be if the 'h' extension in the dialplan
takes a long time to complete. In such cases,
__sip_autodestruct() would complain about the dialog being auto
destroyed with an owner channel still in place. The problem is
that even once the owner channel was hung up, the sip_pvt would
still be linked in its ao2_container because nothing would ever
unlink it. The fix for this is that if __sip_autodestruct() is
called for a sip_pvt that still has an owner channel in place,
the destruction is rescheduled for 10 seconds in the future. This
will continue until the owner channel is finally hung up. (closes
issue ASTERISK-19425) reported by David Cunningham Patches:
ASTERISK-19425.patch uploaded by Mark Michelson (License #5049)
(closes issue ASTERISK-19455) reported by Dean Vesvuio Tested by
Dean Vesvuio
2012-05-08 20:14 +0000 [r365631-365692] Richard Mudgett <rmudgett@digium.com>
* apps/app_followme.c: * Fix FollowMe memory leak on error paths in
app_exec(). * Fix FollowMe leaving recorded caller name file on
error paths in app_exec(). * Use correct buffer dimension define
in struct call_followme.moh[] and struct fm_args.namerecloc[].
This fixes unexpected namerecloc filename length restriction.
* apps/app_followme.c: * Fix accept/decline DTMF buffer overwrite
in FollowMe. * Made use MAX_YN_STRING define to make all
accept/decline DTMF buffers the same size. Just using 20 isn't
good enough when someone didn't get the memo. * Fix stupid use of
a global variable in FollowMe. (ynlongest) * Fix bit field
declarations in FollowMe. * Fix FollowMe n option documentation.
2012-05-08 15:48 +0000 [r365574] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Send more accurate identification
information in dialog-info SIP NOTIFYs. This uses the calling
channel's caller ID and connected line information to populate
the remote and local identities in the dialog-info NOTIFY when an
extension is ringing. There is a bit of an oddity here, and that
is that we seed the remote target with the To header of the
outbound call rather than the from header. This is because it was
reported that seeding with the from header caused hints to be
broken with certain SNOM devices. A comment has been added to the
code to explain this. (closes issue ASTERISK-16735) reported by
Maciej Krajewski patches: local_remote_hint2.diff uploaded by
Mark Michelson (license #5049) 16735_tweak1.diff uploaded by Mark
Michelson (license #5049) Tested by Niccolo Belli
2012-05-07 18:40 +0000 [r365476] Richard Mudgett <rmudgett@digium.com>
* tests/test_config.c: Fix type punned compiler warning in
test_config.c
2012-05-07 18:36 +0000 [r365474] Matthew Jordan <mjordan@digium.com>
* apps/app_voicemail.c, main/pbx.c: Support VoiceMail d() option
when extension does not exist in channel's context The VoiceMail
d([c]) option is documented to accept digits for a new extension
in context <c>, if played during the greeting. This option works
fine if the extension being redirected to has an extension with
the same initial digit in the channel's current context. If that
digit did not happen to exist in some extension, a dialplan match
would fail and the user would not be redirected. This patch fixes
it such that if the <c> option is used, the extensions are
matched in that context as opposed to the caller's original
context. (closes issue ASTERISK-18243) Reported by: mjordan
Tested by: mjordan Review:
https://reviewboard.asterisk.org/r/1892
2012-05-07 16:01 +0000 [r365460] Mark Michelson <mmichelson@digium.com>
* main/audiohook.c, res/res_speech.c, channels/sig_analog.c,
main/abstract_jb.c, res/res_agi.c: Fix findings 0-3, 5, and 8 for
Coverity MISSING_BREAK errors. (Issue ASTERISK-19662)
2012-05-04 22:12 +0000 [r365398] Kinsey Moore <kmoore@digium.com>
* apps/app_followme.c, channels/chan_iax2.c,
channels/sip/config_parser.c, pbx/pbx_config.c,
apps/app_chanspy.c, apps/app_stack.c, main/config.c,
apps/app_voicemail.c, channels/chan_sip.c, funcs/func_aes.c,
main/features.c: Fix many issues from the NULL_RETURNS Coverity
report Most of the changes here are trivial NULL checks. There
are a couple optimizations to remove the need to check for NULL
and outboundproxy parsing in chan_sip.c was rewritten to avoid
use of strtok. Additionally, a bug was found and fixed with the
parsing of outboundproxy when "outboundproxy=," was set. (Closes
issue ASTERISK-19654)
2012-05-04 16:24 +0000 [r365313] Richard Mudgett <rmudgett@digium.com>
* channels/chan_local.c: Fix local channel chains optimizing
themselves out of a call. * Made chan_local.c:check_bridge()
check the return value of ast_channel_masquerade(). In long
chains of local channels, the masquerade occasionally fails to
get setup because there is another masquerade already setup on an
adjacent local channel in the chain. * Made the outgoing local
channel (the ;2 channel) flush one voice or video frame per
optimization attempt. * Made sure that the outgoing local channel
also does not have any frames in its queue before the masquerade.
* Made do the masquerade immediately to minimize the chance that
the outgoing channel queue does not get any new frames added and
thus unconditionally flushed. * Made block indication -1 (Stop
tones) event when the local channel is going to optimize itself
out. When the call is answered, a chain of local channels pass
down a -1 indication for each bridge. This blizzard of -1 events
really slows down the optimization process. (closes issue
ASTERISK-16711) Reported by: Alec Davis Tested by: rmudgett, Alec
Davis Review: https://reviewboard.asterisk.org/r/1894/
2012-05-04 15:48 +0000 [r365298] Mark Michelson <mmichelson@digium.com>
* res/res_rtp_asterisk.c: Fix core FINDING 2, FINDING 3, and
FINDING 4 from Coverity's CONSTANT_EXPRESSION_RESULT report.
These three all are in RTP code that attempts to print the number
of sequence number cycles in an RTCP RR report. The code was
masking out the upper 16 bits and then shifting the number right
by 16 bits. This led to an all zero result in all cases. The fix
is to do the shift without the bit masking. (issue
ASTERISK-19649)
2012-05-03 14:54 +0000 [r365143-365159] Alexandr Anikin <may@telecom-service.ru>
* addons/ooh323c/src/h323/H323-MESSAGES.h,
addons/ooh323c/src/h323/H323-MESSAGESEnc.c,
addons/ooh323c/src/ooh323.c: Fix warning of Coverity Static
analysis, change H225ProtocolIdentifier from value to pointer per
functions that use this. (close issue ASTERISK-19670) Reported
by: Matt Jordan Patches: ASTERISK-19670.patch (License #5415)
* addons/ooh323c/src/ooq931.c: Fix coverity static analysis
warning, allocate full ie structure instead of without data
buffer (close issue ASTERISK-19674) Reported by: Matt Jordan
Patches: ASTERISK-19674.patch (License #5415)
2012-06-04 Asterisk Development Team <asteriskteam@digium.com>
* Asterisk 1.8.13.0 Released.
2012-05-30 Asterisk Development Team <asteriskteam@digium.com>
* Asterisk 1.8.13.0-rc2 Released.
* Resolve crash in subscribing for MWI notifications.
ASTOBJ_UNREF sets the variable to NULL after unreffing it, so the
variable should definitely not be used after that. To solve this in
the two cases that affect subscribing for MWI notifications, we
instead save the ref locally, and unref them in the error
conditions.
(closes issue ASTERISK-19827)
Reported by: B. R.
Review: https://reviewboard.asterisk.org/r/1940/
* AST-2012-007
* AST-2012-008
2012-05-03 Asterisk Development Team <asteriskteam@digium.com>
* Asterisk 1.8.13.0-rc1 Released.
2012-05-02 17:02 +0000 [r365006-365068] Terry Wilson <twilson@digium.com>
* main/cel.c, channels/chan_local.c: Don't leak a ref if out of
memory and can't link the linkedid If the ao2_link fails, we are
most likely out of memory and bad things are going to happen.
Before those bad things happen, make sure to clean up the
linkedid references. This patch also adds a comment explaining
why linkedid can't be passed to both local channel allocations
and combines two ao2_ref calls into 1. Review:
https://reviewboard.asterisk.org/r/1895/
* main/cel.c, channels/chan_local.c: Fix a CEL LINKEDID_END race
and local channel linkedids This patch has the ;2 channel inherit
the linkedid of the ;1 channel and fixes the race condition by no
longer scanning the channel list for "other" channels with the
same linkedid. Instead, cel.c has an ao2 container of linkedid
strings and uses the refcount of the string as a counter of how
many channels with the linkedid exist. Not only does this
eliminate the race condition, but it also allows us to look up
the linkedid by the hashed key instead of traversing the entire
channel list. Review: https://reviewboard.asterisk.org/r/1895/
2012-05-01 23:11 +0000 [r364902] Richard Mudgett <rmudgett@digium.com>
* main/astobj2.c: Fixed __ao2_ref() validating user_data twice.
(closes issue ASTERISK-19755) Reported by: Gunther Kelleter
Patches: ao2_ref.patch (license #6372) patch uploaded by Gunther
Kelleter
2012-05-01 23:08 +0000 [r364899] Mark Michelson <mmichelson@digium.com>
* funcs/func_volume.c: Fix Coverity-reported ARRAY_VS_SINGLETON
error. As it turned out, this wasn't a huge deal. We were calling
ast_app_parse_options() for a set of options of which none took
arguments. The proper thing to do for this case is to pass NULL
for the "args" parameter here. We were instead passing a
seemingly-randomly chosen char * from the function. While this
would never get written to, you can rest assured things would
have gotten bad had new options (which took arguments) been added
to func_volume. (closes issue ASTERISK-19656)
2012-05-01 21:37 +0000 [r364841] Jason Parker <jparker@digium.com>
* main/manager.c: Prevent a potential crash when using manager
hooks. Found by me while poking at DPMA-127.
2012-05-01 21:36 +0000 [r364840] Richard Mudgett <rmudgett@digium.com>
* channels/chan_local.c: * Fix error path resouce leak in
local_request(). * Restructure local_request() to reduce
indentation.
2012-05-01 19:03 +0000 [r364786] Kinsey Moore <kmoore@digium.com>
* apps/app_confbridge.c: Play conf-placeintoconf message to the
correct channel Correct the code in app_confbridge to play the
conf-placeintoconf message to the marked user entering the bridge
instead of to the conference while the marked user hears silence.
(closes issue ASTERISK-19641) Reported-by: Mark A Walters
2012-05-01 18:16 +0000 [r364769] Jonathan Rose <jrose@digium.com>
* main/app.c: Fix bad check in voicemail functions for
ast_inboxcount2_func Check looks for ast_inboxcount_func instead
of ast_inboxcount2_func on ast_inboxcount2_func calls. (closes
issue ASTERISK-19718) Reported by: Corey Farrell Patches:
ast_app_inboxcount2-null-refcheck.patch uploaded by Corey Farrell
(license 5909)
2012-04-30 19:39 +0000 [r364706] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Revert improved identities sent in
dialog-info NOTIFY requests in r360862 Revision 360862 was
intended to improve identities sent in dialog-info NOTIFY
requests. Some users reported that hint became broken once this
was done. It's not clear exactly what part of the patch has
caused this regression, but broken hints are bad. For now, this
revision is being reverted so that the next releases of Asterisk
do not have bad behavior in them. The original reported issue
will have to be fixed differently in the next version of
Asterisk. (issue ASTERISK-16735)
2012-04-30 16:37 +0000 [r364649] Alexandr Anikin <may@telecom-service.ru>
* addons/ooh323cDriver.c: Fix use freed pointer in return value
from call thread (issue ASTERISK-19663) Reported by: Matt Jordan
Patches: ASTERISK-19663-ooh323.patch (License #5415)
2012-04-30 15:51 +0000 [r364635] Mark Murawki <markm@intellasoft.net>
* main/logger.c: Sanatize result from bfd_find_nearest_line
(BETTER_BACKTRACES) bfd_find_nearest_line can possibly set file
to null resulting in a crash when strrchr(file) runs (closes
issue ASTERISK-19815) Reported by Mark Murawski Tested by Mark
Murawski
2012-04-29 19:31 +0000 [r364578] Matthew Jordan <mjordan@digium.com>
* formats/format_g723.c, formats/format_h263.c,
formats/format_h264.c, formats/format_sln16.c,
formats/format_wav_gsm.c, formats/format_siren14.c,
formats/format_gsm.c, formats/format_g719.c,
formats/format_siren7.c, formats/format_g729.c,
formats/format_ilbc.c, formats/format_sln.c,
formats/format_vox.c, formats/format_wav.c, formats/format_pcm.c:
Fix error that caused truncate operations to fail Another very
inappropriate placement of a ')' (again introduced in r362151)
caused the various truncate operations to attempt to truncate the
sound file at a position of '0'. (issue ASTERISK-19655) Reported
by: Matt Jordan (issue ASTERISK-19810) Reported by: colbec
2012-04-27 21:48 +0000 [r364341] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Don't attempt to make use of the
dynamic_exclude_static ACL if DNS lookup fails. (closes issue
ASTERISK-18321) Reported by Dan Lukes Patches:
ASTERISK-18321.patch by Mark Michelson (license #5049)
2012-04-27 21:45 +0000 [r364340] Terry Wilson <twilson@digium.com>
* tests/test_config.c (added), main/config.c: Fix ast_parse_arg
numeric type range checking and add tests ast_parse_arg wasn't
checking for strto* parse errors or limiting the results by the
actual range of the numeric types. This patch fixes that and adds
unit tests as well. Review:
https://reviewboard.asterisk.org/r/1879/
2012-04-27 19:26 +0000 [r364277] Matthew Jordan <mjordan@digium.com>
* include/asterisk/time.h: Prevent overflow in calculation in
ast_tvdiff_ms on 32-bit machines The method ast_tvdiff_ms
attempts to calculate the difference, in milliseconds, between
two timeval structs, and return the difference in a 64-bit
integer. Unfortunately, it assumes that the long tv_sec/tv_usec
members in the timeval struct are large enough to hold the
calculated values before it returns. On 64-bit machines, this
might be the case, as a long may be 64-bits. On 32-bit machines,
however, a long may be less (32-bits), in which case, the
calculation can overflow. This overflow caused significant
problems in MixMonitor, which uses the method to determine if an
audio factory, which has not presented audio to an audiohook, is
merely late in providing said audio or will never provide audio.
In an overflow situation, the audiohook would incorrectly
determine that an audio factory that will never provide audio is
merely late instead. This led to situations where a MixMonitor
never recorded any audio. Note that this happened most frequently
when that MixMonitor was started by the ConfBridge application
itself, or when the MixMonitor was attached to a Local channel.
(issue ASTERISK-19497) Reported by: Ben Klang Tested by: Ben
Klang Patches: 32-bit-time-overflow-10-2012-04-26.diff (license
#6283) by mjordan (closes issue ASTERISK-19727) Reported by: Mark
Murawski Tested by: Michael L. Young Patches:
32-bit-time-overflow-2012-04-27.diff (license #6283) by mjordan)
(closes issue ASTERISK-19471) Reported by: feyfre Tested by:
feyfre (issue ASTERISK-19426) Reported by: Johan Wilfer Review:
https://reviewboard.asterisk.org/r/1889/
2012-04-27 18:57 +0000 [r364258] Kinsey Moore <kmoore@digium.com>
* channels/chan_sip.c: Allow SIP pvts involved in Replaces
transfers to fall out of reference sooner Unref the SIP pvt
stored in the refer structure as soon as it is no longer needed
so that the pvt and associated file descriptors can be freed
sooner. This change makes a reference decrement unnecessary in
code that handles SIP BYE/Also transfers which should not touch
the reference anyway. (related to issue ASTERISK-19579)
2012-04-27 14:42 +0000 [r364203] Matthew Jordan <mjordan@digium.com>
* channels/chan_sip.c: Allow for reloading SRTP crypto keys within
the same SIP dialog As a continuation of the patch in r356604,
which allowed for the reloading of SRTP keys in re-INVITE
transfer scenarios, this patch addresses the more common case
where a new key is requested within the context of a current SIP
dialog. This can occur, for example, when certain phones request
a SIP hold. Previously, once a dialog was associated with an SRTP
object, any subsequent attempt to process crypto keys in any SDP
offer - either the current one or a new offer in a new SIP
request - were ignored. This patch changes this behavior to only
ignore subsequent crypto keys within the current SDP offer, but
allows future SDP offers to change the keys. (issue
ASTERISK-19253) Reported by: Thomas Arimont Tested by: Thomas
Arimont Review: https://reviewboard.asteriskorg/r/1885/
2012-04-26 21:10 +0000 [r364060-364108] Richard Mudgett <rmudgett@digium.com>
* apps/app_directed_pickup.c: Update Pickup application
documentation. (With feeling this time.)
* main/features.c: Fix DTMF atxfer running h exten after the wrong
bridge ends. When party B does an attended transfer of party A to
party C, the attending bridge between party B and C should not be
running an h exten when the bridge ends. Running an h exten now
sets a softhangup flag to ensure that an AGI will run in dead AGI
mode. * Set the AST_FLAG_BRIDGE_HANGUP_DONT on the party B
channel for the attending bridge between party B and C. (closes
issue AST-870) (closes issue ASTERISK-19717) Reported by: Mario
(closes issue ASTERISK-19633) Reported by: Andrey Solovyev
Patches: jira_asterisk_19633_v1.8.patch (license #5621) patch
uploaded by rmudgett Tested by: rmudgett, Andrey Solovyev, Mario
2012-04-26 19:24 +0000 [r364046] Terry Wilson <twilson@digium.com>
* main/asterisk.c: Add more constness to the end_buf pointer in the
netconsole issue ASTERISK-18308 Review:
https://reviewboard.asterisk.org/r/1876/
2012-04-26 13:24 +0000 [r363986] Kinsey Moore <kmoore@digium.com>
* channels/chan_sip.c: Fix reference leaks involving SIP Replaces
transfers The reference held for SIP blind transfers using the
Replaces header in an INVITE was never freed on success and also
failed to be freed in some error conditions. This caused a file
descriptor leak since the RTP structures in use at the time of
the transfer were never freed. This reference leak and another
relating to subscriptions in the same code path have now been
corrected. (closes issue ASTERISK-19579)
2012-04-26 09:44 +0000 [r363934] Alec L Davis <sivad.a@paradise.net.nz>
* channels/chan_sip.c: chan_sip: [general] maxforwards, not checked
for a value greater than 255 The peer maxforwards is checked for
both '< 1' and '> 255', but the default 'maxforwards' in the
[general] section is only checked for '< 1' alecdavis (license
585) Reported by: alecdavis Tested by: alecdavis Review:
https://reviewboard.asterisk.org/r/1888/
2012-04-26 03:11 +0000 [r363375-363875] Richard Mudgett <rmudgett@digium.com>
* apps/app_directed_pickup.c: Update Pickup application
documentation. (Even better)
* apps/app_directed_pickup.c: Update Pickup application
documentation.
* channels/sig_pri.c, channels/chan_dahdi.c: Make
DAHDISendCallreroutingFacility wait 5 seconds for a reply before
disconnecting the call. Some switches may not handle the
call-deflection/call-rerouting message if the call is
disconnected too soon after being sent. Asteisk was not waiting
for any reply before disconnecting the call. * Added a 5 second
delay before disconnecting the call to wait for a potential
response if the peer does not disconnect first. (closes issue
ASTERISK-19708) Reported by: mehdi Shirazi Patches:
jira_asterisk_19708_v1.8.patch (license #5621) patch uploaded by
rmudgett Tested by: rmudgett
* channels/sig_pri.c, channels/sig_pri.h, channels/chan_dahdi.c:
Clear ISDN channel resetting state if the peer continues to use
it. Some ISDN switches occasionally fail to send a RESTART
ACKNOWLEDGE in response to a RESTART request. * Made the second
SETUP received after sending a RESTART request clear the channel
resetting state as if the peer had sent the expected RESTART
ACKNOWLEDGE before continuing to process the SETUP. The peer may
not be sending the expected RESTART ACKNOWLEDGE. (issue
ASTERISK-19608) (issue AST-844) (issue AST-815) Patches:
jira_ast_815_v1.8.patch (license #5621) patch uploaded by
rmudgett (modified)
* main/features.c: Fix recalled party B feature flags for a failed
DTMF atxfer. 1) B calls A with Dial option T 2) B DTMF atxfer to
C 3) B hangs up 4) C does not answer 5) B is called back 6) B
answers 7) B cannot initiate transfers anymore * Add dial
features datastore to recalled party B channel that is a copy of
the original party B channel's dial features datastore. *
Extracted add_features_datastore() from
add_features_datastores(). * Renamed struct ast_dial_features
features_caller and features_callee members to my_features and
peer_features respectively. These better names eliminate the need
for some explanatory comments. * Simplified code accessing the
struct ast_dial_features datastore. (closes issue ASTERISK-19383)
Reported by: lgfsantos
* main/features.c: Hangup affected channel in error paths of
bridge_call_thread().
2012-04-23 16:02 +0000 [r363209] Tilghman Lesher <tilghman@meg.abyt.es>
* main/astfd.c: On some platforms, O_RDONLY is not a flag to be
checked, but merely the absence of O_RDWR and O_WRONLY. The POSIX
specification does not mandate how these 3 flags must be
specified, only that one of the three must be specified in every
call.
2012-04-23 14:33 +0000 [r363141] Jonathan Rose <jrose@digium.com>
* main/manager.c, /: AST-2012-004: Fix an error that allows AMI
users to run shell commands sans authorization. As detailed in
the advisory, AMI users without write authorization for SYSTEM
class AMI actions were able to run system commands by going
through other AMI commands which did not require that
authorization. Specifically, GetVar and Status allowed users to
do this by setting their variable/s options to the SHELL or EVAL
functions. Also, within 1.8, 10, and trunk there was a similar
flaw with the Originate action that allowed users with originate
permission to run MixMonitor and supply a shell command in the
Data argument. That flaw is fixed in those versions of this
patch. (closes issue ASTERISK-17465) Reported By: David Woolley
Patches: 162_ami_readfunc_security_r2.diff uploaded by jrose
(license 6182) 18_ami_readfunc_security_r2.diff uploaded by jrose
(license 6182) 10_ami_readfunc_security_r2.diff uploaded by jrose
(license 6182) ........ Merged revisions 363117 from
http://svn.asterisk.org/svn/asterisk/branches/1.6.2
2012-04-23 14:05 +0000 [r363102-363106] Matthew Jordan <mjordan@digium.com>
* channels/chan_sip.c: AST-2012-006: Fix crash in UPDATE handling
when no channel owner exists If Asterisk receives a SIP UPDATE
request after a call has been terminated and the channel has been
destroyed but before the SIP dialog has been destroyed, a
condition exists where a connected line update would be attempted
on a non-existing channel. This would cause Asterisk to crash.
The patch resolves this by first ensuring that the SIP dialog has
an owning channel before attempting a connected line update. If
an UPDATE request is received and no channel is associated with
the dialog, a 481 response is sent. (closes issue ASTERISK-19770)
Reported by: Thomas Arimont Tested by: Matt Jordan Patches:
ASTERISK-19278-2012-04-16.diff uploaded by Matt Jordan (license
6283)
* /, channels/chan_skinny.c: AST-2012-005: Fix remotely exploitable
heap overflow in keypad button handling When handling a keypad
button message event, the received digit is placed into a fixed
length buffer that acts as a queue. When a new message event is
received, the length of that buffer is not checked before placing
the new digit on the end of the queue. The situation exists where
sufficient keypad button message events would occur that would
cause the buffer to be overrun. This patch explicitly checks that
there is sufficient room in the buffer before appending a new
digit. (closes issue ASTERISK-19592) Reported by: Russell Bryant
........ Merged revisions 363100 from
http://svn.asterisk.org/svn/asterisk/branches/1.6.2
2012-04-21 01:44 +0000 [r362997] Richard Mudgett <rmudgett@digium.com>
* apps/app_dial.c: Update app_dial M and U option GOTO return value
documentation.
2012-04-20 16:09 +0000 [r362815-362868] Terry Wilson <twilson@digium.com>
* main/asterisk.c: OpenBSD doesn't have rawmemchr, use strchr
(closes issue ASTERISK-19758) Reported by: Barry Miller Tested
by: Terry Wilson Patches: 362758-diff uploaded by Barry Miller
(license 5434)
* apps/app_speech_utils.c: Document Speech* apps hangup on failure
and suggest TryExec The Speech API apps return -1 on failure,
which will hang up the channel. This may not be desirable
behavior for some, but it isn't something that can be changed
without breaking people's dialplans or writing an option to all
of the Speech apps that does what TryExec already does. This
patch documents the hangup behavior of the apps, and suggests
TryExec as the solution. (closes issue AST-813)
2012-04-19 21:58 +0000 [r362729] Walter Doekes <walter+asterisk@wjd.nu>
* funcs/func_version.c: Fix documentation for
${VERSION(ASTERISK_VERSION_NUM)}.
2012-04-19 21:05 +0000 [r362680] Michael L. Young <elgueromexicano@gmail.com>
* tests/test_linkedlists.c, tests/test_poll.c: Add leading and
trailing backslashes A couple of unit tests did not have have
leading or trailing backslashes when setting their test category
resulting in a warning message being displayed. Added the
backslash where needed.
2012-04-19 20:59 +0000 [r362677] Richard Mudgett <rmudgett@digium.com>
* configs/queues.conf.sample: Update membermacro and membergosub
documentation in queues.conf.sample.
2012-04-19 15:53 +0000 [r362586] Sean Bright <sean@malleable.com>
* apps/app_externalivr.c: Prevent a crash in ExternalIVR when the
'S' command is sent first. If the first command sent from an
ExternalIVR client is an 'S' command, we were blindly removing
the first element from the play list and deferencing it, even if
it was NULL. This corrects that and also locks appropriately in
one place. (issue ASTERISK-17889) Reported by: Chris Maciejewski
2012-04-19 14:26 +0000 [r362536] Terry Wilson <twilson@digium.com>
* main/asterisk.c: Handle multiple commands per connection via
netconsole Asterisk would accept multiple NULL-delimited CLI
commands via the netconsole socket, but would occasionally miss a
command due to the command not being completely read into the
buffer. This patch ensures that any partial commands get moved to
the front of the read buffer, appended to, and properly sent.
(closes issue ASTERISK-18308) Review:
https://reviewboard.asterisk.org/r/1876/
2012-04-19 02:08 +0000 [r362485] Matthew Jordan <mjordan@digium.com>
* apps/app_sms.c, main/stdtime/localtime.c, utils/extconf.c,
addons/chan_mobile.c, main/asterisk.c, channels/chan_unistim.c,
main/frame.c, main/tdd.c, main/jitterbuf.c: Fix a variety of
potential buffer overflows * chan_mobile: Fixed an overrun where
the cind_state buffer (an integer array of size 16) would be
overrun due to improper bounds checking. At worst, the buffer can
be overrun by a total of 48 bytes (assuming 4-byte integers),
which would still leave it within the allocated memory of struct
hfp. This would corrupt other elements in that struct but not
necessarily cause any further issues. * app_sms: The array imsg
is of size 250, while the array (ud) that the data is copied into
is of size 160. If the size of the inbound message is greater
then 160, up to 90 bytes could be overrun in ud. This would
corrupt the user data header (array udh) adjacent to ud. *
chan_unistim: A number of invalid memmoves are corrected. These
would move data (which may or may not be valid) into the ends of
these buffers. * asterisk: ast_console_toggle_loglevel does not
check that the console log level being set is less then or equal
to the allowed log levels of 32. * frame: In
ast_codec_pref_prepend, if any occurrence of the specified codec
is not found, the value used to index into the array pref->order
would be one greater then the maximum size of the array. *
jitterbuf: If the element being placed into the jitter buffer
lands in the last available slot in the jitter history buffer,
the insertion sort attempts to move the last entry in the buffer
into one slot past the maximum length of the buffer. Note that
this occurred for both the min and max jitter history buffers. *
tdd: If a read from fsk_serial returns a character that is
greater then 32, an attempt to read past one of the statically
defined arrays containing the values that character maps to would
occur. * localtime: struct ast_time and tm are not the same size
- ast_time is larger, although it contains the elements of tm
within it in the same layout. Hence, when using memcpy to copy
the contents of tm into ast_time, the size of tm should be used,
as opposed to the size of ast_time. * extconf: this treats
ast_timing's minmask array as if it had a length of 48, when it
has defined the size of the array as 24. pbx.h defines minmask as
having a size of 48. (issue ASTERISK-19668) Reported by: Matt
Jordan
2012-04-18 16:20 +0000 [r362428] Richard Mudgett <rmudgett@digium.com>
* channels/sig_pri.c, channels/sig_pri.h, channels/chan_dahdi.c,
configs/chan_dahdi.conf.sample: Add ability to ignore layer 1
alarms for BRI PTMP lines. Several telcos bring the BRI PTMP
layer 1 down when the line is idle. When layer 1 goes down,
Asterisk cannot make outgoing calls. Incoming calls could fail as
well because the alarm processing is handled by a different code
path than the Q.931 messages. * Add the layer1_presence
configuration option to ignore layer 1 alarms when the telco
brings layer 1 down. This option can be configured by span while
the similar DAHDI driver teignorered=1 option is system wide.
This option unlike layer2_persistence does not require libpri
v1.4.13 or newer. Related to JIRA AST-598 JIRA ABE-2845
2012-04-17 21:18 +0000 [r362355-362368] Matthew Jordan <mjordan@digium.com>
* main/frame.c: Handle case where an unknown format is used to get
the preferred codec size In ast_codec_pref_getsize, if an unknown
format is passed to the method, no preferred codec will be
selected and a negative number will be used to index into the
format list. The method now logs an unknown format as a warning,
and returns an empty format list. (issue ASTERISK-19655) Reported
by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1863/
* res/res_musiconhold.c, res/res_rtp_asterisk.c, res/res_agi.c: Fix
places in resources where a negative return value could impact
execution This patch addresses a number of modules in resources
that did not handle the negative return value from function calls
adequately. This includes: * res_agi.c: if the result of the read
function is a negative number, indicating some failure, the
result would instead be treated as the number of bytes read. This
patch now treats negative results in the same manner as an end of
file condition, with the exception that it also logs the error
code indicated by the return. * res_musiconhold.c: if spawn_mp3
fails to assign a file descriptor to srcfd, and instead assigns a
negative value, that file descriptor could later be passed to
functions that require a valid file descriptor. If spawn_mp3
fails, we now immediately retry instead of continuing in the
logic. * res_rtp_asterisk.c: if no codec can be matched between
two RTP instances in a peer to peer bridge, we immediately return
instead of attempting to use the codec payload type as an index
to determine the appropriate negotiated codec. (issue
ASTERISK-19655) Reported by: Matt Jordan Review:
https://reviewboard.asterisk.org/r/1863/
* main/asterisk.c, main/manager.c, main/translate.c: Fix places in
main where a negative return value could impact execution This
patch addresses a number of modules in main that did not handle
the negative return value from function calls adequately, or were
not sufficiently clear that the conditions leading to improper
handling of the return values could not occur. This includes: *
asterisk.c: A negative return value from the read function would
be used directly as an index into a buffer. We now check for
success of the read function prior to using its result as an
index. * manager.c: Check for failures in mkstemp and lseek when
handling the temporary file created for processing data returned
from a CLI command in action_command. Also check that the result
of an lseek is sanitized prior to using it as the size of a
memory map to allocate. * translate.c: Note in the appropriate
locations where powerof cannot return a negative value, due to
proper checks placed on the inputs to that function. (issue
ASTERISK-19655) Reported by: Matt Jordan Review:
https://reviewboard.asterisk.org/r/1863/
* funcs/func_env.c: Fix places where a negative return from ftello
could be used as invalid input In a variety of locations in both
reading and writing a file, the result from the C library
function ftello is used as input to other functions. For the
parameters and functions in question, a negative value is invalid
input. This patch checks the return value from the ftello
function to determine if we were able to determine the current
position in the file stream and, if not, fail gracefully. (issue
ASTERISK-19655) Reported by: Matt Jordan Review:
https://reviewboard.asterisk.org/r/1863/
2012-04-17 20:43 +0000 [r362354] Jonathan Rose <jrose@digium.com>
* main/utils.c, res/res_config_curl.c, res/res_config_pgsql.c,
res/res_config_odbc.c: Make use of va_args more appropriate to
form in various res_config modules plus utils. A number of
va_copy operations weren't matched with a corresponding va_end in
res_config_odbc. Also, there was a potential for va_end to be
invoked twice on the same va_arg in utils, which would mean
invoking va_end on an undefined variable... which is bad. va_end
is removed from various functions in config_pgsql and config_curl
since they aren't making their own copy. The invokers of those
functions are responsible for calling va_end on them. (issue
ASTERISK-19451) Reported by: Walter Doekes Review:
https://reviewboard.asterisk.org/r/1848/
2012-04-17 18:25 +0000 [r362304] Matthew Jordan <mjordan@digium.com>
* formats/format_sln16.c, formats/format_wav_gsm.c,
formats/format_siren14.c, formats/format_gsm.c,
formats/format_g719.c, formats/format_siren7.c,
formats/format_sln.c, formats/format_vox.c, formats/format_wav.c,
formats/format_pcm.c: Fix error that caused seek format
operations to set max file size to '1' or '0' A very
inappropriate placement of a ')' (introduced in r362151) caused
the maximum size of a file to be set as the result of a
comparison operation, as opposed to the result of the ftello
operation. This resulted in seeking being restricted to the
beginning of the file, or 1 byte into the file. Thanks to the
Asterisk Test Suite for properly freaking out about this on at
least one test. (issue ASTERISK-19655) Reported by: Matt Jordan
2012-04-17 02:37 +0000 [r362253] Michael L. Young <elgueromexicano@gmail.com>
* channels/chan_sip.c: Turn off warning message when bind address
is set to any. When a bind address is set to an ANY address
(udpbindport=::), a warning message is displayed stating that
"Address remapping activated in sip.conf but we're using IPv6,
which doesn't need it. Please remove 'localnet' and/or
'externaddr' settings." But if one is running dual stack, we
shouldn't be told to turn those settings off. This patch checks
if the bind address is an ANY address or not. The warning message
will now only be displayed if the bind address is NOT an ANY
address and IPv6 is being used. Also, updated the copyright year.
(closes issue ASTERISK-19456) Reported by: Michael L. Young
Tested by: Michael L. Young Patches: chan_sip_ipv6_message.diff
uploaded by Michael L. Young (license 5026)
2012-04-16 21:56 +0000 [r362151-362204] Matthew Jordan <mjordan@digium.com>
* channels/chan_dahdi.c, channels/chan_agent.c: Fix negative return
handling in channel drivers In chan_agent, while handling a
channel indicate, the agent channel driver must obtain a lock on
both the agent channel, as well as the channel the agent channel
is using. To do so, it attempts to lock the other channel first,
then unlock the agent channel which is locked prior to entry into
the indicate handler. If this unlock fails with a negative return
value, which can occur if the object passed to agent_indicate is
an invalid ao2 object or is NULL, the return value is passed
directly to strerror, which can only accept positive integer
values. In chan_dahdi, the return value of dahdi_get_index is
used to directly index into the sub-channel array. If
dahd_get_index returns a negative value, it would use that value
to index into the array, which could cause an invalid memory
access. If dahdi_get_index returns a negative number, we now
default to SUB_REAL. (issue ASTERISK-19655) Reported by: Matt
Jordan Review: https://reviewboard.asterisk.org/r/1863/
* apps/app_voicemail.c: Fix handling of negative return code when
storing voicemails in ODBC storage When storing a voicemail
message using an ODBC connection to a database, the voicemail
message is first stored on disk. The sound file associated with
the message is read into memory before being transmitted to the
database. When this occurs, a failure in the C library's lseek
function would cause a negative value to be passed to the mmap as
the size of the memory map to create. This would almost certainly
cause the creation of the memory map to fail, resulting in the
message being lost. (issue ASTERISK-19655) Reported by: Matt
Jordan Review: https://reviewboard.asterisk.org/r/1863
* formats/format_g723.c, formats/format_h263.c,
formats/format_h264.c, formats/format_sln16.c,
formats/format_wav_gsm.c, formats/format_siren14.c,
formats/format_gsm.c, formats/format_g719.c,
formats/format_siren7.c, formats/format_g729.c,
formats/format_ilbc.c, formats/format_sln.c,
formats/format_vox.c, formats/format_wav.c, formats/format_pcm.c:
Check for IO stream failures in various format's truncate/seek
operations For the formats that support seek and/or truncate
operations, many of the C library calls used to determine or set
the current position indicator in the file stream were not being
checked. In some situations, if an error occurred, a negative
value would be returned from the library call. This could then be
interpreted inappropriately as positional data. This patch checks
the return values from these library calls before using them in
subsequent operations. (issue ASTERISK-19655) Reported by: Matt
Jordan Review: https://reviewboard.asterisk.org/r/1863/
2012-04-13 15:54 +0000 [r362079-362082] Jonathan Rose <jrose@digium.com>
* apps/app_forkcdr.c: Make ForkCDR e option not set end time of the
newly forked CDR log Prior to this patch, ForkCDR's e option
would immediately set the end time of the forked CDR to that of
the CDR that is being terminated. This resulted in the new CDR's
end time being roughly the same as it's beginning time (which is
in turn roughly the same as the original's end time). (closes
issue ASTERISK-19164) Reported by: Steve Davies Patches:
cdr_fork_end.v10.patch uploaded by Steve Davies (license 5012)
* apps/app_meetme.c: Send relative path named recordings to the
meetme directory instead of sounds Prior to this patch, no effort
was made to parse the path name to determine a proper destination
for recordings of MeetMe's r option. This fixes that. Review:
https://reviewboard.asterisk.org/r/1846/
2012-04-12 16:18 +0000 [r361955-361972] Kinsey Moore <kmoore@digium.com>
* channels/chan_iax2.c: Make trunkfreq take effect when set
Previously, setting trunkfreq had no effect on initial load or on
reload and only ever used the default value. This causes
trunkfreq to be used appropriately on initial load and reload.
(closes issue ASTERISK-19521) Patch-by: Jaco Kroon
* Makefile.rules, makeopts.in, codecs/lpc10/Makefile, Makefile,
build_tools/cflags.xml, build_tools/menuselect-deps.in,
codecs/gsm/src/k6opt.s, configure, codecs/gsm/Makefile,
configure.ac: Simplify build system architecture optimization
This change to the build system rips out any usage of PROC along
with architecture-specific optimizations in favor of using
-march=native where it is supported. This fixes broken builds on
64bit Intel systems and results in better optimized code on
systems running GCC 4.2+. Review:
https://reviewboard.asterisk.org/r/1852/ (closes issue
ASTERISK-19462)
2012-04-10 21:43 +0000 [r361854] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c: Prevent invalid access of free'd memory if
DAHDI channel during an MWI event In the MWI processing loop,
when a valid event occurs the temporary caller ID information is
deallocated. If a new DAHDI channel is successfully created, the
event is passed up to the analog_ss_thread without error and the
loop exits. If, however, the DAHDI channel is not created, then
the caller ID struct has been free'd, and the gains reset to
their previous level. This will almost certainly cause an invalid
access to the free'd memory, either in subsequent calls to
callerid_free or calls to callerid_feed. * Rework the -r361705
patch to better manage the cs and mtd allocated resources. *
Fixed use of mwimonitoractive flag to be correct if the
mwi_thread() fails to start.
2012-04-10 19:57 +0000 [r361657-361803] Matthew Jordan <mjordan@digium.com>
* main/http.c: Fix crash caused by unloading or reloading of
res_http_post When unlinking itself from the registered HTTP
URIs, res_http_post could inadvertently free all URIs registered
with the HTTP server. This patch modifies the unregister method
to only free the URI that is actually being unregistered, as
opposed to all of them.
* funcs/func_curl.c: Allow func_curl to exit gracefully if list
allocation fails during write If the global_curl_info data
structure could not be allocated, the datastore associated with
the operation would be free'd, but the function would not return.
This would later dereference the datastore, almost certainly
causing Asterisk to crash. With this patch, if the data structure
is not allocated the method will return an error code, and not
attempt any further operation.
* channels/chan_dahdi.c: Prevent invalid access of free'd memory if
DAHDI channel during an MWI event In the MWI processing loop,
when a valid event occurs the temporary caller ID information is
deallocated. If a new DAHDI channel is successfully created, the
event is passed up to the analog_ss_thread without error and the
loop exits. If, however, the DAHDI channel is not created, then
the caller ID struct has been free'd, and the gains reset to
their previous level. This will almost certainly cause an invalid
access to the free'd memory, either in subsequent calls to
callerid_free or calls to callerid_feed. This patch makes it so
that we only free the caller ID structure if a DAHDI channel is
successfully created, and we bump the gains back up if we fail to
make a DAHDI channel.
* funcs/func_global.c: Change SHARED function to use a safe
traversal when modifying a variable When the SHARED function
modifies a variable, it removes it from its list of variables and
reinserts the new value at the head of the list of variables.
Doing this inside a standard list traversal can be dangerous, as
the standard list traversal does not account for the list being
changed. While the code in question should not cause a use after
free violation due to its breaking out of the loop after freeing
the variable, it could lead to a maintenance issue if the loop
was modified. This also fixes a violation reported by a static
analysis tool, which also makes this code easier to maintain in
the future.
2012-04-06 21:50 +0000 [r361558-361606] Matthew Jordan <mjordan@digium.com>
* res/res_calendar_ews.c: Fix memory leak in res_calendar_ews when
event email address node is empty If the XML calendar data
returned by a Microsoft Exchange Web Service specifies an XML
Event E-Mail Address ("EmailAddress"), and no e-mail address is
provided, a condition existed where an ast_calendar_attendee
struct would be allocated but not appended to the list of
attendees. Because of that, the memory associated with the
attendee would never be freed. This patch frees the memory if no
e-mail address is provided.
* apps/app_meetme.c: Fix memory leak when using MeetMeAdmin 'e'
option with user specified A memory leak/reference counting leak
occurs if the MeetMeAdmin 'e' command (eject last user that
joined) is used in conjunction with a specified user. Regardless
of the command being executed, if a user is specified for the
command, MeetMeAdmin will look up that user. Because the 'e'
option kicks the last user that joined, as opposed to the one
specified, the reference to the user specified by the command
would be leaked when the user variable was assigned to the last
user that joined.
2012-04-06 18:09 +0000 [r361471] Kinsey Moore <kmoore@digium.com>
* apps/app_ices.c, channels/chan_gtalk.c, channels/chan_iax2.c,
res/res_config_sqlite.c, res/res_srtp.c, main/cdr.c,
main/tcptls.c, funcs/func_channel.c, channels/console_gui.c,
apps/app_sms.c, apps/app_chanspy.c, addons/chan_mobile.c,
channels/chan_mgcp.c, main/xmldoc.c, apps/app_voicemail.c,
res/res_clioriginate.c, channels/chan_unistim.c, main/pbx.c,
channels/chan_sip.c, res/res_fax.c, funcs/func_strings.c,
channels/console_video.c, formats/format_ogg_vorbis.c: Add
missing newlines to CLI logging
2012-04-06 16:27 +0000 [r361403-361412] Paul Belanger <paul.belanger@polybeacon.com>
* funcs/func_sysinfo.c: Fix typo in svn:keywords
* bridges/bridge_multiplexed.c, bridges/bridge_builtin_features.c:
Fix typo in svn:keywords
2012-04-06 15:47 +0000 [r361380] Russell Bryant <russell@russellbryant.com>
* apps/rpt_flow.pdf (removed), configs/rpt.conf.sample (removed),
configs/usbradio.conf.sample (removed): Remove a few more files
related to chan_usbradio and app_rpt.
2012-04-06 14:01 +0000 [r361332] Matthew Jordan <mjordan@digium.com>
* channels/chan_sip.c: Fix a typo in the warning messages for an
ignored media stream Added a '\n' to the warning messages when we
ignore a media stream due to the port number being '0'. (closes
issue ASTERISK-19646) Reported by: Badalian Vyacheslav
2012-04-06 13:30 +0000 [r361329] Kinsey Moore <kmoore@digium.com>
* apps/app_dial.c: Remove unnecessary error message in app_dial.c
The error message for failure to stop autoservice after a gosub
or macro call during a dial was removed for macro while Asterisk
1.4 was still being actively developed. The corresponding gosub
error message was never removed. (closes issue ASTERISK-19551)
2012-04-05 16:36 +0000 [r361201-361269] Jonathan Rose <jrose@digium.com>
* apps/app_meetme.c: Fix MusicOnHold in MeetMe so that it always
uses the class if it's been defined There were a few instances of
restarting music on hold in meetme that would cause Asterisk to
revert to the default class of music on hold for no adequate
reason. Review: https://reviewboard.asterisk.org/r/1844/
* addons/ooh323cDriver.c: Fix some stuff involving calls to memcpy
and memset The important parts of the patch were already applied
through other updates. (closes issue ASTERISK-19445) Reported by:
Makoto Dei Patches: memset-memcpy-length.patch uploaded by Makoto
Dei (license 5027)
* funcs/func_devstate.c: Make 'help devstate change' display
properly (get rid of excess comma) (closes issue ASTERISK-19444)
Reported by: Makoto Dei Patches:
devstate-change-usage-truncate.patch uploaded by Makoto Dei
(license 5027)
2012-05-02 Asterisk Development Team <asteriskteam@digium.com>
* Asterisk 1.8.12.0 Released.
2012-05-01 Asterisk Development Team <asteriskteam@digium.com>
* Asterisk 1.8.12.0-rc3 Released.
* channels/chan_sip.c: Revert revision 360862
Revision 360862 was intended to improve identities sent in
dialog-info NOTIFY requests. Some users reported that hint became
broken once this was done. It's not clear exactly what part of
the patch has caused this regression, but broken hints are bad.
For now, this revision is being reverted so that the next releases of
Asterisk do not have bad behavior in them. The original reported
issue will have to be fixed differently in the next version of
Asterisk.
(issue ASTERISK-16735)
2012-04-24 Asterisk Development Team <asteriskteam@digium.com>
* Asterisk 1.8.12.0-rc2 Released.
* AST-2012-004
* AST-2012-005
* AST-2012-006
2012-04-04 Asterisk Development Team <asteriskteam@digium.com>
* Asterisk 1.8.12.0-rc1 Released.
2012-04-04 16:29 +0000 [r361090-361142] Jonathan Rose <jrose@digium.com>
* main/app.c, pbx/pbx_realtime.c, apps/app_externalivr.c,
channels/chan_iax2.c, apps/app_milliwatt.c, main/channel.c,
pbx/pbx_loopback.c, addons/chan_ooh323.c, channels/chan_sip.c:
Replace GNU old-style field designator extensions to fix clang
warnings (issue ASTERISK-19540) Reported by: Makoto Dei Patches:
clang-gnu-designator.patch uploaded by Makoto Dei (license 5027)
* apps/app_meetme.c: Make the MeetMeAdmin N command (mute all
nonadmins) not mute admins (Closes Issue ASTERISK-19335) Reported
by: Johan Wilfer Review: https://reviewboard.asterisk.org/r/1843/
2012-04-03 20:08 +0000 [r360987-361040] Kinsey Moore <kmoore@digium.com>
* apps/app_transfer.c: Fix the display of documentation for
Transfer This came up while fixing documentation generation for
many other cases where the argument separator was not being
displayed properly. Now that it is displayed properly, it shows
up in the wrong place for Transfer since the '/' is only required
if Tech is present. (related to issue ASTERISK-18168)
* channels/chan_sip.c: Stop sending out RTCP if RTP is inactive
This change prevents Asterisk from sending RTCP receiver reports
during a remote bridge since it is no longer receiving media and
should not be reporting anything. (related to ASTERISK-19366)
2012-03-30 21:26 +0000 [r360933] Richard Mudgett <rmudgett@digium.com>
* main/logger.c: Fix logger deadlock on Asterisk shutdown. The
logger_thread() had an exit path that failed to release the
logmsgs list lock. * Make logger_thread() exit path unlock the
logmsgs list lock. * Made ast_log() not queue any messages to the
logmsgs list if the close_logger_thread flag is set. (issue
ASTERISK-19463) Reported by: Matt Jordan
2012-03-29 23:32 +0000 [r360862-360884] Mark Michelson <mmichelson@digium.com>
* main/features.c: Fix potential race condition during call pickup.
Prior to this patch, a connected line update was queued during
call pickup and then an answer frame was queued. The original
caller would presumably then have his connected line updated and
then the call would be answered. In actuality, the answer frame
was not how the call ended up being answered. Rather, an odd
section in app_dial that checks if the called channel's state is
up. The result is that the order of the connected line update and
the answer were variable. In most cases, this wasn't actually a
bad thing. However, if the 'I' option was passed to dial, the
connected line update would be inhibited. The fix is to queued
the connected line after the answer frame is queued. This way the
race in app_dial is between two conditions resulting in an
answer. This way the connected line update occurs after the
answer every time. (closes issue ASTERISK-19183) Reported by:
Thomas Arimont Tested by: Thomas Arimont Mark Michelson Patches:
ASTERISK-19183.patch uploaded by Mark Michelson (license 5049)
* channels/chan_sip.c: Improve accuracy of identifying information
sent in dialog-info SIP NOTIFY requests. This change makes use of
connected party information in addition to caller ID in order to
populate local and remote XML elements in the dialog-info
NOTIFYs. (closes issue ASTERISK-16735) Reported by: Maciej
Krajewski Tested by: Maciej Krajewski Patches:
local_remote_hint2.diff uploaded by Mark Michelson (license 5049)
2012-03-28 19:06 +0000 [r360712] Terry Wilson <twilson@digium.com>
* cdr/cdr_adaptive_odbc.c, addons/cdr_mysql.c,
channels/chan_gtalk.c, channels/chan_jingle.c,
addons/chan_ooh323.c: Destroy configs when they are no longer
used https://reviewboard.asterisk.org/r/1834/
2012-03-27 16:59 +0000 [r360625] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Make a debug message regarding subscription
changes more accurate. I was getting confused during some testing
why Asterisk was saying that a subscription was being added when
it was clearly being removed. This fixes that confusion.
2012-03-27 14:32 +0000 [r360488-360574] Jonathan Rose <jrose@digium.com>
* configure: Updates config with bootstrap where I changed
configure.ac in r360488 (issue ASTERISK-17842) Reported by: Bryon
Clark
* configure.ac: Fix BETTER_BACKTRACES library detection for
Fedora/RedHat/CentOS (closes ASTERISK-17842) Reported by: Bryon
Clark Patches: 20110512__issue19278.diff.txt uploaded by Tilghman
Lesher (license 5003) configure_bfd_with_dl_and_iberty.patch
uploaded by Bryon Clark (license 6157)
2012-03-26 18:37 +0000 [r360471-360474] Paul Belanger <pabelanger@digium.com>
* CHANGES: Update CHANGES for r360471
* CHANGES: Fix Asterisk version typo
* main/dnsmgr.c: Increase verbosity level for ast_verb messages
While this does not fix the issue of the CLI being flooded by
'doing dnsmgr_lookup' messages, increasing the verbosity level
above 5 should help minimize it.
2012-03-24 23:46 +0000 [r360356-360413] Russell Bryant <russell@russellbryant.com>
* funcs/func_curl.c: func_curl: Fix leak of an ast_str in error
handling code path.
* apps/app_page.c: app_page: Fix a memory leak on every Page().
dial_list is a dynamically allocated array that is allocated at
the beginning of Page() based on how many devices will be dialed.
This was never being freed.
* apps/app_jack.c: app_jack: fix datastore memory leak in error
handling path.
* res/ael/ael.tab.h, main/ast_expr2.c, main/ast_expr2.h,
res/ael/ael.tab.c, main/ast_expr2f.c, res/ael/ael_lex.c: Rebuild
parsers. This is needed to include the last fix to
main/ast_expr2.y. The changes look much bigger as this
regeneration of the code was done with newer versions of flex and
bison.
* main/ast_expr2.y: expression parser: Fix (theoretical) memory
leak. Fix a memory leak that is very unlikely to actually happen.
If a malloc() succeeded, but the following strdup() failed, the
memory from the original malloc() would be leaked.
2012-03-24 00:35 +0000 [r360262-360309] Richard Mudgett <rmudgett@digium.com>
* channels/sig_pri.c, main/channel.c: Make number not available
presentation also set screening to network provided. Q.951
indicates that when the presentation indicator is "Number not
available due to interworking" for a number then the screening
indicator field should be "Network provided". * Made
ast_party_id_presentation() return AST_PRES_NUMBER_NOT_AVAILABLE
when the presentation is "Number not available due to
interworking". This fix makes Asterisk consistent and it also
makes it consistent with earlier branches as far as this
presentation value is concerned. * Made pri_to_ast_presentation()
and ast_to_pri_presentation() conversions handle the "Number not
available due to interworking" case better in sig_pri.c. This
change is possible because the minimum required libpri version
(v1.4.11) has the necessary defines in libpri.h.
* channels/chan_sip.c: Add missing initialization of
update_redirecting in chan_sip.c
2012-03-21 14:51 +0000 [r360138] Jonathan Rose <jrose@digium.com>
* contrib/scripts/install_prereq: Update install_prereq script to
include missing GSM library for debian amd move SQLite3. (closes
issue ASTERISK-19367) Reported by: Andrew Latham Patches:
debian_install_prereq.diff uploaded by Andrew Latham (license
5985)
2012-03-21 13:19 +0000 [r360087] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
* configure, configure.ac: Also detect gmime 2.6 Also detect gmime
version 2.6 (Michael Biebl) Signed-off-by: Tzafrir Cohen (License
#5035) <tzafrir.cohen@xorcom.com>
2012-03-21 13:19 +0000 [r360086] Matthew Jordan <mjordan@digium.com>
* channels/chan_sip.c: Ensure Asterisk sends a BYE when pending on
the final response to a re-INVITE When Asterisk detects a hangup
and cannot send a BYE due to a pending INVITE, it sets the
pendingbye flag and waits for the final response to that INVITE.
When the response is received, it transmits the BYE. If, however,
that INVITE request is a pending re-INVITE, it needs to first
send a CANCEL request to terminate the pending re-INVITE. In that
circumstance, Asterisk was, in some scenarios, clearing the
pendingbye flag after processing the CANCEL request and not
checking for a pending BYE when receiving the final 487 response
to the INVITE. This patch ensures that if the pendingbye flag is
set, it is honored regardless of the nature of the INVITE request
currently in flight. (closes issue ASTERISK-19365) Reported by:
Thomas Arimont Tested by: Thomas Arimont Patches:
bugASTERISK-19365_2012_03_08.patch uploaded by mjordan (license
6283) Review: https://reviewboard.asterisk.org/r/1807
2012-03-20 20:32 +0000 [r360033] Kinsey Moore <kmoore@digium.com>
* apps/app_echo.c: Prevent Echo() from relaying control, null, and
modem frames Echo()'s description states that it echoes audio,
video, and DTMF except for # while it actually echoes any frame
that it receives other than DTMF #. This was causing frame storms
in the test suite in some circumstances where Echo() was attached
to both ends of a pair of local channels and control frames were
being periodically generated. Echo()'s behavior and description
have been modifed so that it only echoes media and non-# DTMF
frames.
2012-03-20 17:21 +0000 [r359979] Richard Mudgett <rmudgett@digium.com>
* include/asterisk/manager.h, main/manager.c: Allow AMI action
callback to be reentrant. Fix AMI module reload deadlock
regression from ASTERISK-18479 when it tried to fix the race
between calling an AMI action callback and unregistering that
action. Refixes ASTERISK-13784 broken by ASTERISK-17785 change.
Locking the ao2 object guaranteed that there were no active
callbacks that mattered when ast_manager_unregister() was called.
Unfortunately, this causes the deadlock situation. The patch
stops locking the ao2 object to allow multiple threads to invoke
the callback re-entrantly. There is no way to guarantee a module
unload will not crash because of an active callback. The code
attempts to minimize the chance with the registered flag and the
maximum 5 second delay before ast_manager_unregister() returns.
The trunk version of the patch changes the API to fix the race
condition correctly to prevent the module code from unloading
from memory while an action callback is active. * Don't hold the
lock while calling the AMI action callback. (closes issue
ASTERISK-19487) Reported by: Philippe Lindheimer Review:
https://reviewboard.asterisk.org/r/1818/ Review:
https://reviewboard.asterisk.org/r/1820/
2012-03-16 20:13 +0000 [r359892] Jonathan Rose <jrose@digium.com>
* apps/app_chanspy.c: Prevent chanspy from binding to zombie
channels This patch addresses a bug with chanspy on local
channels which roughly 50% of the time would create a situation
where chanspy can latch onto a zombie channel, keeping the zombie
alive forever and causing the channel doing the spying to never
be able to hang up. (closes issue ASTERISK-19493) Reported by:
lvl Review: https://reviewboard.asterisk.org/r/1819/
2012-03-16 08:22 +0000 [r359809] Alec L Davis <sivad.a@paradise.net.nz>
* channels/sip/include/sip.h: Missed lastinvite CSeq int to
uint32_t change from Review:
https://reviewboard.asterisk.org/r/1699/
2012-03-15 19:01 +0000 [r359656-359706] Matthew Jordan <mjordan@digium.com>
* main/utils.c: Fix remotely exploitable stack overflow in HTTP
manager There exists a remotely exploitable stack buffer overflow
in HTTP digest authentication handling in Asterisk. The
particular method in question is only utilized by HTTP AMI. When
parsing the digest information, the length of the string is not
checked when it is copied into temporary buffers allocated on the
stack. This patch fixes this behavior by parsing out pre-defined
key/value pairs and avoiding unnecessary copies to the stack.
(closes issue ASTERISK-19542) Reported by: Russell Bryant Tested
by: Matt Jordan
* apps/app_milliwatt.c, /: Fix remotely exploitable stack overrun
in Milliwatt Milliwatt is vulnerable to a remotely exploitable
stack overrun when using the 'o' option. This occurs due to the
milliwatt_generate function not accounting for
AST_FRIENDLY_OFFSET when calculating the maximum number of
samples it can put in the output buffer. This patch resolves this
issue by taking into account AST_FRIENDLY_OFFSET when determining
the maximum number of samples allowed. Note that at no point is
remote code execution possible. The data that is written into the
buffer is the pre-defined Milliwatt data, and not custom data.
(closes issue ASTERISK-19541) Reported by: Russell Bryant Tested
by: Matt Jordan Patches: milliwatt_stack_overrun.rev1.txt by
Russell Bryant (license 6283) Note that this patch was written by
Russell, even though Matt uploaded it ........ Merged revisions
359645 from http://svn.asterisk.org/svn/asterisk/branches/1.6.2
2012-03-15 18:17 +0000 [r359609] Richard Mudgett <rmudgett@digium.com>
* apps/app_queue.c, apps/app_dial.c: Add missing connected line
macro calls to initial dial for Dial and Queue apps. The
connected line interception macros do not get executed when the
outgoing channel is initially created and that channel's
caller-id is implicitly imported into the incoming channel's
connected line data. If you are using the interception macros,
you would expect that they get run for every change to a
channel's connected line information outside of normal dialplan
execution. Review: https://reviewboard.asterisk.org/r/1817/
2012-03-15 00:52 +0000 [r359452-359558] Russell Bryant <russell@russellbryant.com>
* channels/chan_iax2.c: chan_iax2: Fix use of uninitialized
sockaddr_in in try_transfer(). Initialize a struct sockaddr_in in
try_transfer() so that the code isn't (potentially) trying to
read from it while uninitialized.
* channels/chan_gtalk.c: chan_gtalk: Fix use of uninitialized vars
in config handling. Fix potential use of context, parkinglot, and
prefs before they are initialized.
* channels/chan_gtalk.c: chan_gtalk: Fix potential use of
uninitialized variable. Avoid potential use of idroster in
gtalk_alloc() before it has been initialized.
* apps/app_chanisavail.c: app_chanisavail: Fix use of uninitialized
variable. Ensure that status is set before it is used by
resetting it during each loop iteration. This could have resulted
in incorrect results from this app.
* main/udptl.c: udptl: Ensure fec[] in udptl_build_packet() is
initialized. Scan results indicated that this array could be used
uninitialized. At a quick look, it looks correct. In any case,
initializing it is a Good Thing (tm).
* include/asterisk/app.h: app.h: Always initialize
AST_DECLARE_APP_ARGS(). This patch ensures that the struct
defined by AST_DECLARE_APP_ARGS() is always fully initialized.
I'm not sure if this fixes any real bugs, but it silences a bunch
of warnings from coverity, and is generally a good thing to do
anyway.
2012-03-14 22:20 +0000 [r359451] Richard Mudgett <rmudgett@digium.com>
* include/asterisk/channel.h, main/channel.c,
channels/chan_agent.c: Fix deadlock potential with some
ast_indicate/ast_indicate_data calls. Calling
ast_indicate()/ast_indicate_data() with the channel lock held can
result in a deadlock with a local channel because of how local
channels need to avoid deadlock.
2012-03-14 17:32 +0000 [r359356] Matthew Jordan <mjordan@digium.com>
* main/jitterbuf.c: Fix incorrect jitter buffer overflow due to
missed resynchronizations When a change in time occurs, such that
the timestamps associated with frames being placed into an
adaptive jitter buffer (implemented in jitterbuf.c) are
significantly different then the previously inserted frames, the
jitter buffer checks to see if it needs to be resynched to the
new time frame. If three consecutive packets break the threshold,
the jitter buffer resynchs itself to the new timestamps. This
currently only occurs when history is calculated, and hence only
on JB_TYPE_VOICE frames. JB_TYPE_CONTROL frames, on the other
hand, are never passed to the history calculations. Because of
this, if the jump in time is greater then the maximum allowed
length of the jitter buffer, the JB_TYPE_CONTROL frames are
dropped and no resynchronization occurs. Alterntively, if the
overfill logic is not triggered, the JB_TYPE_CONTROL frame will
be placed into the buffer, but with a time reference that is not
applicable. Subsequent JB_TYPE_VOICE frames will quickly trigger
the overflow logic until reads from the jitter buffer reach the
errant JB_TYPE_CONTROL frame. This patch allows JB_TYPE_CONTROL
frames to resynch the jitter buffer. As JB_TYPE_CONTROL frames
are unlikely to occur in multiples, it perform the
resynchronization on any JB_TYPE_CONTROL frame that breaks the
resynch threshold. Note that this only impacts chan_iax2, as
other consumers of the adaptive jitter buffer use the abstract
jitter buffer API, which does not use JB_TYPE_CONTROL frames.
Review: https://reviewboard.asterisk.org/r/1814/ (closes issue
ASTERISK-18964) Reported by: Kris Shaw Tested by: Kris Shaw, Matt
Jordan Patches: jitterbuffer-2012-2-26.diff uploaded by Kris Shaw
(license 5722)
2012-03-14 17:17 +0000 [r359344] Richard Mudgett <rmudgett@digium.com>
* apps/app_dial.c, main/channel.c: Fix Dial m and r options and
forked calls generating warnings for voice frames. When connected
line support was added, the wait_for_answer() variable single
changed its meaning slightly. Unfortunately, the places where
single was used did not necessarily get updated to reflect that
change. Also audio/video frames were sent to all forked calls
when the endpoints were never made compatible. * Don't pass
audio/video media frames when the channels have not been made
compatible. * Added handling of AST_CONTROL_SRCCHANGE to
app_dial.c. * Fixed app_dial.c passing on AST_CONTROL_HOLD
because that frame can also pass a requested MOH class. (closes
issue ASTERISK-16901) Reported by: Chris Gentle (closes issue
ASTERISK-17541) Reported by: clint Review:
https://reviewboard.asterisk.org/r/1805/
2012-03-14 10:52 +0000 [r359050-359259] Russell Bryant <russell@russellbryant.com>
* include/asterisk/logger.h, main/logger.c: Fix bogus reads/writes
of console log levels in asterisk.c This patch updates the
NUMLOGLEVELS define in logger.h to 32, to match the fact that
logger.c implements 32 log levels (because of the custom log
level stuff). asterisk.c uses this define to size an array of
levels per remote console. This array is modified in
ast_console_toggle_loglevel(), which is called by the "logger set
level" CLI command. While the documentation for the CLI command
doesn't make it terribly obvious, you can use this CLI command to
toggle a custom log level on a remote console, as well. However,
doing so led to an invalid array index in asterisk.c. This array
is read from any time a log message is written to a console. So,
all custom log level messages resulted in a bogus read if a
remote console was connected.
* apps/app_externalivr.c, channels/chan_iax2.c: Fix invalid
reads/writes due to incorrect sizeof(). These few places in the
code used sizeof() on h_addr in struct hostent. This is
sizeof(char *). The correct way to get the size of this address
is to use h_length. This error would result in reads/writes of 8
bytes instead of 4 on 64-bit machines.
* main/sched.c: Fix inaccurate sizeof() in sched.c. This code just
needed sizeof(int), not sizeof(int *).
* utils/astman.c: Fix incorrect sizeof() in astman.
* res/res_crypto.c: Fix incorrect usage of sizeof() in res_crypto.
In this case, just remove the memset(). There was a redundant
memset that is done correctly just 2 lines later.
* res/res_adsi.c: Fix broken usage of sizeof() in res_adsi.
* main/features.c: Fix incorrect sizeof() usage in features.c. This
didn't actually result in a bug anywhere, luckily. The only place
where the result of these memcpys was used is in app_dial, and
the only field that it read out of ast_call_feature was the first
one, which is an int, so these memcpys always copied just enough
to avoid a problem.
* main/md5.c: Fix incorrect sizeof() on a pointer in MD5Final().
* main/pbx.c: Don't use a buffer after it goes out of scope. 's' is
set to 'workspace'. Make sure 'workspace' doesn't go out of scope
while the reference to it via 's' is still used.
* res/ais/ais.h, res/res_ais.c, res/ais/clm.c, res/ais/evt.c: Dump
cache of published events when a node joins the cluster. Also use
a more reliable method for stopping the poll() thread.
* makeopts.in, apps/app_rpt.c (removed), channels/chan_usbradio.c
(removed), channels/xpmr (removed),
build_tools/menuselect-deps.in, configure,
include/asterisk/autoconfig.h.in, configure.ac: Remove
chan_usbradio and app_rpt. These modules are being maintained
outside of the tree and have been for a long time now, so it
doesn't make sense to keep them here. Review:
https://reviewboard.asterisk.org/r/1764/
2012-03-13 20:31 +0000 [r358943-358978] Terry Wilson <twilson@digium.com>
* main/features.c: Fix setting CDR variables in the hangup
extension A previous CDR fix for setting CDR variables during a
bridge via custom dialplan features broke setting CDR variables
in the hangup extension. This patch fixes the issue. Review:
https://reviewboard.asterisk.org/r/1794/
* main/devicestate.c, include/asterisk/devicestate.h,
channels/chan_sip.c, tests/test_devicestate.c: Make hints for
invalid SIP devices return Unavail, not idle This patch
drastically simplifies the device state aggegation code. The old
method was not only overly complex, but also made it impossible
to return AST_DEVICE_INVALID from the aggregation code. The unit
test update is as a result of fixing that bug. The SIP change
stems from a bug introduced by removing a DNS lookup for
hostname-based SIP channels. (closes issue ASTERISK-16702)
Review: https://reviewboard.asterisk.org/r/1808/
2012-03-13 16:54 +0000 [r358810-358859] Tilghman Lesher <tilghman@meg.abyt.es>
* UPGRADE.txt, CHANGES: Requested changes documenting the fixed AEL
functionality.
* utils/ael_main.c, apps/app_stack.c, utils/conf2ael.c,
res/ael/pval.c, funcs/func_dialplan.c, tests/test_gosub.c: Enable
macros in 1.8 to find the next highest "h" extension in a
context, like in 1.4. This change restores functionality that was
present in 1.4, when AEL macros were implemented with the Macro
dialplan application. Macros are fraught with functionality
issues, because they consume a large portion of the underlying
application stack. This limits the ability of AEL users to call
many layers of subroutines, an issue which Gosub does not have
(originally tested to 100,000 levels deep). Therefore, starting
in 1.6.0, AEL macros were implemented with Gosub. However, there
were some implicit behaviors of Macro, which were not replicated
at the same time as with the transition to Gosub, one of which is
documented in the related issue. In particular, the "h" extension
is designed to execute not in the Macro context, but in the
topmost calling context. Due to legacy issues with a misapplied
bugfix many years ago, when a macro exited in 1.4, it looks in
all calling contexts, bubbling up from the deepest level until it
finds an "h" extension. Since AEL hides the complexity of the
underlying dialplan logic from the AEL programmer, it's
reasonable to assume that this behavior should not change in the
transition from Asterisk 1.4 LTS to Asterisk 1.8 LTS, lest we
break working AEL configurations in the transition to Asterisk
1.8 LTS. This fix is the result, which implements a search for
the "h" extension in all calling Gosub contexts. Fixes
ASTERISK-19336 Patch: 20120308__ael_bugfix_for_trunk__2.diff
(License #5003) by Tilghman Lesher (with slight modifications for
1.8) Tested by: Johan Wilfer Review:
https://reviewboard.asterisk.org/r/1776/
2012-03-08 16:39 +0000 [r358643] Jonathan Rose <jrose@digium.com>
* channels/chan_sip.c: Make transfer not ignore port information
with SIP. Attempting to transfer with SIP to an address like
1XXXXX@ip.ad.re.ss:5061 would fail because port would be cut from
the host string and ignored. This simply keeps chan_sip from
cutting off the port number during these kinds of transfers.
(closes issue ASTERISK-19321) Reported by: Federico Alves Review:
https://reviewboard.asterisk.org/r/1790/diff/#index_header
2012-03-07 18:25 +0000 [r358530] Richard Mudgett <rmudgett@digium.com>
* channels/sig_ss7.c: Change directly setting _softhangup in
sig_ss7.c to use ast_softhangup_nolock(). Update to: (issue
ASTERISK-19372)
2012-03-07 16:11 +0000 [r358484] Sean Bright <sean@malleable.com>
* codecs/codec_dahdi.c: Return g729 and g723.1 frames with the
number of samples set properly. If the wctc4xxp returns more than
a single packet, we need to update the number of samples in the
returned frame accordingly. Acked-by: Shaun Ruffell
<sruffell@digium.com>
2012-03-07 15:16 +0000 [r358435-358438] Terry Wilson <twilson@digium.com>
* configs/cdr_adaptive_odbc.conf.sample: Set snarkiness = 0 in
cdr_adaptive_odbc.conf.sample
* cdr/cdr_adaptive_odbc.c, cel/cel_odbc.c: Add detection for ODBC
WCHAR fields Without detecting these types, cel_odbc blows up
when the character set for the table is utf8. This also wraps
cdr_adaptive_odbc's use of those types in the HAVE_ODBC_WCHAR
#ifdef seen in other parts of the code.
2012-03-06 17:44 +0000 [r358260-358377] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c: Fix ring cadance setup for outgoing calls
on FXS ports. * Fix referencing the wrong variable in
chan_dahdi.c:my_set_cadence(). Thanks to Sean Bright for
compiling with -Wshadow and finding this bug.
* channels/sig_ss7.c: Drop SS7 call if not connected yet when
INCOMPLETE/BUSY/CONGESTION. SS7 is a trunk protocol and should
clear a failed call as soon as possible. * Made SS7 hangup a call
immediately if it has not connected yet for
INCOMPLETE/BUSY/CONGESTION causes. Otherwise, play an appropriate
inband tone. (closes issue ASTERISK-19372) Reported by: Igor
Nikolaev
* channels/sig_ss7.c, channels/chan_dahdi.c, channels/sig_ss7.h:
Setup DSP when SS7 call is connected or early media is available.
Outgoing SS7 calls fail to detect incoming DTMF so any bridged
channel that requires out-of-band DTMF will not work. * Added
sig_ss7_open_media() calls at appropriate places in sig_ss7.c.
The new call converts conditionaled out unconverted code and
shows that the code really did something useful. * Improved some
chan_dahdi DTMF debug messages to help track DTMF handling.
(closes issue ASTERISK-19312) Reported by: Igor Nikolaev
2012-03-05 18:49 +0000 [r358214] Jonathan Rose <jrose@digium.com>
* main/manager.c: Eliminate double close of file descriptor in
manager.c The process_output function in manager.c attempted to
call fclose and close immediately afterwards. Since fclose
implies close, this resulted in a potential double free on file
descriptors. This patch changes that behavior and also adds error
checking to fclose and close depending on which was deemed
necessary. Also error messages. Thanks to Rosen Iliev for
pointing out the location of the problem. (closes issue
ASTERISK-18453) Reported By: Jaco Kroon Review:
https://reviewboard.asterisk.org/r/1793/
2012-03-05 16:41 +0000 [r358162] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: Defer sending the connected line reinvite if
a reinvite is already in progress. (issue ASTERISK-19355)
Reported by: tomaso (closes issue AST-825)
2012-03-05 15:54 +0000 [r358115] Kinsey Moore <kmoore@digium.com>
* channels/chan_sip.c: Ensure Asterisk acknowledges ACKs to 4xx on
Replaces errors Asterisk was not setting pendinginvite in the
upper half of handle_request_invite such that the 4xx was
retransmitted repeatedly even though an ack was received for
every retransmission. (closes issue ASTERISK-19303) Patch-by:
Jeremiah Gowdy
2012-03-02 23:27 +0000 [r357986-358029] Terry Wilson <twilson@digium.com>
* channels/xpmr/xpmr.c, channels/chan_usbradio.c: Fix
unused-but-set-variable warnings All of these were pretty
obviously unused. Some were unused because the code that used
them was #if 0'd. In those cases, I just commented out the
unused-but-set variables.
* channels/misdn/isdn_msg_parser.c, channels/misdn/isdn_lib.c,
channels/chan_misdn.c: Correct some set-but-unused variable
warnings in the mISDN library. (from kpfleming's commit to trunk
r356292)
* channels/xpmr/xpmr.c: Make chan_usbradio compile under dev mode
x=++x and x=x=1? Really?
2012-03-02 21:02 +0000 [r357940] Kinsey Moore <kmoore@digium.com>
* main/event.c, include/asterisk/strings.h, main/ccss.c,
tests/test_event.c: Fix case-sensitivity for device-specific
event subscriptions and CCSS This change fixes case-sensitivity
for device-specific subscriptions such that the technology
identifier is case-insensitive while the remainder of the device
string is still case-sensitive. This should also preserve the
original case of the device string as passed in to the event
system. CCSS is the only feature affected as it is the only
consumer of device-specific event subscriptions. The second part
of this patch addresses similar case-sensitivity issues within
CCSS itself that prevented it from functioning correctly after
the fix to the events system. This adds a unit test to verify
that the event system works as expected. (closes issue
ASTERISK-19422) Review: https://reviewboard.asterisk.org/r/1780/
2012-03-02 18:34 +0000 [r357894] Richard Mudgett <rmudgett@digium.com>
* channels/sig_pri.c, main/channel.c: Remove ISDN hold restriction
for non-bridged calls. The check if an ISDN call is bridged
before it could be placed on hold is not necessary and is overly
restrictive. The check was originally done to prevent problems
with call transfers in case a user tried to transfer a call
connected to an application to another call connected to an
application. The ISDN transfer code has not required this
restriction for quite some time because ECT could transfer any
two active calls to each other. * Remove ISDN hold restriction
for calls connected to applications. * Made
ast_waitfordigit_full() ignore AST_CONTROL_HOLD and
AST_CONTROL_UNHOLD instead of generating a warning message.
(closes issue ASTERISK-19388) Reported by: Birger Harzenetter
Tested by: rmudgett
2012-03-02 15:58 +0000 [r357811] Sean Bright <sean@malleable.com>
* channels/chan_iax2.c: The default value for mohinterpret is the
empty string, so when resetting to default values don't
explicitly set the value to "default."
2012-03-02 15:45 +0000 [r357809] Richard Mudgett <rmudgett@digium.com>
* apps/app_chanspy.c: Fix channel reference leak in ChanSpy. * Fix
next_channel() channel reference leak in ChanSpy. (closes issue
ASTERISK-19461) Reported by: Irontec Patches:
app_chanspy_iteartor_next_unref.patch (license #6213) patch
uploaded by Irontec (issue ASTERISK-17515)
2012-03-02 00:59 +0000 [r357760-357761] Mark Michelson <mmichelson@digium.com>
* main/channel.c: Fix race condition that can cause important
control frames (such as a hangup) to be missed. This takes two
actions. 1. Move the reading of the alertpipe in __ast_read() to
immediately before the removal of frames from the readq. This
means we won't do something silly like read from the alertpipe,
then ignore the fact that there's a frame to get from the readq
since channel's fdno is the AST_TIMING_FD. 2. When
ast_settimeout() sets the rate to 0 and the timingfunc to NULL,
if the channel's fdno is the AST_TIMING_FD, then set the fdno to
-1. This is because if the rate is 0 and the timingfunc is NULL,
it means that the channel's timing fd is being invalidated, so
any pending reads should not occur. This may actually solve more
issues than the referenced one below, but it's not known at this
time for sure. (closes issue ASTERISK-19223) reported by
Frank-Michael Wittig Review:
https://reviewboard.asterisk.org/r/1779
* main/translate.c: Second attempt to get optimal translation paths
when codec_resample is used. This borrows code heavily from
changes made in translation code in Asterisk 10. This uses the
quality and sample rate change of translation in order to pick
paths rather than the computational cost of translations.
Computational cost is used solely in determining if a single
translation step from a specific translator is better than the
same translation step provided by a different translator. (closes
issue ASTERISK-16821) reported by Andrew Lindh Review:
https://reviewboard.asterisk.org/r/1772
2012-03-01 14:18 +0000 [r357665] Kinsey Moore <kmoore@digium.com>
* main/acl.c: Prevent outbound SIP NOTIFY packets from displaying a
port of 0 In the change from 1.6.2 to 1.8, ast_sockaddr was
introduced which changed the behavior of ast_find_ourip such that
port number was wiped out. This caused the port in internip
(which is used for Contact and Call-ID on NOTIFYs) to be 0. This
change causes ast_find_ourip to be port-preserving again. (closes
issue ASTERISK-19430)
2012-02-29 19:41 +0000 [r357575] Walter Doekes <walter+asterisk@wjd.nu>
* apps/app_dial.c: Fix copying of CDR(accountcode) to local
channels. In r203638, during the addition of the Channel Event
Logging, in mid-2009, this got broken in trunk and ended up in
asterisk 1.8 and higher. This fixes so the CDR(accountcode) from
the calling channel is available to dialed channels again as well
as showing up properly in the CDR's. (closes issue
ASTERISK-19384) Patches: accountcode.patch (License #6033) by
jamicque Review: https://reviewboard.asterisk.org/r/1775/
Reviewed by: Richard Mudgett
2012-02-28 22:27 +0000 [r357455-357490] Jonathan Rose <jrose@digium.com>
* UPGRADE.txt, configs/sip.conf.sample: Adding transport=udp to
sample sip.conf - Also changes version of Asterisk 1.8 in UPGRADE
(issue ASTERISK-19352) Reported by: jamicque Patches:
asterisk-19352-transport-warning-message-v1.patch uploaded by
Michael L. Young (license 5026)
* cdr/cdr_adaptive_odbc.c: Add additional character type types to
supported data types for cdr_adaptive_odbc The reporter was uable
to use varchar utf8_unicode_ci with cdr_adaptive_odbc, so this
patch adds those along with some other character types to the
list of types cdr_adaptive_odbc will work using the varchar
conditions. The problem wasn't really UTF8 characters as much as
it was a failure to respond to the exact type that was
declared/in use on that database. (closes issue ASTERISK-19334)
Reported By: Igor Nikolaev Patches: cdr_adaptive_odbc.patch
uploaded by Igor Nikolaev (license 6236)
2012-02-28 21:19 +0000 [r357416] Tilghman Lesher <tilghman@meg.abyt.es>
* apps/app_stack.c: Correctly reset the dialplan priority. When the
stack frame is allocated, we save the address to which we should
return, when the Gosub returns. However, if we just want to
restore the priority, then we need to subtract 1 before setting
it. Otherwise, when a Gosub goes to a nonexistent address, it
will skip a priority in the dialplan. This is because when we
return from an application, the PBX increments the priority for
us.
2012-02-28 20:57 +0000 [r357407] Richard Mudgett <rmudgett@digium.com>
* channels/sig_pri.c: Use more reasonable cause code when rejecting
incoming call waiting calls. (closes issue ASTERISK-19397)
Reported by: Birger Harzenetter Patches: nochannel-cause.patch
(license #5870) patch uploaded by Birger Harzenetter
2012-02-28 20:26 +0000 [r357356-357386] Jonathan Rose <jrose@digium.com>
* UPGRADE.txt: Moves UPGRADE.txt notes from r357356 to a new
section specific to 1.8.12 (issue ASTERISK-19352) reported by:
jamicque
* UPGRADE.txt: Adds UPGRADE.txt notes to r357266 indicating changes
to transport option (issue ASTERISK-19352) Reported by: jamicque
2012-02-28 19:32 +0000 [r357352] Richard Mudgett <rmudgett@digium.com>
* apps/app_page.c: Remove dupliate 'i' option table entry in
app_page.c. (closes issue ASTERISK-19310) Reported by: Makoto Dei
Patches: app_page-duplicate-i-option.patch (license #5027) patch
uploaded by Makoto Dei
2012-02-28 18:00 +0000 [r357266] Jonathan Rose <jrose@digium.com>
* channels/chan_sip.c: Changes transport option in sip.conf so that
using multiple instances doesn't stack. Prior to this patch,
Using "transport=" multiple times would cause them to add to one
another like allow/deny. This patch changes that behavior to
simply use the transport option specified last. Also, if no
transport option is applied now, the default will automatically
be UDP. (closes ASTERISK-19352) Reported by: jamicque Patches:
asterisk-19352-transport-warning-message-v1.patch uploaded by
Michael L. Young (license 5026)
issueA19352_no_transport_is_udp.patch uploaded by Walter Doekes
(license 5674) Review:
https://reviewboard.asterisk.org/r/1745/diff/#index_header
2012-02-28 14:45 +0000 [r357212] Kevin P. Fleming <kpfleming@digium.com>
* Makefile.rules: Make COMPILE_DOUBLE magic actually work. The
build system has some special magic to ensure that if Asterisk is
built with --enable-dev-mode *and* DONT_OPTIMIZE, that all the
source is still compiled with the optimizer enabled (even though
the result will be thrown away), because the compiler is able to
find a great deal of coding errors and bugs as a result of
running its optimizers. Unfortunately at some point this mode got
broken, and the 'throwaway' compile of the code was no longer
done with the optimizer enabled. This patch corrects that
problem.
2012-02-27 23:34 +0000 [r357093] Richard Mudgett <rmudgett@digium.com>
* main/channel.c: Fix callerid of Originated calls. Thanks to Matt
Riddell for tracking this down. (closes issue ASTERISK-19385)
Reported by: ornix
2012-03-29 Asterisk Development Team <asteriskteam@digium.com>
* Asterisk 1.8.11.0 Released.
2012-03-26 Asterisk Development Team <asteriskteam@digium.com>
* Asterisk 1.8.11.0-rc3 Released.
* AST-2012-003
* AST-2012-002
* /main/manager.c, /include/asterisk/manager.h: Fix AMI deadlock
regression by allowing AMI action callback to be reentrant
Fix AMI module reload deadlock from ASTERISK-18479 when it tried
to fix the race between calling an AMI action callback and
unregistering that action. Refixes ASTERISK-13784 broken by
ASTERISK-17785 change.
Locking the ao2 object guaranteed that there were no active
callbacks that mattered when ast_manager_unregister() was called.
Unfortunately, this causes the deadlock situation. The patch stops
locking the ao2 object to allow multiple threads to invoke the
callback re-entrantly. There is no way to guarantee a module unload
will not crash because of an active callback. The code attempts to
minimize the chance with the registered flag and the maximum 5
second delay before ast_manager_unregister() returns.
The trunk version of the patch changes the API to fix the race
condition correctly to prevent the module code from unloading from
memory while an action callback is active.
* Don't hold the lock while calling the AMI action callback.
(closes issue ASTERISK-19487)
Reported by: Philippe Lindheimer
Review: https://reviewboard.asterisk.org/r/1818/
2012-03-06 Asterisk Development Team <asteriskteam@digium.com>
* Asterisk 1.8.11.0-rc2 Released.
* main/acl.c: Prevent outbound SIP NOTIFY packets from displaying
a port of 0.
In the change from 1.6.2 to 1.8, ast_sockaddr was
introduced which changed the behavior of ast_find_ourip such
that port number was wiped out. This caused the port in
internip (which is used for Contact and Call-ID on NOTIFYs) to be
0. This change causes ast_find_ourip to be port-preserving again.
2012-01-30 21:57 +0000 [r353368-353320] Alec L Davis <sivad.a@paradise.net.nz>
* channels/sip/include/sip.h, channels/sip/include/dialog.h,
channels/chan_sip.c: RFC3261 Section 8.1.1.5. The sequence number
value MUST be expressible as a 32-bit unsigned integer * fix: use
%u instead of %d when dealing with CSeq numbers - to remove
possibility of -ve numbers. * fix: change all uses of seqno and
friends (ocseq icseq) from 'int' or 'unsigned int' to uint32_t.
Summary of CSeq numbers. An initial CSeq number must be less than
2^31 A CSeq number can increase in value up to 2^32-1 An
incrementing CSeq number must not wrap around to 0. Tested with
Asterisk 1.8.8.2 with Grandstream phones. alecdavis (license 585)
Tested by: alecdavis Review:
https://reviewboard.asterisk.org/r/1699/
* channels/chan_sip.c: prevent debug messsges displaying -ve Cseq
numbers. Missed in R353320
2012-01-30 23:17 +0000 [r353371] Terry Wilson <twilson@digium.com>
* include/asterisk/dnsmgr.h, main/dnsmgr.c, channels/chan_sip.c:
Re-link peers by IP when dnsmgr changes the IP Asterisk's dnsmgr
currently takes a pointer to an ast_sockaddr and updates it
anytime an address resolves to something different. There are a
couple of issues with this. First, the ast_sockaddr is usually
the address of an ast_sockaddr inside a refcounted struct and we
never bump the refcount of those structs when using dnsmgr. This
makes it possible that a refresh could happen after the
destructor for that object is called (despite ast_dnsmgr_release
being called in that destructor). Second, the module using dnsmgr
cannot be aware of an address changing without polling for it in
the code. If an action needs to be taken on address update (like
re-linking a SIP peer in the peers_by_ip table), then polling for
this change negates many of the benefits of having dnsmgr in the
first place. This patch adds a function to the dnsmgr API that
calls an update callback instead of blindly updating the address
itself. It also moves calls to ast_dnsmgr_release outside of the
destructor functions and into cleanup functions that are called
when we no longer need the objects and increments the refcount of
the objects using dnsmgr since those objects are stored on the
ast_dnsmgr_entry struct. A helper function for returning the
proper default SIP port (non-tls vs tls) is also added and used.
This patch also incorporates changes from a patch posted by Timo
Teräs to ASTERISK-19106 for related dnsmgr issues. (closes issue
ASTERISK-19106) Review: https://reviewboard.asterisk.org/r/1691/
2012-01-31 16:51 +0000 [r353454] Richard Mudgett <rmudgett@digium.com>
* include/asterisk/channel.h, main/manager.c: Fix memory leak in
error paths for action_originate(). * Fix memory leak of vars in
error paths for action_originate(). * Moved struct
fast_originate_helper tech and data members to stringfields. *
Simplified ActionID header handling for fast_originate(). * Added
doxygen note to ast_request() and ast_call() and the associated
channel callbacks that the data/addr parameters should be treated
as const char *. Review: https://reviewboard.asterisk.org/r/1690/
2012-01-31 23:41 +0000 [r353502] Terry Wilson <twilson@digium.com>
* res/res_calendar.c: Allow res_calendar to be unloaded The
calendaring tech modules depend on res_calendar and initially
res_calendar just bumped the use count so that it couldn't be
unloaded. res_calendar can potentially create many threads and
I've seen issues where the Asterisk shutdown has failed where it
looked like these threads could be the culprit. This patch adds
unload support for res_calendar. Unloading res_calendar will also
unload the dependant tech modules as well. (closes issue
ASTERISK-16744) Review: https://reviewboard.asterisk.org/r/1657/
2012-02-01 15:02 +0000 [r353550] Matthew Jordan <mjordan@digium.com>
* contrib/init.d/etc_default_asterisk: Added clarification for the
VERBOSITY setting to etc_default_asterisk Clarified that using
the VERBOSITY setting in etc_default_asterisk is the same as
using the -v command line switch, which causes Asterisk to launch
in console mode. (closes issue ASTERISK-17030) Reported by: Jonas
2012-02-01 15:50 +0000 [r353598] Sean Bright <sean@malleable.com>
* include/asterisk/audiohook.h: Resolve an overlap in the
ast_audiohook_flags values. AST_AUDIOHOOK_TRIGGER_WRITE and
AST_AUDIOHOOK_WANTS_DTMF were overlapping which may have caused
unintended side effects. This patch moves
AST_AUDIOHOOK_TRIGGER_WRITE, and updates
AST_AUDIOHOOK_TRIGGER_MODE to reflect the original intention.
This will affect existing modules that use these flags, so be
sure to recompile as necessary. (closes issue ASTERISK-19246)
Reported by: feyfre
2012-02-01 21:05 +0000 [r353769-353720] Jonathan Rose <jrose@digium.com>
* channels/chan_sip.c: Use ast_sockaddr_stringify_fmt wrappers for
various functions in chan_sip There are a number of cleaner
looking wrappers for ast_sockaddr_stringify_fmt available which
are slightly more readable than using a direct call to
ast_sockaddr_stringify_fmt. This patch switches a number of those
calls in chan_sip to use those wrappers and is generally
harmless. (Closes issue ASTERISK-16930) Reported by: Michael L.
Young Patches: chan_sip-broken-registration-1.8.diff uploaded by
Michael L. Young (license 5026)
* channels/chan_sip.c: Fix sip show peers port output, align
columns, and fix ami port output. A previous patch I committed
from ASTERISK-16930 unexpectedly changed some output for the AMI
action "sippeers" which this patch changes back. Also, this
aligns the output for the cli command "sip show peers" and fixes
another issue that patch introduced by using
ast_sockaddr_stringify calls multiple times without immediately
using the pointer. I also went ahead and did a little janitorial
work to clean up whitespace in _sip_show_peers. (issue
ASTERISK-16930) (closes issue ASTERISK-19281) Reported by:
Patrick El Youssef Patches: ASTERISK-19281.diff uploaded by
Walter Doekes (license 5674)
2012-02-02 16:58 +0000 [r353770] Mark Michelson <mmichelson@digium.com>
* UPGRADE.txt, configs/manager.conf.sample,
include/asterisk/manager.h, configs/http.conf.sample,
main/manager.c, main/http.c: Fix TLS port binding behavior as
well as reload behavior: * Removes references to tlsbindport from
http.conf.sample and manager.conf.sample * Properly bind to port
specified in tlsbindaddr, using the default port if specified. *
On a reload, properly close socket if the service has been
disabled. A note has been added to UPGRADE.txt to indicate how
ports must be set for TLS. (closes issue ASTERISK-16959) reported
by Olaf Holthausen (closes issue ASTERISK-19201) reported by
Chris Mylonas (closes issue ASTERISK-19204) reported by Chris
Mylonas Review: https://reviewboard.asterisk.org/r/1709
2012-02-02 18:31 +0000 [r353818] Jonathan Rose <jrose@digium.com>
* funcs/func_curl.c: Backports some documentation for func_curl
from 10 to 1.8 For some reason this function was completely
undocumented in 1.8. I copied the 10 docs over to 1.8 and removed
references to an enumerator that was added in the Asterisk 10
version of func_curl. That was the only change I noted. (closes
issue ASTERISK-19186) Reported by: Olivier Krief
2012-02-02 20:01 +0000 [r353867] Richard Mudgett <rmudgett@digium.com>
* channels/sig_pri.c, channels/sig_pri.h, channels/chan_dahdi.c:
Restore the 'w' modifier support for ISDN spans.
Dial(DAHDI/g0/1234w888) This feature also causes the sending
complete ie to be sent for switch types that do not automatically
send the ie. (EuroISDN/ETSI) The main difference between dialing
Dial(DAHDI/g0/1234w888) and Dial(DAHDI/g0/1234,,D(888)) is the
sending of the sending complete ie. (closes issue ASTERISK-19176)
Reported by: rmudgett Tested by: rmudgett
2012-02-02 22:26 +0000 [r353915] Kinsey Moore <kmoore@digium.com>
* channels/chan_sip.c: Ensure entering T.38 passthrough does not
cause an infinite loop After R340970 Asterisk was still polling
the RTCP file descriptor after RTCP is shut down and removed. If
the descriptor happened to have data ready when the removal
occured then Asterisk would go into an infinite loop trying to
read data that it can never actually access. This change disables
the audio RTCP file descriptor for the duration of the T.38
transaction. (closes issue ASTERISK-18951) Reported-by: Kristijan
Vrban
2012-02-03 21:24 +0000 [r353999] Jonathan Rose <jrose@digium.com>
* channels/chan_agent.c: Fixes deadlocks occuring in chan_agent due
to r335976 Bad locking order was added to chan_agent to prevent
segfaults from having no locking in a patch by irroot. This patch
addresses the bad locking order by releasing locks before getting
the right locking order to stop deadlocks from occuring when
doing multiple interactions with agents. (closes issue
ASTERISK-19285) Reported by: Alex Villacis Lasso Review:
https://reviewboard.asterisk.org/r/1708/
2012-02-06 17:28 +0000 [r354216-354116] Richard Mudgett <rmudgett@digium.com>
* main/features.c: Add missing headers to AMI UnParkedCall event to
uniquely identify the call. The AMI UnParkedCall event was
missing the Parkinglot and Uniqueid headers that the AMI
ParkedCall event contains. (closes issue ASTERISK-19240) Reported
by: Michael Yara
* pbx/pbx_config.c: Improved documentation of CLI "dialplan add
extension" command. * Documented dialplan add extension
<exten>,<priority>,<app(<app-data>)> format. * Allow acceptance
of command without the app-data value. There are many
applications that do no need any parameters so it is silly to
require that field for all commands. * Fixed a couple
ast_malloc/ast_free mismatches with ast_add_extension2() calls.
(closes issue ASTERISK-19222) Reported by: Andrey Solovyev Tested
by: rmudgett
2012-02-07 15:04 +0000 [r354263] Jonathan Rose <jrose@digium.com>
* cdr/cdr_pgsql.c: Fix column duplication bug in module reload for
cdr_pgsql. Prior to this patch, attempts to reload cdr_pgsql.so
would cause the column list to keep its current data and then add
a second copy during the reload. This would cause attempts to log
the CDR to the database to fail. This patch also cleans up some
unnecessary null checks for ast_free and deals with a few
potential locking problems. (closes issue ASTERISK-19216)
Reported by: Jacek Konieczny Review:
https://reviewboard.asterisk.org/r/1711/
2012-02-07 20:53 +0000 [r354348] Terry Wilson <twilson@digium.com>
* contrib/realtime/postgresql/realtime.sql, channels/chan_sip.c:
Fix multiple SIP realtime issues 1. Set lastms to 0 when clearing
instead of "" 2. Don't set ipaddr or port to the string "(null)"
when they are empty 3. Add missing required fields, set default
for lastms to 0, and modify the length of the ipaddr field to 45
in the Postgresql realtime.sql file. (closes issue
ASTERISK-19172) Review: https://reviewboard.asterisk.org/r/1703/
2012-02-09 02:23 +0000 [r354492] Russell Bryant <russell@russellbryant.com>
* main/channel.c: Remove some unnecessary locking from
ast_hangup(). This patch removes some unnecessary locking of the
channels container in ast_hangup(). The reason this came up is
that this lock can very quickly block the entire system. If any
of the channel cleanup code decides to block, it causes a problem
for the whole system. For example, when audiohooks get destroyed,
if that blocks for a while waiting on the mixmonitor thread to
exit because it's busy blocking on some I/O, it causes a problem
for many other threads in the meantime. Review:
https://reviewboard.asterisk.org/r/1712/
2012-02-09 02:52 +0000 [r354495] Richard Mudgett <rmudgett@digium.com>
* apps/app_parkandannounce.c: Fix crash in ParkAndAnnounce. Well,
thats embarrasing. I forgot to initialize the caller_id storage.
(closes issue ASTERISK-19311) Reported by: tootai Tested by:
rmudgett
2012-02-09 16:30 +0000 [r354542] Matthew Jordan <mjordan@digium.com>
* channels/chan_sip.c: Fix SIP INFO DTMF handling for non-numeric
codes In ASTERISK-18924, SIP INFO DTMF handlingw as changed to
account for both lowercase alphatbetic DTMF events, as well as
uppercase alphabetic DTMF events. When this occurred, the
comparison of the character buffer containing the event code was
changed such that the buffer was first compared again '0' and '9'
to determine if it was numeric. Unfortunately, since the first
character in the buffer will typically be '1' in the case of
non-numeric event codes (10-16), this caused those codes to be
converted to a DTMF event of '1'. This patch fixes that, and
cleans up handling of both application/dtmf-relay and
application/dtmf content types. Review:
https://reviewboard.asterisk.org/r/1722/ (closes issue
ASTERISK-19290) Reported by: Ira Emus Tested by: mjordan
2012-02-09 16:56 +0000 [r354545] Mark Michelson <mmichelson@digium.com>
* CHANGES, res/res_fax.c: Adding reload support to res_fax.so
(closes issue ASTERISK-16712) reported by Frank DiGennaro Review:
https://reviewboard.asterisk.org/r/1713
2012-02-09 17:07 +0000 [r354547] Matthew Jordan <mjordan@digium.com>
* channels/chan_sip.c: Clean-up of minor formatting issues in
r354542/3/4 rmudgett pointed out some formatting issues in the
check-in for ASTERISK-19290. This cleans those up. Review:
https://reviewboards.asterisk.org/r/1722/
2012-02-09 17:32 +0000 [r354640-354594] Mark Michelson <mmichelson@digium.com>
* main/translate.c: Fix translation path choices. This change makes
it so computational cost is not taken into account when deciding
if a multistep path is better than a single-step path. This means
that the only time a multistep path will be chosen is if no
single-step path exists. This ensures a better quality
translation even if it turns out to be slightly slower. (closes
issue ASTERISK-16821) reported by Andrew Lindh Review:
https://reviewboard.asterisk.org/r/1715
* main/translate.c: Remove outdated comment.
2012-02-09 19:52 +0000 [r354702-354655] Kinsey Moore <kmoore@digium.com>
* main/config.c: Make the config parser remove escaping backslashes
The config parser in Asterisk does not currently remove a
backslash that is used to escape a semicolon which would
otherwise be interpreted as the start of a comment. The change
here causes that backslash to be removed, but does not create a
real escape system in the config parser. The biggest complication
with a real escape system would be breaking existing configs
everywhere (parsing \\ as \ and breaking on escaped non-semicolon
characters) even though it would be the "right" way to do things.
(closes issue ASTERISK-17121) Review:
https://reviewboard.asterisk.org/r/1724/
* channels/chan_sip.c: Fix parsing of SIP headers where compact and
non-compact headers are mixed Change parsing of SIP headers so
that compactness of the header no longer influences which header
will be chosen. Previously, a non-compact header would be chosen
instead of a preceeding compact-form header. (closes issue
ASTERISK-17192) Review: https://reviewboard.asterisk.org/r/1728/
2012-02-09 22:01 +0000 [r354749] Terry Wilson <twilson@digium.com>
* funcs/func_cdr.c: Note that CDRs are immutable once a bridge is
torn down CDRs cannot be modified after a bridge is torn down,
(e.g. after Dial() returns) even though the CDR() function may be
called. Since modifying the CDR code to change this behavior
could very easily break all kinds of things, this patch just
documents this limitation. (closes issues ASTERISK-16923) Review:
https://reviewboard.asterisk.org/r/1720/
2012-02-10 18:03 +0000 [r354835] Richard Mudgett <rmudgett@digium.com>
* main/manager.c: Fix AMI Redirect ExtraChannel not redirecting to
the same exten and context. The astman_get_header() never returns
NULL so the check by the code for NULL would never fail. (closes
issue ASTERISK-16974) Reported by: Nuno Borges Patches:
0018325.patch (license #6116) patch uploaded by Nuno Borges
(modified)
2012-02-10 21:45 +0000 [r354889] Jason Parker <jparker@digium.com>
* apps/app_voicemail.c: Fix a voicemail memory leak with
heard/deleted messages. open_mailbox() was changed quite a long
time ago to allocate this memory. close_mailbox() should have
been changed to be responsible for freeing it.
2012-02-13 17:22 +0000 [r354953] Richard Mudgett <rmudgett@digium.com>
* res/res_config_pgsql.c, configs/extconfig.conf.sample: Fix
reconnecting to pgsql database after connection loss. There can
only be one database connection in res_config_pgsql just like
res_config_sqlite. If the connection is lost, the connection may
not get reestablished to the same database if the res_pgsql.conf
and extconfig.conf files are inconsistent. * Made only use the
configured database from res_pgsql.conf. * Fixed potential buffer
overwrite of last[] in config_pgsql(). (closes issue
ASTERISK-16982) Reported by: german aracil boned Review:
https://reviewboard.asterisk.org/r/1731/
2012-02-13 19:49 +0000 [r355009] Joshua Colp <jcolp@digium.com>
* pbx/pbx_config.c: Only allow one 'dialplan reload' to execute at
a time as otherwise they would share the same common local
context list. (closes issue AST-758)
2012-02-13 22:02 +0000 [r355056] Richard Mudgett <rmudgett@digium.com>
* pbx/pbx_spool.c: Fix occasional incorrectly delayed call-file
execution. Since the dir timestamp is available at one second
resolution, we cannot know if it was updated within the same
second after we scanned it. Therefore, we will force another scan
if the dir was just modified. * Changed to force another scan if
the directory was just modified. (closes issue ASTERISK-19081)
Reported by: Knut Bakke Review:
https://reviewboard.asterisk.org/r/1688/
2012-02-14 09:41 +0000 [r355136] Alexandr Anikin <may@telecom-service.ru>
* addons/chan_ooh323.c: call manager_event only if there is not
null channel structure (Closes issue ASTERISK-19298) Reported by:
robinfood Patches: issue19298.patch uploaded by may213 (License
#5415)
2012-02-14 13:33 +0000 [r355182] Sean Bright <sean@malleable.com>
* channels/chan_iax2.c: Clear the high order bit from the
destination call number before sending. send_apathetic_reply
takes the incoming frame's source call number as the destination
call number for the outgoing frame. If the incoming frame was a
full frame, then the high order bit of the source call number is
set and will be interpreted as a retransmit when sent back out as
the destination call number.
2012-02-14 15:50 +0000 [r355228] Jason Parker <jparker@digium.com>
* configs/cdr_sqlite3_custom.conf.sample: Don't enable sqlite3 CDRs
by default in sample configs.
2012-02-14 16:26 +0000 [r355268] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Properly invert the return of a strncmp
call. This was causing identification that should have been made
private to be public. (closes issue AST-814) reported by Patrick
Anderson Patches: chan_sip.c.diff uploaded by Patrick Anderson
(license 5430)
2012-02-14 18:12 +0000 [r355365-355319] Richard Mudgett <rmudgett@digium.com>
* cel/cel_sqlite3_custom.c: Fix lock typo that should be unlock in
cel_sqlite_custom reload. (closes issue ASTERISK-19356) Reported
by: Alex Villacis Lasso Patches:
asterisk-1.8.9.2-cel_sqlite3_custom-fix-reload-locking-typo.patch
(license #5617) patch uploaded by Alex Villacis Lasso Review:
https://reviewboard.asterisk.org/r/1740/
* configure, include/asterisk/autoconfig.h.in, configure.ac,
formats/format_ogg_vorbis.c: Fix voicemail problems when using
ogg/vorbis. Ogg/vorbis was fairly useless as a voicemail file
format because it did not implement the seek and tell format
callbacks among other problems. Since we were already using the
libvorbis and libvorbisenc libraries we can use libvorbisfile as
it is also part of the vorbis library package. * Made use the
libvorbisfile to handle the ogg/vorbis file stream. The
format_ogg_vorbis.c is now mostly a wrapper around libvorbisfile.
(closes issue ASTERISK-16926) Reported by: sque Patches:
ogg_vorbis_use_libvorbisfile.patch (license #6108) patch uploaded
by sque
2012-02-15 17:24 +0000 [r355529-355448] Sean Bright <sean@malleable.com>
* channels/chan_iax2.c: Use TRUNK_CALL_START as originally
intended. Back in r646, TRUNK_CALL_START was added and defined as
0x4000. That same value was also hard-coded in one part of the
IAX2 code instead of using the #define. TRUNK_CALL_START has
changed over the years (for dealing with LOW_MEMORY), but the
hard-coded usage was never updated to match. This patch fixes
that.
* channels/chan_iax2.c: Only use maxtrunkcall and maxnontrunkcall
in chan_iax2 if IAX_OLD_FIND is specified. These variables are
only accessed from the IAX_OLD_FIND path, so there is no reason
to keep them updated otherwise.
* channels/chan_iax2.c: When IAX2 debugging is enabled, make sure
to log 'apathetic' messages too.
2012-02-16 18:26 +0000 [r355608-355574] Richard Mudgett <rmudgett@digium.com>
* res/res_monitor.c: Fix AMI Monitor action without File header
converting channel name into filename. * Fix potential Solaris
crash if Monitor application has a urlbase and no fname_base
option.
* configure, include/asterisk/autoconfig.h.in,
autoconf/ast_c_declare_check.m4 (added), configure.ac,
formats/format_ogg_vorbis.c: Fix compile problem when old version
of libvorbisfile v1.1.2 is used. The principle difference between
libvorbisfile v1.1.2 and newer (at least v1.2.0) is the addition
of the predefined callbacks OV_CALLBACKS_xxx in
vorbis/vorbisfile.h used for ov_open_callbacks(). * Updated the
configure script to detect if libvorbisfile.h declares
OV_CALLBACKS_NOCLOSE. * Copied the declaration of
OV_CALLBACKS_NOCLOSE from v1.2.0 to allow v1.1.2 to compile.
(closes issue ASTERISK-19370) Reported by: Jonn Taylor
2012-02-16 20:01 +0000 [r355622] Sean Bright <sean@malleable.com>
* main/audiohook.c: Revert a change to audio_audiohook_write_list
that had no affect. When I made this change initially, I was
under the false impression that the audiohooks structure remained
on the channel after all of the hooks had been detached. This is
not the case, ast ast_read takes care of removing the audiohooks
structure if the lists are empty.
2012-02-16 23:53 +0000 [r355711-355700] Paul Belanger <pabelanger@digium.com>
* addons/ooh323cDriver.c, addons/ooh323c/src/ooSocket.c,
addons/ooh323c/src/ooq931.c, addons/ooh323c/src/ooTimer.c,
addons/ooh323c/src/ooCapability.c, addons/ooh323c/src/perutil.c:
Fix -Werror=unused-but-set-variable compiler error (gcc 4.6.2)
* addons/ooh323c/src/ooSocket.c: Missed a variable
* addons/ooh323cDriver.c, addons/ooh323c/src/ooSocket.c,
addons/ooh323c/src/ooq931.c, addons/ooh323c/src/ooTimer.c,
addons/ooh323c/src/ooCapability.c, addons/ooh323c/src/perutil.c:
Revert 355700 and 355701
2012-02-17 16:04 +0000 [r355732-355721] Mark Michelson <mmichelson@digium.com>
* main/translate.c: Revert change to translate.c as it has caused
an infinite loop to occur in circumstances.
* channels/chan_sip.c: Fix regressions with regards to route-set
creation on early dialogs. This fixes two main issues: 1.
Asterisk would send a CANCEL to the route created by the
provisional response instead of using the same destination it did
in the initial INVITE. 2. If a new route set arrives in a 200 OK
than was in the 1XX response (perfectly possible if our outbound
INVITE gets forked), then the route set in the 200 OK needs to
overwrite the route set in the 1XX response. (closes issue
ASTERISK-19358) Reported by: Karsten Wemheuer Tested by: Karsten
Wemheuer patches: ASTERISK-19358.patch uploaded by Mark Michelson
(license 5049) ASTERISK-19358.patch uploaded by Stefan Schmidt
(license 6034) Review: https://reviewboard.asterisk.org/r/1749
2012-02-17 19:32 +0000 [r355793-355746] Sean Bright <sean@malleable.com>
* channels/chan_iax2.c: Pass the correct value to
ast_timer_set_rate() for IAX2 trunking. IAX2 uses the trunkfreq
variable to determine how often to send trunk packets, but this
value is in milliseconds while ast_timer_set_rate() expects the
rate argument to be ticks per second. So we divide 1000 by
trunkfreq and pass that in instead. With a default of 20ms, this
change makes IAX2 send trunk packets every 20ms instead of every
50ms. Tracked down by myself and Bob Wienholt.
* channels/chan_iax2.c, configs/iax.conf.sample: Don't allow
trunkfreq to be greater than 1000ms.
2012-02-18 03:59 +0000 [r355839] Paul Belanger <pabelanger@digium.com>
* res/res_pktccops.c: Fix -Werror=unused-but-set-variable compiler
error (gcc 4.6.2)
2012-02-18 07:55 +0000 [r355850] Alec L Davis <sivad.a@paradise.net.nz>
* channels/sig_pri.c, channels/sig_ss7.c, channels/sig_pri.h,
channels/chan_dahdi.c, channels/sig_analog.c, channels/sig_ss7.h,
channels/sig_analog.h: push 'outgoing' flag from sig_XXX up to
chan_dahdi 'p->outgoing' in chan_dahdi and sig_analog wern't kept
in sync, particulary FXS ast_hangup didn't clear the 'outgoing'
flag. sig_pri and sig_ss7 were keeping 'outgoing' flag insync.
Now provides a callback for all the low level sig_XXX modules.
(issue ASTERISK-19316) alecdavis (license 585) Reported by:
Jeremy Pepper Tested by: alecdavis Review:
https://reviewboard.asterisk.org/r/1747/
2012-02-19 17:49 +0000 [r356107-355901] Sean Bright <sean@malleable.com>
* channels/chan_iax2.c: Set the length of the ast_sockaddr, so that
we can set it's port later. Without this, the call to
ast_sockaddr_set_port a few lines later is a noop.
* channels/chan_iax2.c: Add some boilerplate documentation for
IAXVAR and IAXPEER.
* channels/chan_dahdi.c: Change some debug messages from LOG_DEBUG
to ast_debug.
* channels/chan_dahdi.c: This was a LOG_NOTICE, so roll it back.
* channels/chan_iax2.c: Remove spurious warning when
'qualifyfreqnotok' is set successfully. (closes issue
ASTERISK-17176) Reported by: John Covert Tested by: Sean Bright
Patches: chan_iax2.c.qualifyfreqnotok.patch uploaded by John
Covert (license 5512)
* channels/chan_iax2.c: Make 'iax2 show callnumber usage' output
make sense when an IP is passed in.
2012-02-22 14:50 +0000 [r356214] Matthew Jordan <mjordan@digium.com>
* channels/chan_sip.c: Fix potential buffer overrun and memory leak
when executing "sip show peers" The "sip show peers" command uses
a fix sized array to sort the current peers in the peers
ao2_container. The size of the array is based on the current
number of peers in the container. However, once the size of the
array is determined, the number of peers in the container can
change, as the peers container is not locked. This could cause a
buffer overrun when populating the array, if peers were added to
the container after the array was created. Additionally, a memory
leak of the allocated array would occur if a user caused the
_show_peers method to return CLI_SHOWUSAGE. We now create a
snapshot of the current peers using an ao2_callback with the
OBJ_MULTIPLE flag. This size of the array is set to the number of
peers that the iterator will iterate over; hence, if peers are
added or removed from the peers container it will not affect the
execution of the "sip show peers" command. Review:
https://reviewboard.asterisk.org/r/1738/ (closes issue
ASTERISK-19231) (closes issue ASTERISK-19361) Reported by: Thomas
Arimont, Jamuel Starkey Tested by: Thomas Arimont, Jamuel Starkey
Patches: sip_show_peers_2012_02_16.diff uploaded by mjordan
(license 6283)
2012-02-22 20:20 +0000 [r356290] Paul Belanger <pabelanger@digium.com>
* apps/app_rpt.c: Fix -Werror=unused-but-set-variable compiler
error (gcc 4.6.2) Review:
https://reviewboard.asterisk.org/r/1763/
2012-02-22 21:08 +0000 [r356291] Terry Wilson <twilson@digium.com>
* include/asterisk/calendar.h, main/loader.c, res/res_calendar.c:
Track module use count for res_calendar If the res_calendar
module was followed immediately by one of the calendar tech
modules and "core stop gracefully" was run, Asterisk would crash.
This patch adds use count tracking for res_calendar so that it is
unloaded after the tech modules when shutting down gracefully. It
is now not possible to unload all the of the calendar modules via
"module unload res_calednar.so", but it is still possible to
unload them all via "module unload -h res_calendar.so". Review:
https://reviewboard.asterisk.org/r/1752/
2012-02-22 21:29 +0000 [r356430-356335] Paul Belanger <pabelanger@digium.com>
* apps/app_rpt.c: Add back strsep() function for previous commit
* apps/app_rpt.c: Missed one strsep() function
* addons/chan_ooh323.c: Fix -Werror=unused-but-set-variable
compiler error (gcc 4.6.2)
2012-02-23 15:37 +0000 [r356475] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Fix ACK routing for non-2xx responses. When
we send an ACK for a 2xx response to an INVITE, we are supposed
to use the learned route set. However, when we receive a non-2xx
final response to an INVITE, we are supposed to send the ACK to
the same place we initially sent the INVITE. We had been doing
this up until the changes went in that would build a route set
from provisional responses. That introduced a regression where we
would use the learned route set under all circumstances. With
this change, we now will set the destination of our ACK based on
the invitestate. If it is INV_COMPLETED then that means that we
have received a non-2xx final response (INV_TERMINATED indicates
a 2xx response was received). If it is INV_CANCELLED, then that
means the call is being canceled, which means that we should be
ACKing a 487 response. The other change introduced here is
setting the invitestate to INV_CONFIRMED when we send an ACK
*after* the reqprep instead of before. This way, we can tell in
reqprep more easily what the invitestate is prior to sending the
ACK. (closes issue ASTERISK-19389) reported by Karsten Wemheuer
patches: ASTERISK-19389v2.patch uploaded by Mark Michelson
(license #5049) (with some slight modifications prior to commit)
2012-02-23 19:49 +0000 [r356521] Richard Mudgett <rmudgett@digium.com>
* channels/chan_sip.c, main/features.c: Fix blind transfer parking
issues if the dialed extension is not recognized as a parking
extension. Custom parking extensions may not be coded such that
the first and only extension priority is the Park application.
These custom parking extensions will not be recognized as parking
extensions. When a call is blind transferred to an extension that
is not recognized as a parking extension, the normal blind
transfer code causes the transferred channel to start executing
dialplan. Calls that get parked in this manner do not know the
original channel name that parked the call so the original parker
could never be called back if the parked call is not retrieved
before the timeout time. The parking space is also announced to
the call being parked as a side effect of not knowing the
original parking channel. * Fix handling of BLINDTRANSFER channel
variable for call parking. * Fixed SIP blind transfer using the
wrong dialplan context variable to check for the parking
extension. (closes issue ASTERISK-19322) Reported by: aragon
Tested by: rmudgett, jparker Review:
https://reviewboard.asterisk.org/r/1730/ JIRA AST-766
2012-02-24 15:07 +0000 [r356650-356604] Matthew Jordan <mjordan@digium.com>
* include/asterisk/rtp_engine.h, res/res_srtp.c,
channels/sip/sdp_crypto.c, include/asterisk/res_srtp.h,
main/rtp_engine.c: Allow SRTP policies to be reloaded Currently,
when using res_srtp, once the SRTP policy has been added to the
current session the policy is locked into place. Any attempt to
replace an existing policy, which would be needed if the remote
endpoint negotiated a new cryptographic key, is instead rejected
in res_srtp. This happens in particular in transfer scenarios,
where the endpoint that Asterisk is communicating with changes
but uses the same RTP session. This patch modifies res_srtp to
allow remote and local policies to be reloaded in the underlying
SRTP library. From the perspective of users of the SRTP API, the
only change is that the adding of remote and local policies are
now added in a single method call, whereas they previously were
added separately. This was changed to account for the differences
in handling remote and local policies in libsrtp. Review:
https://reviewboard.asterisk.org/r/1741/ (closes issue
ASTERISK-19253) Reported by: Thomas Arimont Tested by: Thomas
Arimont Patches: srtp_renew_keys_2012_02_22.diff uploaded by Matt
Jordan (license 6283) (with some small modifications for this
check-in)
* res/res_srtp.c: Remove srtp_shutdown from res_srtp The patch for
ASTERISK-19253 included properly shutting down the libsrtp
library in the case of module unload. Unfortunately, not all
distributions have the srtp_shutdown call. As such, this patch
removes calling srtp_shutdown.
2012-02-24 18:23 +0000 [r356677] Richard Mudgett <rmudgett@digium.com>
* include/asterisk/tcptls.h, channels/sip/include/sip.h,
channels/chan_sip.c: Fix worker thread resource leak in SIP
TCP/TLS. The SIP TCP/TLS worker threads were created joinable but
noone could join them if they died on their own. * Fix the SIP
TCP/TLS worker threads to not be created joinable. *
_sip_tcp_helper_thread() only needs one parameter since the pvt
parameter is only passed in as NULL and never used. (closes issue
ASTERISK-19203) Reported by: Steve Davies Review:
https://reviewboard.asterisk.org/r/1714/
2012-02-25 17:21 +0000 [r356797] Matthew Jordan <mjordan@digium.com>
* apps/app_voicemail.c: Fix crash in app_voicemail during
close_mailbox In r354890, a memory leak in app_voicemail was
fixed by properly disposing of the allocated heard/deleted
pointers. However, there are situations, particularly when no
messages are found in a folder, where these pointers are not
allocated and not NULL. In that case, an invalid free would be
attempted, which could crash app_voicemail. As there are a number
of code paths where this could occur, this patch uses the number
of messages detected in the folder before it attempts to free the
pointers. This resolves the crash detected in the Asterisk Test
Suite's check_voicemail_nominal test.
2012-02-27 15:14 +0000 [r356917] Jonathan Rose <jrose@digium.com>
* res/res_odbc.c: Remove possible segfaults from res_odbc by adding
locks around usage of odbc handle (closes issue ASTERISK-19011)
Reported by: Walter Doekes Patches:
issueA19011_combine_read_and_write_locks_WORK_IN_PROGRESS.patch
uploaded by Walter Doekes (license 5674) review:
https://reviewboard.asterisk.org/r/1719/ review:
https://reviewboard.asterisk.org/r/1622/
2012-02-27 16:03 +0000 [r356963] Terry Wilson <twilson@digium.com>
* main/features.c: Copy CDR variables when set during a bridge This
patch makes sure amaflags, accountcode, and userfield get copied
to the bridge CDR when set during a bridge (like via a custom
feature). (closes issue ASTERISK-16990) Review:
https://reviewboard.asterisk.org/r/1721/
2012-02-27 23:34 +0000 [r357093] Richard Mudgett <rmudgett@digium.com>
* main/channel.c: Fix callerid of Originated calls. Thanks to Matt
Riddell for tracking this down. (closes issue ASTERISK-19385)
Reported by: ornix
2012-03-06 Asterisk Development Team <asteriskteam@digium.com>
* Asterisk 1.8.11.0-rc2 Released.
2012-03-05 Asterisk Development Team <asteriskteam@digium.com>
* Asterisk 1.8.10.0 Released.
2012-03-01 Asterisk Development Team <asteriskteam@digium.com>
* Asterisk 1.8.10.0-rc4 Released.
* main/acl.c: Prevent outbound SIP NOTIFY packets from displaying
a port of 0.
In the change from 1.6.2 to 1.8, ast_sockaddr was introduced which
changed the behavior of ast_find_ourip such that port number was
wiped out. This caused the port in internip (which is used for
Contact and Call-ID on NOTIFYs) to be 0. This change causes
ast_find_ourip to be port-preserving again.
2012-02-28 Asterisk Development Team <asteriskteam@digium.com>
* Asterisk 1.8.10.0-rc3 Released.
* main/channel.c: Fix callerid of Originated calls.
The callerid of originated calls (independent of mechanism) was not
being passed to the outbound channel. This patch fixes that. Thanks
to Matt Riddell for tracking this down.
(closes issue ASTERISK-19385)
Reported by: ornix
* channels/chan_sip.c: Fix ACK routing for non-2xx responses.
When we send an ACK for a 2xx response to an INVITE, we are supposed
to use the learned route set. However, when we receive a non-2xx
final response to an INVITE, we are supposed to send the ACK to the
same place we initially sent the INVITE.
We had been doing this up until the changes went in that would build
a route set from provisional responses. That introduced a regression
where we would use the learned route set under all circumstances.
With this change, we now will set the destination of our ACK based on
the invitestate. If it is INV_COMPLETED then that means that we have
received a non-2xx final response (INV_TERMINATED indicates a 2xx
response was received). If it is INV_CANCELLED, then that means the
call is being canceled, which means that we should be ACKing a 487
response.
The other change introduced here is setting the invitestate to
INV_CONFIRMED when we send an ACK *after* the reqprep instead of
before. This way, we can tell in reqprep more easily what the
invitestate is prior to sending the ACK.
(closes issue ASTERISK-19389)
reported by Karsten Wemheuer
patches:
ASTERISK-19389v2.patch uploaded by Mark Michelson (license #5049)
* channels/chan_sip.c: Fix regressions with regards to route-set
creation on early dialogs.
This fixes two main issues:
1. Asterisk would send a CANCEL to the route created by the provisional
response instead of using the same destination it did in the initial
INVITE.
2. If a new route set arrives in a 200 OK than was in the 1XX response
(perfectly possible if our outbound INVITE gets forked), then the
route set in the 200 OK needs to overwrite the route set in the 1XX
response.
(closes issue ASTERISK-19358)
Reported by: Karsten Wemheuer
Tested by: Karsten Wemheuer
patches:
ASTERISK-19358.patch uploaded by Mark Michelson (license 5049)
ASTERISK-19358.patch uploaded by Stefan Schmidt (license 6034)
Review: https://reviewboard.asterisk.org/r/1749
2012-02-10 Asterisk Development Team <asteriskteam@digium.com>
* Asterisk 1.8.10.0-rc2 Released.
* channels/chan_sip.c: Fix SIP INFO DTMF handling for non-numeric
codes. In ASTERISK-18924, SIP INFO DTMF handling was changed to
account for both lowercase alphatbetic DTMF events, as well as
uppercase alphabetic DTMF events. When this occurred, the comparison
of the character buffer containing the event code was changed such
that the buffer was first compared against '0' and '9' to determine if
it was numeric. Unfortunately, since the first character in the
buffer will typically be '1' in the case of non-numeric event codes
(10-16), this caused those codes to be converted to a DTMF event of
'1'. This patch fixes that, and cleans up handling of both
application/dtmf-relay and application/dtmf content types.
Review: https://reviewboard.asterisk.org/r/1722/
(closes issue ASTERISK-19290) Reported by: Ira Emus
Tested by: mjordan
* apps/app_parkandannounce.c: Fix crash in ParkAndAnnounce from
uninitialized caller_id storage (closes issue ASTERISK-19311)
Reported by: tootai
Tested by: rmudgett
* channels/chan_agent.c: Fixes deadlocks occuring in chan_agent due to
r335976. Bad locking order was added to chan_agent to prevent
segfaults from having no locking in a patch by irroot. This patch
addresses the bad locking order by releasing locks before getting the
right locking order to stop deadlocks from occuring when doing
multiple interactions with agents. (closes issue ASTERISK-19285)
Reported by: Alex Villacis Lasso
Review: https://reviewboard.asterisk.org/r/1708/
* channels/chan_sip.c: Ensure entering T.38 passthrough does not cause
an infinite loop. After R340970 Asterisk was still polling the RTCP
file descriptor after RTCP is shut down and removed. If the
descriptor happened to have data ready when the removal occured then
Asterisk would go into an infinite loop trying to read data that it
can never actually access. This change disables the audio RTCP file
descriptor for the duration of the T.38 transaction. (closes issue
ASTERISK-18951) Reported-by: Kristijan Vrban
* channels/chan_sip.c,include/asterisk/dnsmgr.h,main/dnsmgr.c: Re-link
peers by IP when dnsmgr changes the IP Asterisk's dnsmgr currently
takes a pointer to an ast_sockaddr and updates it anytime an address
resolves to something different. There are a couple of issues with
this. First, the ast_sockaddr is usually the address of an ast_sockaddr
inside a refcounted struct and we never bump the refcount of those
structs when using dnsmgr. This makes it possible that a refresh could
happen after the destructor for that object is called (despite
ast_dnsmgr_release being called in that destructor). Second, the
module using dnsmgr cannot be aware of an address changing without
polling for it in the code. If an action needs to be taken on address
update (like re-linking a SIP peer in the peers_by_ip table), then
polling for this change negates many of the benefits of having dnsmgr
in the first place.
2012-02-01 Asterisk Development Team <asteriskteam@digium.com>
* Asterisk 1.8.10.0-rc1 Released.
* Test results:
http://bamboo.asterisk.org/browse/TESTING-ASTERISK18100RCS-2
2012-01-30 12:42 +0000 [r353260] Kevin P. Fleming <kpfleming@digium.com>
* channels/chan_sip.c: Clarify log WARNING message when port-zero
SDP 'm' lines received. Previously, if an m-line in an SDP offer
or answer had a port number of zero, that line was skipped, and
resulted in an 'Unsupported SDP media type...' warning message.
This was misleading, as the media type was not unsupported, but
was ignored because the m-line indicated that the media stream
had been rejected (in an answer) or was not going to be used (in
an offer).
2012-01-29 02:42 +0000 [r353175] Russell Bryant <russell@russellbryant.com>
* main/netsock.c: Find even more network interfaces. The previous
change made the code look for emN and pciN in addition to what it
did originally, which was search for ethN. However, it needed to
be looking for pciN#N, so that's what it does now. This also
moves the memset() to be before every ioctl().
2012-01-28 14:49 +0000 [r353126] Kevin P. Fleming <kpfleming@digium.com>
* main/rtp_engine.c: Add 'L16-256' MIME subtype alias for slin16.
Asterisk has supported the 'L16' MIME subtype for 16kHz signed
linear (PCM) audio for quite some time, but some endpoints refer
to it as 'L16-256'. This commit adds this as an alias for the
existing format.
2012-01-28 04:25 +0000 [r353077] Russell Bryant <russell@russellbryant.com>
* main/netsock.c: Update ast_set_default_eid() to find more network
interfaces. As of Fedora 15, ethN is not the name of ethernet
interfaces. The names are emN or pciN. Update some code that
searched for interfaces named ethN to look for the new names, as
well. For more information about why this change was made, see
this page: http://domsch.com/blog/?p=455
2012-01-27 19:12 +0000 [r352959] Jonathan Rose <jrose@digium.com>
* res/res_monitor.c: Make failed PauseMonitor and UnpauseMonitor
with no valid channel not close AMI session. I also went ahead
and took a little time to make sure that the manager value
AMI_SUCCESS was used instead of just return 0 being thrown around
everywhere since that's how we handle this stuff these days.
(closes issue ASTERISK-19249) Reporter: Jamuel Starkey Patches:
res_monitor.c-ASTERISK-19249.diff uploaded by Jamuel Starkey
(license 5766)
2012-01-27 18:22 +0000 [r352955] Richard Mudgett <rmudgett@digium.com>
* res/snmp/agent.c, main/taskprocessor.c, apps/app_queue.c,
channels/chan_iax2.c, apps/app_chanspy.c, main/indications.c,
res/res_odbc.c, res/res_srtp.c, main/pbx.c, channels/chan_sip.c,
include/asterisk/indications.h: Audit of ao2_iterator_init()
usage for v1.8. Fixes numerous reference leaks and missing
ao2_iterator_destroy() calls as a result. Review:
https://reviewboard.asterisk.org/r/1697/
2012-01-27 00:05 +0000 [r352862] Alec L Davis <sivad.a@paradise.net.nz>
* channels/sip/include/sip.h, channels/chan_sip.c: rfc4235 -
Section 4.1: Versions MUST be representable using a non-negative
32 bit integer. If a BLF subscription exists for long enough,
using %d may print negative version numbers. Unlikely, as 2^32 at
1 update per second is ~137 years, or half that before the
versions number started going negative. Tested with Asterisk
1.8.8.2 with Grandstream phones. alecdavis (license 585) Tested
by: alecdavis Review: https://reviewboard.asterisk.org/r/1694/
2012-01-26 20:14 +0000 [r352807] Alexandr Anikin <may@telecom-service.ru>
* addons/chan_ooh323.c: Fix outbound DTMF for inband mode (tell
asterisk core to generate DTMF sounds). (Closes issue
ASTERISK-19233) Reported by: Matt Behrens Patches:
chan_ooh323.c.patch uploaded by Matt Behrens (License #6346)
2012-01-26 19:06 +0000 [r352755] Jonathan Rose <jrose@digium.com>
* channels/chan_sip.c: Copy amaflags to sip_pvt from peer during
create_addr_from_peer For whatever reason, we don't have a single
function for copying data like this from SIP peers to the SIP
pvt. This patch adds the copying of amaflags to the sip_pvt, but
it would probably be worth discussing this function along with
the others that essentially just copy some amount of data from a
peer to a private. (Closes issue ASTERISK-19029) Reported by:
Matt Lehner
2012-01-26 06:27 +0000 [r352704] Alec L Davis <sivad.a@paradise.net.nz>
* channels/chan_sip.c: Cleanup dialog-info+xml Notify dialog Make
similar to other Notify messages. sample output: <?xml
version="1.0"?> <dialog-info
xmlns="urn:ietf:params:xml:ns:dialog-info" version="715"
state="full" entity="sip:8523@192.168.x.xx"> <dialog id="8523">
<state>terminated</state> </dialog> </dialog-info> Tested with
Asterisk 1.8.8.2 with Grandstream phones. alecdavis (license 585)
Tested by: alecdavis Review:
https://reviewboard.asterisk.org/r/1693/
2012-01-25 22:21 +0000 [r352643] Paul Belanger <pabelanger@digium.com>
* apps/app_voicemail.c: Fix -Werror=unused-but-set-variable
compiler error (gcc 4.6.2)
2012-01-25 21:16 +0000 [r352612] Kevin P. Fleming <kpfleming@digium.com>
* main/test.c: Avoid unnecessary rebuilds of main/test.c.
main/test.c includes "asterisk/version.h", when it should include
"asterisk/ast_version.h" instead (and it should use the
ast_get_version() and ast_get_version_num() functions). This
commit modifies it to extract the Asterisk version information
using the proper APIs, and as a result means that main/test.c no
longer needs to be rebuilt when a Subversion checkout is updated
or modified.
2012-01-25 17:28 +0000 [r352514-352551] Terry Wilson <twilson@digium.com>
* channels/chan_sip.c: Remove some extraneous debugging from
registry memleak fix
* channels/chan_sip.c: Clean up some SIP registry-related memory
leaks 1) Be sure and free at unload the epa_backend we allocate
at startup 2) Do the same sip_registry cleanup at unload we do at
reload Review: https://reviewboard.asterisk.org/r/1689/
2012-01-25 16:39 +0000 [r352511] Jonathan Rose <jrose@digium.com>
* configs/sip.conf.sample: Redocuments sip types peer, user, friend
in sip.conf.sample There was faulty information in the sample
config describing user as a synonym for friend so it has been
changed to better elaborate on the differences between the three
entity types. (closes issue ASTERISK-15537) Reported by: yarique
2012-01-24 22:17 +0000 [r352424] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Don't do a DNS lookup on an outbound
REGISTER host if there is an outbound proxy configured. (closes
issue ASTERISK-16550) reported by: Olle Johansson
2012-01-24 20:33 +0000 [r352367] Jonathan Rose <jrose@digium.com>
* sounds/Makefile: Set core sounds version to 1.4.22. Now that we
have the right license for the Russian 1.4.22 sounds as well as
the sounds for the Australian English 1.4.22 sounds, we can
finally set the sounds to use 1.4.22! (closes issue
ASTERISK-18978) Reported by: Cameron Twomey Patches:
confbridge.tar.001 uploaded by Cameron Twomey confbridge.tar.002
uploaded by Cameron Twomey
2012-01-24 16:59 +0000 [r352291] Richard Mudgett <rmudgett@digium.com>
* funcs/func_odbc.c: Fix locking issues with channel datastores in
func_odbc.c. * Fixed a potential memory leak when an existing
datastore is manually destroyed by inline code instead of calling
ast_datastore_free(). (closes issue ASTERISK-17948) Reported by:
Archie Cobbs Review: https://reviewboard.asterisk.org/r/1687/
2012-01-24 16:30 +0000 [r352287] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: Move RTP timeout check to before bridged
channel check so it is actually executed. (issue ASTERISK-19179)
Reported by: TSAREGORODTSEV Yury (closes issue ASTERISK-14534)
Reported by: kriborgen Patches: chan_sip.patch uploaded by
kriborgen (license 6138)
2012-01-23 20:30 +0000 [r352199-352230] Mark Michelson <mmichelson@digium.com>
* main/features.c: Fix grammar of comment.
* main/features.c: Fix blind transfers from failing if an 'h'
extension is present. This prevents the 'h' extension from being
run on the transferee channel when it is transferred via a native
transfer mechanism such as SIP REFER. (closes ASTERISK-19173)
Reported by: Ross Beer Tested by: Kristjan Vrban Patches:
ASTERISK-19173 by Mark Michelson (license 5049) Review:
https://reviewboard.asterisk.org/r/1685
2012-01-23 19:12 +0000 [r352144] Matthew Jordan <mjordan@digium.com>
* res/res_fax_spandsp.c: Correctly apply FAXOPT settings (V17, V27,
V29) before starting spandsp layer While the FAXOPT function
could be used to set the modem capabilities, the input to that
function was not being applied correctly to the spandsp layer.
This patch applies the current model capabilities before starting
the spandsp layer. (closes issue: ASTERISK-16409) Reported by:
Kristijan Vrban Tested by: Matt Jordan, Matthew Nicholson
Patches: spandsp-modems-1.8.diff uploaded by mnicholson (license
5081) spandsp-modems-10.diff uploaded by mnicholson (license
5081)
2012-01-23 17:33 +0000 [r352090] Richard Mudgett <rmudgett@digium.com>
* channels/chan_sip.c: Fix sip_cfg.notifycid to be set with the
defined enum values. The invalid value used when notifycid was
enabled was benign. As far as the code was concerned -1 and 1 are
equivalent. (closes issue ASTERISK-19232) Reported by: Eike
Kuiper
2012-01-21 00:20 +0000 [r352029] Richard Mudgett <rmudgett@digium.com>
* main/app.c, funcs/func_timeout.c: Fix ast_app_dtget() time unit
inconsistency. Note: Noone calls ast_app_dtget() with the timeout
parameter of zero so the bad code normally will never get
executed. * Fix unnecessary floating point division in
func_timeout.c timeout_write() when all other values are
integers. (closes issue ASTERISK-16817) Reported by: Dmitry
Andrianov
2012-01-21 00:08 +0000 [r352014-352016] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Remove XXX comment that is not necessary.
* channels/chan_sip.c: Fix RTP reference leak. If a blind transfer
were initiated using a REFER without a prior reINVITE to place
the call on hold, AND if Asterisk were sending RTCP reports, then
there was a reference for the RTP instance of the transferer.
This fixes the issue by merging two similar but slightly
conflicting sections of code into a single area. It also adds a
stop_media_flows() call in the case that the transferer's UA
never sends a BYE to us like it is supposed to. (issue
ASTERISK-19192) Review: https://reviewboard.asterisk.org/r/1681/
2012-01-20 19:34 +0000 [r351858-351860] Kinsey Moore <kmoore@digium.com>
* codecs/ilbc/iLBC_test.c: More corrections for the ilbc code These
changes are in a file that is not compiled by default, and so
were missed on earlier checks.
* codecs/ilbc/LPCencode.c, codecs/ilbc/iLBC_decode.c: Allow ilbc
code to build under dev mode GCC 4.6.3 found some set/unused
variables in the ILBC code.
2012-01-20 16:01 +0000 [r351765] Jonathan Rose <jrose@digium.com>
* channels/chan_sip.c: Accidentally left off a semicolon only in
1.8 somehow for previous patch.
2012-01-20 15:48 +0000 [r351760] Matthew Jordan <mjordan@digium.com>
* codecs/ilbc/helpfun.c: Remove unused variable 'tmp' from helpfun
in ilbc codec gcc version 4.6.2 caught an unused variable in the
ilbc codec library. This would prevent compilation with
--enable-dev-mode; variable removed.
2012-01-20 15:42 +0000 [r351759] Jonathan Rose <jrose@digium.com>
* channels/chan_sip.c: Adds setting of mwi_from field to
check_auth_result check_peer_ok (closes ASTERISK-19057) Reported
By: Yuri Patches: 348360chan_sip.diff uploaded by Yuri (license
5242)
2012-01-20 12:59 +0000 [r351707] Stefan Schmidt <sst@sil.at>
* contrib/asterisk-ng-doxygen: enable doxygen build for files in
the channels/sip folder like reqresp_parser.c
2012-01-19 23:17 +0000 [r351618] Richard Mudgett <rmudgett@digium.com>
* channels/chan_sip.c, channels/sip/reqresp_parser.c: Misc minor
fixes in reqresp_parser.c and chan_sip.c. * Fix corner cases in
get_calleridname() parsing and ensure that the output buffer is
nul terminated. * Make get_calleridname() truncate the name it
parses if the given buffer is too small rather than abandoning
the parse and not returning anything for the name. Adjusted
get_calleridname_test() unit test to handle the truncation
change. * Fix get_in_brackets_test() unit test to check the
results of get_in_brackets() correctly. * Fix
parse_name_andor_addr() to not return the address of a local
buffer. This function is currently not used. * Fix potential NULL
pointer dereference in sip_sendtext(). * No need to
memset(calleridname) in check_user_full() or tmp_name in
get_name_and_number() because get_calleridname() ensures that it
is nul terminated. * Reply with an accurate response if
get_msg_text() fails in receive_message(). This is academic in
v1.8 because get_msg_text() can never fail.
2012-01-19 22:36 +0000 [r351611] Kinsey Moore <kmoore@digium.com>
* res/res_rtp_asterisk.c: Correct output of RTCP jitter statistics
in SR and RR reports Change the RTCP RR and SR generation code to
convert Asterisk's internal jitter statistics to be represented
in RTP timestamp units based on the rate of the codec in use
instead of in seconds. (closes issue ASTERISK-14530)
2012-01-19 21:46 +0000 [r351559] Jonathan Rose <jrose@digium.com>
* include/asterisk/netsock2.h, channels/chan_sip.c: Eliminates
doubling the :port part of SIP Notify Message-Account headers.
This patch prevents the domain string from getting mangled during
the initreqprep step by moving the initialization to before its
immediate use. It also documents this pitfall for the
ast_sockaddr_stringify functions. (issue ASTERISK-19057) Reported
by: Yuri Review: https://reviewboard.asterisk.org/r/1678/
2012-01-19 21:11 +0000 [r351504] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: Prevent crash when an SDP offer is received
with an encrypted video stream when support for video is disabled
and res_srtp is loaded. (closes issue ASTERISK-19202) Reported
by: Catalin Sanda
2012-01-18 20:54 +0000 [r351450] Matthew Jordan <mjordan@digium.com>
* codecs/ilbc/StateConstructW.c (added), codecs/ilbc/packing.c
(added), codecs/ilbc/StateConstructW.h (added),
codecs/ilbc/packing.h (added), codecs/ilbc/getCBvec.c (added),
codecs/ilbc/LPCdecode.c (added), codecs/ilbc/enhancer.c (added),
codecs/ilbc/lsf.c (added), codecs/ilbc/iLBC_encode.c (added),
codecs/ilbc/getCBvec.h (added), codecs/ilbc/LPCdecode.h (added),
codecs/ilbc/enhancer.h (added), codecs/ilbc/FrameClassify.c
(added), codecs/ilbc/iLBC_define.h (added), codecs/ilbc/lsf.h
(added), codecs/ilbc/extract-cfile.awk (added),
codecs/ilbc/iLBC_encode.h (added), codecs/ilbc/Makefile,
codecs/ilbc/FrameClassify.h (added), codecs/ilbc/helpfun.c
(added), codecs/ilbc/LICENSE_ADDENDUM (added),
codecs/ilbc/doCPLC.c (added), codecs/ilbc/anaFilter.c (added),
codecs/ilbc/helpfun.h (added), codecs/ilbc/createCB.c (added),
codecs/ilbc/doCPLC.h (added), codecs/ilbc/anaFilter.h (added),
codecs/ilbc/constants.c (added), codecs/ilbc/iLBC_decode.c
(added), codecs/ilbc/createCB.h (added), codecs/ilbc/constants.h
(added), codecs/ilbc/iLBC_decode.h (added),
codecs/ilbc/iCBSearch.c (added), codecs/ilbc/filter.c (added),
codecs/ilbc/hpInput.c (added), codecs/ilbc/gainquant.c (added),
codecs/ilbc/iCBSearch.h (added), codecs/ilbc/hpOutput.c (added),
codecs/ilbc/rfc3951.txt (added), codecs/ilbc/filter.h (added),
codecs/ilbc/gainquant.h (added), codecs/ilbc/LPCencode.c (added),
codecs/ilbc/hpInput.h (added), codecs/codec_ilbc.c,
codecs/ilbc/PATENTS (added), codecs/ilbc/StateSearchW.c (added),
codecs/ilbc/hpOutput.h (added),
contrib/scripts/get_ilbc_source.sh, codecs/ilbc/LICENSE (added),
codecs/ilbc/LPCencode.h (added), codecs/ilbc/StateSearchW.h
(added), codecs/ilbc/iCBConstruct.c (added),
codecs/ilbc/syntFilter.c (added), codecs/ilbc/iCBConstruct.h
(added), codecs/ilbc/iLBC_test.c (added),
codecs/ilbc/syntFilter.h (added): Include iLBC source code for
distribution with Asterisk This patch includes the iLBC source
code for distribution with Asterisk. Clarification regarding the
iLBC source code was provided by Google, and the appropriate
licenses have been included in the codecs/ilbc folder. Review:
https://reviewboard.asterisk.org/r/1675 Review:
https://reviewboard.asterisk.org/r/1649 (closes issue:
ASTERISK-18943) Reporter: Leif Madsen Tested by: Matt Jordan
2012-01-18 14:57 +0000 [r351396] Stefan Schmidt <sst@sil.at>
* channels/chan_sip.c: The get_pai function in chan_sip.c didn't
recognized a proper callerid name and number from a
P-Asserted-Identity cause the header parsing logic was wrong.
Changing the parsing functions to the sip header parsing APIs in
reqresp_parser.h solves this problem. Review:
https://reviewboard.asterisk.org/r/1673 Reviewed by: wdoekes2 and
Mark Michelson
2012-01-17 17:22 +0000 [r351306] Mark Michelson <mmichelson@digium.com>
* res/res_rtp_asterisk.c: Eliminate odd initialization of probation
variable.
2012-01-17 16:55 +0000 [r351287] Jonathan Rose <jrose@digium.com>
* CHANGES, res/res_rtp_asterisk.c, configs/rtp.conf.sample: Adds
pjmedia probation concepts to res_rtp_asterisk's learning mode.
In order to better handle RTP sources with strictrtp enabled
(which is now default in 10) using the learning mode to figure
out new sources when they change is handled by checking for a
number of consecutive (by sequence number) packets received to an
rtp struct based on a new configurable value called 'probation'.
Also, during learning mode instead of liberally accepting all
packets received, we now reject packets until a clear source has
been determined. Review: https://reviewboard.asterisk.org/r/1663/
2012-01-17 16:41 +0000 [r351284] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Use built-in parsing functions for Contact
and Record-Route headers. If a Contact or a Record-Route header
had a quoted string with an item in angle brackets, then we would
mis-parse it. For instance, "Bob <1234>" <1234@example.org> would
be misparsed as having the URI "1234" The fix for this is to use
parsing functions from reqresp_parser.h since they are heavily
tested and are awesome. (issue ASTERISK-18990)
2012-01-17 16:06 +0000 [r351233] Matthew Jordan <mjordan@digium.com>
* channels/chan_sip.c: Fix udptl issue with initial INVITE
introduced by r351027 When an inital INVITE occurs that contains
image media, a channel is not yet associated with the SIP dialog.
The file descriptor associated with the udptl session needs to be
set in initialize_udptl or in sip_new to account for this
scenario.
2012-01-17 01:37 +0000 [r351182] Russell Bryant <russell@russellbryant.com>
* channels/chan_sip.c: Add some missing locking in chan_sip. This
patch adds some missing locking to the function
send_provisional_keepalive_full(). This function is called from
the scheduler, which is processed in the SIP monitor thread. The
associated channel (or pbx) thread will also be using the same
sip_pvt and ast_channel so locking must be used. The
sip_pvt_lock_full() function is used to ensure proper locking
order in a safe manner. In passing, document a suspected
reference counting error in this function. The "fix" is left
commented out because when the "fix" is present, crashes occur.
My theory is that fixing it is exposing a reference counting
error elsewhere, but I don't know where. (Or my analysis of this
being a problem could have been completely wrong in the first
place). Leave the comment in the code for so that someone may
investigate it again in the future. Also add a bit of doxygen to
transmit_provisional_response(). (closes issue ASTERISK-18979)
Review: https://reviewboard.asterisk.org/r/1648
2012-01-16 21:12 +0000 [r351080-351130] Terry Wilson <twilson@digium.com>
* channels/chan_sip.c: Ensure ACK retransmit & hangup on non-200
response to INVITE When handling a non-2xx final response on an
INVITE transaction, we have to keep the transaction around after
we send an ACK in case we receive a retransmission of the
response so we can re-transmit the ACK, but also tear down the
ast_channel as soon as we transmit the ACK. Before this patch, we
could fail at both of these things. Calling
sip_alreadygone/needdestroy prevented us from keeping the
transaction up and retransmitting the ACK, and queueing
CONGESTION was not sufficient to cause the channel to be torn
down when originating calls via the CLI, for example. This patch
queues a hangup with CONGESTION instead of just queueing
CONGESTION for these responses and removes the sip_alreadygone
and sip_needdestroy calls from handle_response_invite on non-2xx
responses. It relies on the hangup calling sip_scheddestroy. For
more information, see section 17.1.1.1 of RFC 3261. (closes issue
ASTERISK-17717) Review: https://reviewboard.asterisk.org/r/1672/
* channels/chan_sip.c: Don't prematurely stop SIP session timer
When Asterisk is the UAS (incoming call, endpoint is re-inviting)
the SIP session timer expires after half the time the sip
endpoint indicates in the Session-expires header in
proc_session_timer(). The session timer was being stopped totally
and being handled as an error case instead of running again until
the second expiry. This patch treats the half-time expiry as a
non-error case and continues the timer until the true expiry.
(closes issue ASTERISK-18996) Reported by: Thomas Arimont Tested
by: Thomas Arimont Patches: session_timer_fix.diff by Terry
Wilson (License #5357) based on session_timer.patch by Thomas
Arimont (License #5525)
2012-01-16 19:09 +0000 [r351027] Matthew Jordan <mjordan@digium.com>
* channels/chan_sip.c: Create and initialize udptl only when dialog
negotiates for image media Prior to this patch, the udptl struct
was allocated and initialized when a dialog was associated with a
peer that supported T.38, when a new SIP channel was allocated,
or what an INVITE request was received. This resulted in any
dialog associated with a peer that supported T.38 having udptl
support assigned to it, including the UDP ports needed for
communication. This occurred even in non-INVITE dialogs that
would never send image media. This patch creates and initializes
the udptl structure only when the SDP for a dialog specifies that
image media is supported, or when Asterisk indicates through the
appropriate control frame that a dialog is to support T.38.
(closes issue ASTERISK-16698) Reported by: under Tested by:
Stefan Schmidt Patches: udptl_20120113.diff uploaded by mjordan
(License #6283) (closes issue ASTERISK-16794) Reported by: Elazar
Broad Tested by: Stefan Schmidt review:
https://reviewboard.asterisk.org/r/1668/
2012-01-16 17:04 +0000 [r350975] Joshua Colp <jcolp@digium.com>
* main/rtp_engine.c: Add missing code to set direct RTP setup
information during dialing.
2012-01-15 20:07 +0000 [r350885-350888] Walter Doekes <walter+asterisk@wjd.nu>
* main/asterisk.c: Allow only one thread at a time to do asterisk
cleanup/shutdown. Add locking around the really-really-quit part
of the core stop/restart part. Previously more than one thread
could be called to do cleanup, causing atexit handlers to be run
multiple times, in turn causing segfaults. (issue ASTERISK-18883)
Reviewed by: Terry Wilson Review:
https://reviewboard.asterisk.org/r/1662/ Review:
https://reviewboard.asterisk.org/r/1658/
* utils/extconf.c: Fix -Werror=unused-but-set-variable compile
error in utils/extconf.c. Note that I'm not confirming legitimacy
of having that file in tree at all. Is anyone using
aelparse/conf2ael? (issue ASTERISK-15350)
2012-01-14 16:40 +0000 [r350788-350837] Kevin P. Fleming <kpfleming@digium.com>
* autoconf/libcurl.m4, configure, autoconf/ast_gcc_attribute.m4,
configure.ac: Ensure that all AC_LANG_PROGRAM calls in the
configure script are properly quoted. Recent versions of autoconf
(2.68 on my system) won't properly process the configure script
unless every call to AC_LANG_PROGRAM is m4-quoted. Many calls in
the script were, but many were not. This patch corrects the
unquoted calls.
* addons/chan_mobile.c, channels/chan_h323.c: Correct some
'set-but-not-used' variable warnings.
* contrib/scripts/install_prereq: Ensure that two prerequisites are
properly installed on Debian-style distributions. * Don't specify
a specific version of libgmime; newer versions are available now
and acceptable. * Install libsrtp so that res_srtp can be built.
2012-01-13 22:05 +0000 [r350736] Kinsey Moore <kmoore@digium.com>
* configure, include/asterisk/autoconfig.h.in: Run bootstrap.sh for
the for the ASTERISK-18929 fix configure and autoconfig.h.in were
not regenerated when the fix was committed.
2012-01-13 21:51 +0000 [r350733] Richard Mudgett <rmudgett@digium.com>
* configs/cel_pgsql.conf.sample, configs/cel_odbc.conf.sample:
Correct eventtype names in cel_odbc and cel_pgsql sample files
2012-01-13 21:40 +0000 [r350730] Kinsey Moore <kmoore@digium.com>
* bootstrap.sh, main/asterisk.c, configure.ac: Make sure asterisk
builds on OpenBSD OpenBSD defines SO_PEERCRED, but it returns a
'struct sockpeercred', not 'struct ucred', which causes
compilation of main/asterisk.c to fail in read_credentials().
This allows configure to check for sockpeercred and asterisk to
deal with it properly. (closes issue ASTERISK-18929) Reported-by:
Barry Miller Patch-by: Barry Miller
2012-01-13 20:29 +0000 [r350679] Mark Michelson <mmichelson@digium.com>
* channels/sip/config_parser.c: Set port to a default sane value if
a bogus one is provided when parsing hostnames.
2012-01-13 17:23 +0000 [r350555-350571] Richard Mudgett <rmudgett@digium.com>
* configs/cel_pgsql.conf.sample, configs/cel_odbc.conf.sample,
cel/cel_pgsql.c, cel/cel_odbc.c, cel/cel_manager.c: Use
compatible names for event extra data for various CEL backends. *
Change eventextra to extra in cel_psql.c and cel_odbc.c. * Change
EventExtra to Extra in cel_manager.c. (issue ASTERISK-17190)
* configs/cel_pgsql.conf.sample, configs/cel_odbc.conf.sample,
main/cel.c, configs/cel_custom.conf.sample, cel/cel_pgsql.c,
configs/cel_sqlite3_custom.conf.sample, cel/cel_odbc.c,
configs/cel.conf.sample, cel/cel_manager.c: Add missing CEL
logging fields to various CEL backends. * Add missing eventextra
to cel_psql.c and cel_odbc.c. * Add missing PeerAccount and
EventExtra to cel_manager.c. * Add missing userdeftype support
for cel_custom.conf.sample and cel_sqlite3_custom.conf.sample.
(closes issue ASTERISK-17190) Reported by: Bryant Zimmerman
2012-01-13 16:57 +0000 [r350552] Matthew Jordan <mjordan@digium.com>
* apps/app_queue.c: Realtime queues failed to load queue
information without queue member table Previously, realtime
queues could be loaded without defining the queue member table.
This allowed for queue members to be dynamic, while the realtime
queue definitions could exist in some backing storage. Revision
342223 broke this when it changed the return value for
realtime_multientry to return NULL when no results are returned.
Previously, an empty ast_config object was expected. (closes
issue ASTERISK-19170) Reported by: Rene Mendoza Tested by: Rene
Mendoza Patches: rt_queue_member_patch.diff uploaded by Matt
Jordan (license 6283)
2012-01-12 15:57 +0000 [r350501] Jonathan Rose <jrose@digium.com>
* main/features.c: Adds peer to CEL report on CEL_BRIDGE_START and
CEL_BRIDGE_END (closes issue ASTERISK-17940) Reporter: Nic
Colledge Patches: features_18.patch uploaded by Nic Colledge
(license 6245)
2012-01-11 22:50 +0000 [r350311-350452] Richard Mudgett <rmudgett@digium.com>
* main/cel.c: Remove extraneous BRIDGEPEER AMI VarSet event on a
CEL dummy channel. (closes issue ASTERISK-19180) Reported by:
Corey Farrell Patches: asterisk_cel_noevent_varset.diff (license
#5909) patch uploaded by Corey Farrell
* CHANGES, apps/app_followme.c, apps/app_dial.c: Make FollowMe
optionally update connected line information when the accepting
endpoint is bridged. Like Dial and Queue, FollowMe needs to deal
with AST_CONTROL_CONNECTED_LINE information so when the parties
are initially bridged, the connected line information will be
correct. * Added the 'I' option just like the app_dial and
app_queue 'I' option. (closes issue ASTERISK-18969) Reported by:
rmudgett Tested by: rmudgett Review:
https://reviewboard.asterisk.org/r/1656/
* funcs/func_lock.c: Fix absolute/relative time mismatch in LOCK
function. The time passed by the LOCK function to an internal
function was relative time when the function expected absolute
time. * Don't use C++ keywords in get_lock(). (closes issue
ASTERISK-16868) Reported by: Andrey Solovyev Patches:
20101102__issue18207.diff.txt (license #5003) patch uploaded by
Andrey Solovyev (modified)
2012-01-09 21:54 +0000 [r350075-350220] Richard Mudgett <rmudgett@digium.com>
* channels/chan_iax2.c: Fix joinable thread terminating without
joiner memory leak in chan_iax.c. The iax2_process_thread() can
exit without anyone waiting to join the thread. If noone is
waiting to join the thread then a large memory leak occurs. *
Made iax2_process_thread() deatach itself if nobody is waiting to
join the thread. (closes issue ASTERISK-17339) Reported by:
Tzafrir Cohen Patches:
asterisk-1.8.4.4-chan_iax2-detach-thread-on-non-stop-exit.patch
(license #5617) patch uploaded by Alex Villacis Lasso (modified)
(closes issue ASTERISK-17825) Reported by: wangjin
* contrib/scripts/live_ast: live_ast: valgrind: run asterisk under
valgrind Adds a new sub-command, "valgrind" to live_ast. It runs
asterisk under valgrind. The extra command-line parameters are
passed to Asterisk as usual, and parameters to valgrind are
passed through LIVE_AST_VALGRIND_ARGS in live.conf . Review:
https://reviewboard.asterisk.org/r/1109/ Merged revisions 326636
from http://svn.asterisk.org/svn/asterisk/branches/10
* contrib/scripts/live_ast, contrib/scripts/valgrind_compare
(added): Update contrib script live_ast to invoke Asterisk with
valgrind and suppression file. * Added valgrind_compare script to
compare two valgrind log files for differences. (issue
ASTERISK-17339) Reported by: Tzafrir Cohen Patches:
valgrind_compare (license #5035) script uploaded by Tzafrir Cohen
live_ast_valgrind.diff (license #5035) patch uploaded by Tzafrir
Cohen live_ast_valgrind_v2.diff (license #5185) patch uploaded by
Paul Belanger
* main/asterisk.c: Make Asterisk -x command line parameter imply -r
parameter presence. The Asterisk -x command line parameter is
documented inconsistently. * Made the -x documentation and
behavior consistent. * Since this is also a new year, updated the
copyright notices while here. (closes issue ASTERISK-19094)
Reported by: Eugene Patches:
issueA19094_correct_asterisk_option_x.patch (license #5674) patch
uploaded by Walter Doekes (modified) Tested by: Eugene
2012-01-09 15:37 +0000 [r350023] Kinsey Moore <kmoore@digium.com>
* apps/app_meetme.c: Prevent SLA settings from getting wiped out on
reload If SLA was reloaded without the config file being changed,
current settings got wiped out before the SLA reload code decided
it wasn't going to reload the file since nothing was changed.
Moving the settings reset later in the reload process fixes this.
(closes issue AST-744)
2012-01-06 23:17 +0000 [r349968] Terry Wilson <twilson@digium.com>
* channels/chan_sip.c: Don't leak CID in From header when
presentation=unavailable When someone does
Set(CALLERPRES()=unavailable) (or
Set(CALLERID(pres)=unavailable)) when sendrpid=no, the From
header shows "Anonymous" <anonymous@anonymous.invalid>. When
sendrpid=yes/pai, the From header will still display the callerid
info, even though we supply an rpid header with the anonymous
info. It seems like we shouldn't leak that info in any case.
Skimming http://tools.ietf.org/html/draft-ietf-sip-privacy-04
seems to indicate that one shouldn't send identifying info in the
From in this case. This patch anonymizes the From header as well
even when sendrpid=yes/pai. (closes issue ASTERISK-16538) Review:
https://reviewboard.asterisk.org/r/1649/
2012-01-06 16:46 +0000 [r349819-349872] Richard Mudgett <rmudgett@digium.com>
* apps/app_followme.c: Fix memory leaks in app_followme
find_realtime(). (closes issue ASTERISK-19055) Reported by: Matt
Jordan
* cel/cel_sqlite3_custom.c: Make not assume that the
cel_sqlite3_custom SQL table primary key is AcctId. If a table is
created by some other application and the primary key is not
named "AcctId", cel/cel_sqlite3_custom.c will always try to
create the table and fail because it already exists. * Change the
SQL table query to not require AcctId as the primary key. (closes
issue ASTERISK-18963) Reported by: socketpair Patches: fix.patch
(license #6337) patch uploaded by socketpair
2012-01-05 22:06 +0000 [r349731] Kinsey Moore <kmoore@digium.com>
* main/file.c: Allow playback of formats that don't support seeking
ast_streamfile previously did unconditional seeking on files that
broke playback of formats that don't support that functionality.
This patch avoids the seek that was causing the problem. This
regression was introduced in r158062. (closes issue
ASTERISK-18994) Patch-by: Timo Teras
2012-01-05 21:46 +0000 [r349672-349728] Jonathan Rose <jrose@digium.com>
* main/dsp.c: Fix an issue where dsp.c would interpret multiple
dtmf events from a single key press. When receiving calls from a
mobile phone into a DISA system on a connection with significant
interference, the reporter's Asterisk system would interpret DTMF
incorrectly and replicate digits received. This patch resolves
that by increasing the number of frames a mismatch has to be
detected before assuming the DTMF is over by 1 frame and adjusts
dtmf_detect function to reset hits and misses only when an edge
is detected. (closes issue ASTERISK-17493) Reported by: Alec
Davis Patches: bug18904-refactor.diff.txt uploaded by Alec Davis
(license 5546) Review: https://reviewboard.asterisk.org/r/1130/
* main/asterisk.c: Ensures Asterisk closes when receiving terminal
signals in 'no fork' mode. When catching a signal, in no fork
mode the console thread is identical to the thread responsible
for catching the signal and closing Asterisk, which requires it
to first dispense with the console thread. Prior to this patch,
if these threads were identical, upon receiving a killing signal,
the thread will send an URG signal to itself, which we also catch
and then promptly do nothing with. Obviously this isn't useful
behavior. (closes issue ASTERISK-19127) Reported By: Bryon Clark
Patches: quit_on_signals.patch uploaded by Bryon Clark (license
6157)
2012-01-04 20:46 +0000 [r349558] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c: Fix segfault in chan_dahdi for
CHANNEL(dahdi_span) evaluation on hangup. * Added NULL private
pointer checks in the following chan_dahdi channel callbacks:
dahdi_func_read(), dahdi_func_write(), dahdi_setoption(), and
dahdi_queryoption(). (closes issue ASTERISK-19142) Reported by:
Diego Aguirre Tested by: rmudgett
2012-01-04 20:23 +0000 [r349504-349529] Kinsey Moore <kmoore@digium.com>
* contrib/init.d/rc.debian.asterisk: Make debian init script
conform to the LSB standard Previously, this init script would
return 1 if Asterisk was already running. This is incorrect
behavior according to the LSB standard and has been fixed by
returning 0 instead. (closes issue ASTERISK-17958) Reported-by:
johnc
* contrib/scripts/autosupport, contrib/scripts/autosupport.8:
Update autosupport script and man page Added information
collection from the output of the utilities: top, free, uptime,
ifconfig Added information collection from the output of the
Asterisk command 'dahdi show status' Added option / flag '-n,
--non-interactive' Updated man page to reflect new option / flag
'-n, --non-interactive' Patch-by: John Bigelow (itzanger) (closes
issue AST-749)
2012-01-04 19:27 +0000 [r349450-349482] Jonathan Rose <jrose@digium.com>
* channels/chan_sip.c: Adds Subscription-State header to notify
with call completion. per RFC3265 (Closes issue ASTERISK-17953)
Reported by: George Konopacki Patches: 19400.patch uploaded by
mmichelson (license 5049)
* main/pbx.c: Fix documentation for SayNumber to reflect the fact
that language is changed in CHANNEL() (closes issue
ASTERISK-18962) reported by: Nir Simionovich
2012-01-27 Asterisk Development Team <asteriskteam@digium.com>
* Asterisk 1.8.9.0 Released.
* Test results:
http://bamboo.asterisk.org/browse/TESTING-ASTERISK1890RCS-6
2012-01-24 Asterisk Development Team <asteriskteam@digium.com>
* Asterisk 1.8.9.0-rc3 Released.
* Test results:
http://bamboo.asterisk.org/browse/TESTING-ASTERISK1890RCS-4
* main/file.c: Allow playback of formats that don't support
seeking. ast_streamfile previously did unconditional seeking
on files that broke playback of formats that don't support that
functionality. This patch avoids the seek that was causing the
problem. (closes issue ASTERISK-18994) Patch-by: Timo Teras
* channels/chan_sip.c: AST-2012-001: prevent crash when an SDP offer
is received with an encrypted video stream when support for video
is disabled and res_srtp is loaded. (closes issue ASTERISK-19202)
Reported by: Catalin Sanda
* channels/chan_sip.c: Fix RTP reference leak. If a blind transfer
were initiated using a REFER without a prior reINVITE to place the
call on hold, AND if Asterisk were sending RTCP reports, then there
was a reference leak for the RTP instance of the transferer.
(closes issue ASERISK-19192) Reported by: Tyuta Vitali
* main/features.c: Fix blind transfers from failing if an 'h' extension
is present. This prevents the 'h' extension from being run on the
transferee channel when it is transferred via a native transfer
mechanism such as SIP REFER. (closes issue ASTERISK-19173) Reported
by: Ross Beer Tested by: Kristjan Vrban Patches: ASTERISK-19173 by
Mark Michelson (license 5049)
2012-01-13 Asterisk Development Team <asteriskteam@digium.com>
* Asterisk 1.8.9.0-rc2 Released.
* Test results:
http://bamboo.asterisk.org/browse/TESTING-ASTERISK1890RCS-3
* apps/app_queue.c: Realtime queues failed to load queue
information without queue member table. Revision 342223
broke this when it changed the return value for
realtime_multientry to return NULL when no results are
returned. (closes issue ASTERISK-19170) Reported by: Rene
Mendoza Tested by: Rene Mendoza
2011-12-30 Asterisk Development Team <asteriskteam@digium.com>
* Asterisk 1.8.9.0-rc1 Released.
* Test results:
http://bamboo.asterisk.org/browse/TESTING-ASTERISK1890RCS-2
2011-12-29 15:13 +0000 [r349339] Matthew Jordan <mjordan@digium.com>
* main/rtp_engine.c: Handle AST_CONTROL_UPDATE_RTP_PEER frames in
local bridge loop Failing to handle AST_CONTROL_UPDATE_RTP_PEER
frames in the local bridge loop causes the loop to exit
prematurely. This causes a variety of negative side effects,
depending on when the loop exits. This patch handles the frame by
essentially swallowing the frame in the local loop, as the
current channel drivers expect the RTP bridge to handle the
frame, and, in the case of the local bridge loop, no additional
action is necessary. (issue ASTERISK-19040) (issue
ASTERISK-19128) (issue ASTERISK-17725) (issue ASTERISK-18340)
(closes issue ASTERISK-19095) Reported by: Stefan Schmidt Tested
by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1640/
2011-12-28 21:30 +0000 [r349289] Sean Bright <sean@malleable.com>
* main/audiohook.c: Use ast_audiohook_write_list_empty to determine
if our lists are empty instead of duplicating that logic.
2011-12-27 20:48 +0000 [r349194] Matthew Jordan <mjordan@digium.com>
* res/res_musiconhold.c, res/res_timing_pthread.c,
include/asterisk/module.h, res/res_timing_dahdi.c,
res/res_timing_timerfd.c: Fix timing source dependency issues
with MOH Prior to this patch, res_musiconhold existed at the same
module priority level as the timing sources that it depends on.
This would cause a problem when music on hold was reloaded, as
the timing source could be changed after res_musiconhold was
processed. This patch adds a new module priority level,
AST_MODPRI_TIMING, that the various timing modules are now loaded
at. This now occurs before loading other resource modules, such
that the timing source is guaranteed to be set prior to resolving
the timing source dependencies. (closes issue ASTERISK-17474)
Reporter: Luke H Tested by: Luke H, Vladimir Mikhelson, zzsurf,
Wes Van Tlghem, elguero, Thomas Arimont Patches:
asterisk-17474-dahdi_timing-infinite-wait-fix_v3_branch-1.8.diff
uploaded by elguero (License #5026)
asterisk-17474-dahdi_timing-infinite-wait-fix_v3_branch-10.diff
uploaded by elguero (License #5026)
asterisk-17474-dahdi_timing-infinite-wait-fix_v3.diff uploaded by
elguero (License #5026) Review:
https://reviewboard.asterisk.org/r/1578/
2011-12-27 17:09 +0000 [r349144] Sean Bright <sean@malleable.com>
* main/audiohook.c: Once an audiohook is attached to a channel, we
continue to transcode all of the frames, even after all of the
hooks are detached. This patch short-cicuits us out before we
transcode unnecessarily.
2011-12-23 17:25 +0000 [r349044] Sean Bright <sean@malleable.com>
* apps/app_chanspy.c: In ChanSpy, don't create audiohooks that will
never be used. When ChanSpy is initialized it creates and
attaches 3 audiohooks: 1) Read audio off of the channel that we
are spying on 2) Write audio to the channel that we are spying on
3) Write audio to the channel that is bridged to the channel that
we are spying on. The first is always necessary, but the others
are used only when specific options are passed to the ChanSpy
application (B, d, w, and W to be specific). When those flags are
not passed, neither of those audiohooks are ever sent frames, but
we still try to process the hooks for each voice frame that we
recieve on the channel. So in short - only create and attach
audiohooks that we actually need.
2011-12-23 15:24 +0000 [r348992] Kinsey Moore <kmoore@digium.com>
* apps/app_dial.c: Fix missing doc tags found while fixing
ASTERISK-18689 Add missing <variable></variable> tags in app_dial
documentation.
2011-12-23 02:09 +0000 [r348940] Richard Mudgett <rmudgett@digium.com>
* include/asterisk/pbx.h, main/pbx.c, channels/chan_sip.c: Fix
extension state callback references in chan_sip. Chan_sip gives a
dialog reference to the extension state callback and assumes that
when ast_extension_state_del() returns, the callback cannot
happen anymore. Chan_sip then reduces the dialog reference count
associated with the callback. Recent changes (ASTERISK-17760)
have resulted in the potential for the callback to happen after
ast_extension_state_del() has returned. For chan_sip, this could
be very bad because the dialog pointer could have already been
destroyed. * Added ast_extension_state_add_destroy() so chan_sip
can account for the sip_pvt reference given to the extension
state callback when the extension state callback is deleted. *
Fix pbx.c awkward statecbs handling in
ast_extension_state_add_destroy() and handle_statechange() now
that the struct ast_state_cb has a destructor to call. * Ensure
that ast_extension_state_add_destroy() will never return -1 or 0
for a successful registration. * Fixed pbx.c statecbs_cmp() to
compare the correct information. The passed in value to compare
is a change_cb function pointer not an object pointer. * Make
pbx.c ast_merge_contexts_and_delete() not perform callbacks with
AST_EXTENSION_REMOVED with locks held. Chan_sip is notorious for
deadlocking when those locks are held during the callback. *
Removed unused lock declaration for the pbx.c store_hints list.
(closes issue ASTERISK-18844) Reported by: rmudgett Review:
https://reviewboard.asterisk.org/r/1635/
2011-12-22 22:31 +0000 [r348888] Matthew Jordan <mjordan@digium.com>
* cel/cel_pgsql.c: Fix for memory leaks / cleanup in cel_pgsql
There were a number of issues in cel_pgsql's pgsql_log method: *
If either sql or sql2 could not be allocated, the method would
return while the pgsql_lock was still locked * If the execution
of the log statement succeeded, the sql and sql2 structs were
never free'd * Reconnection successes were logged as ERRORs. In
general, the severity of several logging statements was reduced
(closes issue ASTERISK-18879) Reported by: Niolas Bouliane Tested
by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1624/
2011-12-22 18:38 +0000 [r348833] Terry Wilson <twilson@digium.com>
* include/asterisk/frame.h: Allow packetization vaules > 127
According to the RTP packetization documentation, and the maximum
values listed in AST_FORMAT_LIST, we should support values > that
the signed char array that ast_codec_pref makes available to
store the value. All places in the code treat the framing field
as though it were an int array instaead of a char array anyway,
so this just fixes the type of the array. (closes issue
ASTERISK-18876) Review: https://reviewboard.asterisk.org/r/1639/
2011-12-20 23:08 +0000 [r348735] Richard Mudgett <rmudgett@digium.com>
* channels/chan_iax2.c: Fix chan_iax2 to not report an RDNIS number
if it is blank. Some ISDN switches complain or block the call if
the RDNIS number is empty. * Made chan_iax2 not save a RDNIS
number into the ast_channel if the string is blank. This is what
other channel drivers do. (closes issue ASTERISK-17152) Reported
by: rmudgett
2011-12-19 21:31 +0000 [r348647] Richard Mudgett <rmudgett@digium.com>
* configure, configure.ac: Fix crashes on other platforms caused by
interference from Darwin weak symbol support. Support weak
symbols on a platform specific basis. The Mac OS X (Darwin)
support must be isolated from the other platforms because it has
caused other platforms to crash. Several other platforms
including Linux have GCC versions that define the weak attribute.
However, this attribute is only setup for use in the code by
Darwin. (closes issue ASTERISK-18728) Reported by: Ben Klang
Review: https://reviewboard.asterisk.org/r/1617/
2011-12-18 18:27 +0000 [r348516] Kevin P. Fleming <kpfleming@digium.com>
* configs/sip.conf.sample, /: Correct two flaws in sip.conf.sample
related to AST-2011-013. * The sample file listed *two* values
for the 'nat' option as being the default. Only 'force_rport' is
the default. * The warning about having differing 'nat' settings
confusingly referred to both peers and users. ........ Merged
revisions 348515 from
http://svn.asterisk.org/svn/asterisk/branches/1.6.2
2011-12-16 23:51 +0000 [r348310-348464] Richard Mudgett <rmudgett@digium.com>
* main/channel.c, main/features.c: Clean-up on isle five for
__ast_request_and_dial() and ast_call_forward(). * Add locking
when a channel inherits variables and datastores in
__ast_request_and_dial() and ast_call_forward(). Note: The
involved channels are not active so there was minimal potential
for problems. * Remove calls to ast_set_callerid() in
__ast_request_and_dial() and ast_call_forward() because the set
information is for the wrong direction. * Don't use C++ keywords
for variable names in ast_call_forward(). * Run the redirecting
interception macro if defined when forwarding a call in
ast_call_forward(). Note: Currently will never execute because
the only callers that supply a calling channel supply a hungup or
zombie channel. * Make feature_request_and_dial() put the
transferee into autoservice when it calls ast_call_forward() in
case a redirection interception macro is run. Note: Currently
will never happen because the caller channel (Party B) is always
hungup at this time. * Make feature_request_and_dial() ignore the
AST_CONTROL_PROCEEDING frame to silence a log message.
* main/channel.c: Fix cut and past error in ast_call_forward().
(issue ASTERISK-18836)
* include/asterisk/cdr.h, apps/app_followme.c, apps/app_queue.c,
res/res_monitor.c, main/channel.c, main/pbx.c,
apps/app_authenticate.c, funcs/func_cdr.c, main/features.c: Fix
crash during CDR update. The ast_cdr_setcid() and
ast_cdr_update() were shown in ASTERISK-18836 to be called by
different threads for the same channel. The channel driver thread
and the PBX thread running dialplan. * Add lock protection around
CDR API calls that access an ast_channel pointer. (closes issue
ASTERISK-18836) Reported by: gpluser Review:
https://reviewboard.asterisk.org/r/1628/
* apps/app_parkandannounce.c: Fix ParkAndAnnounce to pass the
CallerID to the announcing channel. ParkAndAnnounce tried to pass
the CallerID to the announcing channel but the ID was wiped out
by the channel masquerade done when parking the call. * Save the
CallerID before parking the channel to pass it to the announcing
channel. * Fixed a minor memory leak in ParkAndAnnounce. *
Updated some ParkAndAnnounce log messages.
2011-12-14 22:01 +0000 [r348212] Matthew Nicholson <mnicholson@digium.com>
* res/res_fax.c: Don't clear LOCALSTATIONID before sending or
receiving. The user may set that variable. ASTERISK-18921
2011-12-14 20:34 +0000 [r348154-348157] Jonathan Rose <jrose@digium.com>
* configs/features.conf.sample: Fix accidental use of tabs instead
of spaces from previous features.conf.sample change
* configs/features.conf.sample: Document PARKINGSLOT variable in
features.conf.sample (issue ASTERISK-16239)
2011-12-13 23:00 +0000 [r348101] Richard Mudgett <rmudgett@digium.com>
* apps/app_followme.c, bridges/bridge_builtin_features.c: Fix
FollowMe CallerID on outgoing calls. The addition of the
Connected Line support changed how CallerID is passed to outgoing
calls. The FollowMe application was not updated to pass CallerID
to the outgoing calls. * Fix FollowMe CallerID on outgoing calls.
* Restructured findmeexec() to fix several memory leaks and
eliminate some duplicated code. * Made check the return value of
create_followme_number(). Putting a NULL into the numbers list is
bad if create_followme_number() fails. * Fixed a couple uses of
ast_strdupa() inside loops. * The changes to
bridge_builtin_features.c fix a similar CallerID issue with the
bridging API attended and blind transfers. (Not used at this
time.) (closes issue ASTERISK-17557) Reported by: hamlet505a
Tested by: rmudgett Review:
https://reviewboard.asterisk.org/r/1612/
2011-12-13 15:16 +0000 [r348048] Stefan Schmidt <sst@sil.at>
* channels/chan_sip.c: Fix possible misshandling of an incoming SIP
response as a peer poke response. Also make sure peer has even
qualify enabled when handle a peer poke response. (closes issue
ASTERISK-18940) Reported by: Vitaliy Tested by: Vitaliy and
UnixDev Review: https://reviewboard.asterisk.org/r/1620 Reviewed
by: David Vossel
2011-12-12 19:22 +0000 [r347995] Terry Wilson <twilson@digium.com>
* res/res_srtp.c: Add a separate buffer for SRTCP packets The
function ast_srtp_protect used a common buffer for both SRTP and
SRTCP packets. Since this function can be called from multiple
threads for the same SRTP session (scheduler for SRTCP and
channel for SRTP) it was possible for the packets to become
corrupted as the buffer was used by both threads simultaneously.
This patch adds a separate buffer for SRTCP packets to avoid the
problem. (closes issue ASTERISK-18889, Reported/patch by Daniel
Collins)
2011-12-09 01:19 +0000 [r347811] Richard Mudgett <rmudgett@digium.com>
* main/pbx.c: Fix some parsing issues in
add_exten_to_pattern_tree(). * Simplify compare_char() and avoid
potential sign extension issue. * Fix infinite loop in
add_exten_to_pattern_tree() handling of character set escape
handling. * Added buffer overflow checks in
add_exten_to_pattern_tree() character set collection. * Made
ignore empty character sets. * Added escape character handling to
end-of-range character in character sets. This has a slight
change in behavior if the end-of-range character is an escape
character. You must now escape it. * Fix potential sign extension
issue when expanding character set ranges. * Made remove
duplicated characters from character sets. The duplicate
characters lower extension matching priority and prevent
duplicate extension detection. * Fix escape character handling
when the escape character is trying to escape the end-of-string.
We could have continued processing characters after the end of
the exten string. We could have added the previous character to
the pattern matching tree incorrectly. (closes issue
ASTERISK-18909) Reported by: Luke-Jr
2011-12-08 21:28 +0000 [r347718] Walter Doekes <walter+asterisk@wjd.nu>
* channels/chan_sip.c: Fix regression when using tcpenable=no and
tlsenable=yes. The tlsenable settings are tucked away in
main/tcptls.c, so I missed them when resolving ASTERISK-18837.
This should resolve the test suite breakage of the sip tls tests.
Review: https://reviewboard.asterisk.org/r/1615 Reviewed by: Matt
Jordan
2011-12-08 17:50 +0000 [r347595] Richard Mudgett <rmudgett@digium.com>
* main/features.c: Mark channel running the h exten with the
soft-hangup flag. When a bridge is broken, ast_bridge_call()
might execute the h exten on the calling channel. However, that
channel may not have been the channel that broke the bridge by
hanging up. The channel executing the h exten must be in a hung
up state so things like AGI run in the correct mode. * Make sure
ast_bridge_call() marks the channel it is executing the h exten
on as hung up. (The AST_SOFTHANGUP_APPUNLOAD flag is used so as
to match the pbx.c main dialplan execution loop when it executes
the h exten.) (closes issue ASTERISK-18811) Reported by: David
Hajek Patches: jira_asterisk_18811_v1.8.patch (license #5621)
patch uploaded by rmudgett Tested by: David Hajek, rmudgett
2011-12-08 16:19 +0000 [r347531] Terry Wilson <twilson@digium.com>
* /, channels/chan_sip.c: Don't crash on INFO automon request with
no channel AST-2011-014. When automon was enabled in
features.conf, it was possible to crash Asterisk by sending an
INFO request if no channel had been created yet. (closes issue
ASTERISK-18805) ........ Merged revisions 347530 from
http://svn.asterisk.org/svn/asterisk/branches/1.6.2
2011-12-07 21:36 +0000 [r347438] Richard Mudgett <rmudgett@digium.com>
* main/manager.c: Update AMI Getvar and Setvar documentation about
supplying a channel name. (closes issue ASTERISK-18958) Reported
by: Red Patches: jira_asterisk_18958_v1.8.patch (license #5621)
patch uploaded by rmudgett
2011-12-07 20:23 +0000 [r347369] Jonathan Rose <jrose@digium.com>
* apps/app_meetme.c: Fix: Meetme recording variables from realtime
DB use null entries over channel variables Meetme would attempt
to substitute the realtime values of RECORDING_FILE and
RECORDING_FORMAT from the meetme db entry instead of using the
channel variable set for those variables in spite of those
database entries being NULL or even lacking a column to represent
them. (closes issue ASTERISK-18873) Reported by: Byron Clark
Patches: ASTERISK-18873-1.patch uploaded by Byron Clark (license
6157)
2011-12-06 23:47 +0000 [r347292] Richard Mudgett <rmudgett@digium.com>
* channels/chan_sip.c: Make SIP INFO messages for dtmf-relay
signals case insensitive. (closes issue ASTERISK-18924) Reported
by: Kevin Taylor
2011-12-06 21:44 +0000 [r347239] Jonathan Rose <jrose@digium.com>
* main/pbx.c: Documents CHANNEL(musicclass) taking priority over
m([x]) in waitExten If waitExten specifies a music class to use
with its music on hold option, it will use CHANNEL(musicclass)
instead if that channel variable has been set on the initiating
channel. This documents that behavior in the waitExten app so
that this can be known without checking the documentation of the
code in function local_ast_moh_start. (closes issue
ASTERISK-18804)
2011-12-06 19:39 +0000 [r347111-347166] Walter Doekes <walter+asterisk@wjd.nu>
* channels/chan_sip.c: Don't allow transport=tcp when tcpenable=no.
When tcpenable=no, sending to transport=tcp hosts was still
allowed. Resolving the source address wasn't possible and yielded
the string "(null)" in SIP messages. Fixed that and a couple of
not-so-correct log messages. (closes issue ASTERISK-18837)
Reported by: Andreas Topp Review:
https://reviewboard.asterisk.org/r/1585 Reviewed by: Matt Jordan
* apps/app_voicemail.c: Add regression tests for issue
ASTERISK-18838. Review: https://reviewboard.asterisk.org/r/1572
Reviewed by: Matt Jordan
* apps/app_voicemail.c: Move setting of voicemail zonetag and
locale up a bit. The voicemail [general] zonetag and locale
variables weren't loaded until after the mailboxes were
initialized. This caused the settings to be unset for those
mailboxes until a reload was performed. (closes issue
ASTERISK-18838) Review: https://reviewboard.asterisk.org/r/1570
Reviewed by: Matt Jordan
2011-12-06 17:05 +0000 [r347058] Matthew Jordan <mjordan@digium.com>
* channels/sip/include/sip.h, channels/chan_sip.c: Fixed crash from
orphaned MWI subscriptions in chan_sip This patch resolves the
issue where MWI subscriptions are orphaned by subsequent SIP
SUBSCRIBE messages. When a peer is removed, either by pruning
realtime SIP peers or by unloading / loading chan_sip, the MWI
subscriptions that were orphaned would still be on the event
engine list of valid subscriptions but have a pointer to a peer
that no longer was valid. When an MWI event would occur, this
would cause a seg fault. (closes issue ASTERISK-18663) Reported
by: Ross Beer Tested by: Ross Beer, Matt Jordan Patches:
blf_mwi_diff_12_06_11.txt uploaded by Matt Jordan (license 6283)
Review: https://reviewboard.asterisk.org/r/1610/
2011-12-05 17:39 +0000 [r347006] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c, channels/sig_analog.c,
channels/sig_analog.h: Restore call progress code for analog
ports. Extracting sig_analog from chan_dahdi lost call progress
detection functionality. * Fix analog ports from considering a
call answered immediately after dialing has completed if the
callprogress option is enabled. (closes issue ASTERISK-18841)
Reported by: Richard Miller Patches: chan_dahdi.diff (license
#5685) patch uploaded by Richard Miller (Modified by me)
sig_analog.c.diff (license #5685) patch uploaded by Richard
Miller (Modified by me) sig_analog.h.diff (license #5685) patch
uploaded by Richard Miller
2011-12-05 14:56 +0000 [r346954] Jonathan Rose <jrose@digium.com>
* main/pbx.c: Resolve duplicate label used in multiple priorities
for the same extension. Prior to this patch, if labels with the
same name were used for different priorities in the same
extension, the new label would be accepted, but it would be
unusable since attempts to reach that label would just go to the
first one. Now pbx.c detects this, generates a warning in logs,
and culls the label before adding it to the dialplan. (closes
issue ASTERISK-18807) Reported by: Kenneth Shumard Patches:
pbx.c.patch uploaded by Kenneth Shumard (License 5077)
2011-12-05 14:45 +0000 [r346951] Kinsey Moore <kmoore@digium.com>
* res/res_jabber.exports.in: Fix chan_jingle/gtalk load regression
introduced in r346087 Add missing symbol exports for
ast_aji_client_destroy and ast_aji_buddy_destroy for usage
outside res_jabber. Testing of these changes focused on
res_jabber itself, so this problem was missed. Reported-by:
Michael Spiceland
2011-12-04 09:57 +0000 [r346899] Walter Doekes <walter+asterisk@wjd.nu>
* channels/chan_sip.c: For SIP REGISTER fix domain-only URIs and
domain ACL bypass. The code that allowed admins to create users
with domain-only uri's had stopped to work in 1.8 because of the
reqresp parser rewrites. This is fixed now: if you have a
[mydomain.com] sip user, you can register with useraddr
sip:mydomain.com. Note that in that case -- if you're using
domain ACLs (a configured domain list) -- mydomain.com must be in
the allow list as well. Reviewboard r1606 shows a list of
registration combinations and which SIP response codes are
returned. Review: https://reviewboard.asterisk.org/r/1533/
Reviewed by: Terry Wilson (closes issue ASTERISK-18389) (closes
issue ASTERISK-18741)
2011-12-02 16:19 +0000 [r346762] Alexandr Anikin <may@telecom-service.ru>
* addons/chan_ooh323.c, channels/chan_h323.c: process null frame
pointer returned by ast_rtp_instance_read correctly (closes issue
ASTERISK-16697) Reported by: under Patches: segfault.diff
(License #5871) patch uploaded by under
2011-12-01 21:11 +0000 [r346700] Richard Mudgett <rmudgett@digium.com>
* configs/res_stun_monitor.conf.sample, include/asterisk/stun.h,
main/stun.c, res/res_stun_monitor.c: Re-resolve the STUN address
if a STUN poll fails for res_stun_monitor. The STUN socket must
remain open between polls or the external address seen by the
STUN server is likely to change. However, if the STUN request
poll fails then the STUN server address needs to be re-resolved
and the STUN socket needs to be closed and reopened. * Re-resolve
the STUN server address and create a new socket if the STUN
request poll fails. * Fix ast_stun_request() return value
consistency. * Fix ast_stun_request() to check the received
packet for expected message type and transaction ID. * Fix
ast_stun_request() to read packets until timeout or an associated
response packet is found. The stun_purge_socket() hack is no
longer required. * Reduce ast_stun_request() error messages to
debug output. * No longer pass in the destination address to
ast_stun_request() if the socket is already bound or connected to
the destination. (closes issue ASTERISK-18327) Reported by:
Wolfram Joost Tested by: rmudgett Review:
https://reviewboard.asterisk.org/r/1595/
2011-12-01 20:36 +0000 [r346564-346697] Jonathan Rose <jrose@digium.com>
* channels/chan_sip.c: Change 183 Ringing in sipfrag body to 180
ringing. 183 Ringing isn't even a thing. 183 is actually a
session progress message. (closes issue ASTERISK-18925) Reported
by: Sebastian Denz Tested by: jrose Patches:
asterisk18-use_180_instead_of_183_in_sipfrag.diff by Sebastian
Denz (License #6139)
* include/asterisk/tcptls.h, main/tcptls.c, channels/chan_sip.c:
r346525 | jrose | 2011-11-30 15:10:38 -0600 (Wed, 30 Nov 2011) |
18 lines Cleaning up chan_sip/tcptls file descriptor closing.
This patch attempts to eliminate various possible instances of
undefined behavior caused by invoking close/fclose in situations
where fclose may have already been issued on a
tcptls_session_instance and/or closing file descriptors that
don't have a valid index for fd (-1). Thanks for more than a
little help from wdoekes. (closes issue ASTERISK-18700) Reported
by: Erik Wallin (issue ASTERISK-18345) Reported by: Stephane
Cazelas (issue ASTERISK-18342) Reported by: Stephane Chazelas
Review: https://reviewboard.asterisk.org/r/1576/
2011-11-30 19:36 +0000 [r346472] Leif Madsen <lmadsen@digium.com>
* configs/queues.conf.sample: Update queues.conf.sample
documentation. Update the documentation surrounding the use of
MONITOR_EXEC to make it more clear that it can be used for both
Monitor() and MixMonitor() usage. (closes issue ASTERISK-17413)
Reported by: David Woolley Patches:
issue18817_mixmonitor_queues_doc.diff by Michael L. Young
(License #5026)
2011-11-28 14:30 +0000 [r346292] Stefan Schmidt <sst@sil.at>
* res/res_rtp_asterisk.c: Fix regression that 'rtp/rtcp set debup
ip' only works when also a port was specified. (closes issue
ASTERISK-18693) Reported by: Davide Dal Fra Review:
https://reviewboard.asterisk.org/r/1600/ Reviewed by: Walter
Doekes
2011-11-23 22:52 +0000 [r346239] Richard Mudgett <rmudgett@digium.com>
* channels/chan_iax2.c, include/asterisk/acl.h,
channels/chan_skinny.c, channels/chan_h323.c, main/acl.c: Fix
calls to ast_get_ip() not initializing the address family.
2011-11-23 20:15 +0000 [r346144-346147] Walter Doekes <walter+asterisk@wjd.nu>
* channels/chan_sip.c: Minor cleanup in chan_sip get_msg_text()
function. In r116240, get_msg_text() got an extra parameter to
fix the unwanted addition of trailing newlines to SIP MESSAGE
bodies. This caused all linefeeds to be trimmed, which isn't
right either. This is a stop-gap; the right fix is to return the
original SIP request body. Review:
https://reviewboard.asterisk.org/r/1586 Reviewed by: Matt Jordan
* include/asterisk/strings.h: Fix ast_str_truncate signedness
warning and documentation. Review:
https://reviewboard.asterisk.org/r/1594
2011-11-23 17:12 +0000 [r346086] Kinsey Moore <kmoore@digium.com>
* channels/chan_gtalk.c, res/res_jabber.c, channels/chan_jingle.c,
include/asterisk/jabber.h: Fix res_jabber resource leaks This
should fix almost all resource leaks in res_jabber that involve
ASTOBJ_CONTAINER_FIND and resolves an ambiguous situation where
ast_aji_get_client would sometimes bump an object's refcount and
sometimes not. Review: https://reviewboard.asterisk.org/r/1553
2011-11-23 16:09 +0000 [r346030] Terry Wilson <twilson@digium.com>
* res/res_musiconhold.c: Resume playing existing hold music for
cached realtime MOH As a result of the fix for ASTERISK-18039,
realtime caching MOH no longer properly resumes playing back a
file between different holds in the same call. This is because
scanning for new files causes the existing file array to be
emptied and we were just comparing that the saved pointer to the
filename matched the pointer to the filename in a particular
position in the array. An easy fix is to save the filename
instead of a pointer to it and then do a strcmp instead of
comparing the addresses. (closes issue ASTERISK-18912) Review:
https://reviewboard.asterisk.org/r/1596/
2011-11-22 22:55 +0000 [r345976] Richard Mudgett <rmudgett@digium.com>
* include/asterisk/dnsmgr.h, main/dnsmgr.c: Fix dnsmgr entries to
ask for the same address family each time. The dnsmgr refresh
would always get the first address found regardless of the
original address family requested. So if you asked for only IPv4
addresses originally, you might get an IPv6 address on refresh. *
Saved the original address family requested by
ast_dnsmgr_lookup() to be used when the address is refreshed.
2011-11-22 20:29 +0000 [r345923] Walter Doekes <walter+asterisk@wjd.nu>
* include/asterisk/logger.h: Clarify why the AST_LOG_* macros exist
next to the LOG_* macros. (issue ASTERISK-17973)
2011-11-21 21:03 +0000 [r345828-345829] Terry Wilson <twilson@digium.com>
* CHANGES: Change nat=yes to nat=force_rport in CHANGES Fix a small
documentation merge issue ASTERISK-18862
* configs/sip.conf.sample, CHANGES, /, channels/chan_sip.c: Default
to nat=yes; warn when nat in general and peer differ It is
possible to enumerate SIP usernames when the general and
user/peer nat settings differ in whether to respond to the port a
request is sent from or the port listed for responses in the Via
header. In 1.4 and 1.6.2, this would mean if one setting was
nat=yes or nat=route and the other was either nat=no or
nat=never. In 1.8 and 10, this would mean when one was
nat=force_rport and the other was nat=no. In order to address
this problem, it was decided to switch the default behavior to
nat=yes/force_rport as it is the most commonly used option and to
strongly discourage setting nat per-peer/user when at all
possible. For more discussion of the issue, please see:
http://lists.digium.com/pipermail/asterisk-dev/2011-November/052191.html
(closes issue ASTERISK-18862) Review:
https://reviewboard.asterisk.org/r/1591/ ........ Merged
revisions 345776 from
http://svn.asterisk.org/svn/asterisk/branches/1.4 ........ Merged
revisions 345800 from
http://svn.asterisk.org/svn/asterisk/branches/1.6.2
2011-11-19 15:08 +0000 [r345682] Tilghman Lesher <tilghman@meg.abyt.es>
* main/db.c: Update the documentation to better clarify how the
existing commands work. Review:
https://reviewboard.asterisk.org/r/1593/
2011-11-17 17:06 +0000 [r345546] Richard Mudgett <rmudgett@digium.com>
* channels/sig_pri.c: Remove dead code since pri_grab() can never
fail. Dead code makes programmers sick. I am sick of looking at
it.
2011-11-17 17:04 +0000 [r345545] Jason Parker <jparker@digium.com>
* apps/app_confbridge.c: Fix documentation of 's' option. The menu
key is #, not *. Reported by p3nguin on #asterisk.
2011-11-16 14:42 +0000 [r345487] Jonathan Rose <jrose@digium.com>
* apps/app_voicemail.c: Guarantee messages go into the right
folders with multiple recipients Before, using the U flag in
Voicemail with multiple recipients would put urgent messages in
the INBOX folder for all users past the first thanks to a bug
with the message copying function. This would also cause messages
to fail to be sent if the INBOX directory hadn't been created for
that mailbox yet. (closes issue ASTERISK-18245) Reported by: Matt
Jordan (closes issue ASTERISK-18246) Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/1589/
2011-11-15 20:09 +0000 [r345219-345431] Richard Mudgett <rmudgett@digium.com>
* res/res_agi.c: Make FastAGI HANGUP show up in AGI debug output. *
Change from using send() to ast_agi_send() so the HANGUP shows up
in the AGI debug output. (closes issue ASTERISK-18723) Reported
by: James Van Vleet Patches: jira_asterisk_18723_v1.8.patch
(license #5621) patch uploaded by rmudgett
* channels/sig_pri.c: Fix typo in sig_pri using wrong structure
name. It is fortunate that the typo does not alter generated code
since the e->restart.channel and e->ring.channel members are in
the same position. (closes issue ASTERISK-18868) Reported by:
zvision Patches: sig_pri.c.diff (License #5755) patch uploaded by
zvision
* apps/app_queue.c: Make queue log indicate if ADDMEMBER is paused
for AMI and realtime. * Add parameter to queue log ADDMEMBER to
indicate if the member is paused. (closes issue ASTERISK-18645)
Reported by: garlew Patches: paused.diff (License #5337) patch
uploaded by garlew Tested by: rmudgett, garlew Review:
https://reviewboard.asterisk.org/r/1469/
* UPGRADE.txt, configs/sip.conf.sample, channels/sip/include/sip.h,
channels/chan_sip.c: Restore SIP DTMF overlap dialing method. The
recent fix for ASTERISK-17288 to get RFC3578 SIP overlap support
working correctly removed a long standing ability to do overlap
dialing using DTMF in the early media phase of a call. See
ASTERISK-18702 it has a very good description of the issue. I
started with Pavel Troller's chan_sip.diff patch on issue
ASTERISK-18702. * Added 'dtmf' enum value to sip.conf
allowoverlap config option. The new option value causes the
Incomplte application to not send anything with chan_sip so the
caller can supply more digits via DTMF. * Renames
SIP_GET_DEST_PICKUP_EXTEN_FOUND to SIP_GET_DEST_EXTEN_MATCHMORE
since that is what it really means. * Fixed get_destination()
inconsistency with the pickup extension matching. * Fixed
initialization of PAGE3 of global_flags in reload_config().
(closes issue ASTERISK-18702) Reported by: Pavel Troller Review:
https://reviewboard.asterisk.org/r/1517/ Review:
https://reviewboard.asterisk.org/r/1582/
* main/pbx.c: Fix Progress spelling error in main/pbx.c. (closes
issue ASTERISK-18857) Reported by: David M Patches:
mainpbx-trivial.patch (License #6326) patch uploaded by David M
2011-11-14 19:05 +0000 [r345163] Terry Wilson <twilson@digium.com>
* main/channel.c: Don't read past end of input when calling write()
int blah = 1; ... write(chan->alertpipe[1], &blah, new_frames *
sizeof(blah)) != (new_frames * sizeof(blah))) is only valid when
new_frames == 1. Otherwise we start reading into adjacent
variables declared on the stack. The read end discards what is
read, so the values don't matter but it's not a good idea to read
past where we want even though new_frames is almost always 1 and
should never be large. This patch is basically taken out of
kpfleming's eventfd branch, as he mentioned that he remembered
fixing it there when I talked to him about this issue. Review:
https://reviewboard.asterisk.org/r/1583/
2011-11-14 19:00 +0000 [r345160] Walter Doekes <walter+asterisk@wjd.nu>
* channels/sip/include/reqresp_parser.h: Update reqresp_parser
parse_uri doxygen comments. The issue mentioned in the bug report
had been fixed recently by twilson. The reporter included this
documentation fix. (closes issue ASTERISK-18572) Reported by:
Richard Miller Patch by: Richard Miller (modified)
2011-11-14 15:08 +0000 [r345063] Kinsey Moore <kmoore@digium.com>
* channels/chan_sip.c: Ensure that a null vmexten does not cause a
segfault When sip_send_mwi_to_peer was modified recently to avoid
deadlocks, vmexten was not expected to be null. This change
handles that situation to avoid a segfault.
2011-11-14 15:00 +0000 [r345062] Jonathan Rose <jrose@digium.com>
* apps/app_voicemail.c: Moves voicemail setup password entry to the
end of the setup process. This change was made because
forcegreeting and forcename settings in voicemail could be
circumvented by hanging up after entering a password, because the
only way voicemail currently observes whether a mailbox is new or
not is by checking to see if the password is the same as the
mailbox number or not. (closes issue ASTERISK-18282) Reported by:
Matt Jordan Review: https://reviewboard.asterisk.org/r/1581/
2011-11-12 16:05 +0000 [r344965] Gregory Nietsky <gregory@distrotech.co.za>
* channels/chan_misdn.c: mISDN Round Robin break when no channel is
available Prevent channels been parsed repetitively.
2011-11-12 00:24 +0000 [r344899] Terry Wilson <twilson@digium.com>
* res/res_musiconhold.c: Don't forget to rescan MOH files for
cached realtime classes Realtime MOH class caching was
implemented because without it, you would build a completely new
MOH class and would start the music over at the beginning each
time hold was pressed in a conversation. Unfortunately, this
broke re-scanning for file changes for realtime MOH classes. This
patch corrects that issue. (closes issue ASTERISK-18039) Review:
https://reviewboard.asterisk.org/r/1579/
2011-11-11 21:54 +0000 [r344835-344843] Walter Doekes <walter+asterisk@wjd.nu>
* main/utils.c, include/asterisk/stringfields.h,
include/asterisk/utils.h: Use __alignof__ instead of sizeof for
stringfield length storage. Kevin P Fleming suggested that
r343157 should use __alignof__ instead of sizeof. For most
systems this won't be an issue, but better fix it now while it's
still fresh. Review: https://reviewboard.asterisk.org/r/1573
* channels/sip/reqresp_parser.c: Remove unneeded if(params) checks
in reqresp_parser. Nick Lewis added them in
https://reviewboard.asterisk.org/r/549/diff/1-2/ for no apparent
reason. There is no way that params could become NULL in that
piece of code, so I removed these excess checks again.
* main/manager.c: Fix bad quoting of multiline mxml opaque_data
that caused invalid xml. The opaque_data was added and enclosed
in single quotes, assuming it would be only a single line. The
rest of the lines were appended after the closing quote. (closes
issue ASTERISK-18852) Reported by: peep_ on IRC Review:
https://reviewboard.asterisk.org/r/1577
2011-11-11 20:42 +0000 [r344823] Matthew Jordan <mjordan@digium.com>
* main/file.c: Video format was treated as audio when removed from
the file playback scheduler This patch fixes the format type
check in ast_closestream and filestream_destructor. Previously a
comparison operator was used, but since audio formats are no
longer contiguous (and AST_FORMAT_AUDIO_MASK includes formats
that have a value greater than the video formats), a bitwise AND
operation is used instead. Duplicated code was also moved to
filestream_close. (closes issue ASTERISK-18682) Reported by: Aldo
Bedrij Tested by: Matt Jordan Review:
https://reviewboard.asterisk.org/r/1580/
2011-11-11 20:10 +0000 [r344769] Kinsey Moore <kmoore@digium.com>
* channels/chan_sip.c: Fix regression introduced by SDP fixups If
capability is adjusted when switching to UDPTL during fax
transmission, fax teardown fails. Make sure capability is only
touched if RTP is active. This regression was introduced in
R344385.
2011-11-11 18:35 +0000 [r344661-344715] Richard Mudgett <rmudgett@digium.com>
* channels/chan_sip.c: Check sip.conf maxforwards parameter for
range 1 <= x <= 255. JIRA AST-710
* main/cli.c: Make CLI "core show channel" not hold the channel
lock during console output. Holding the channel lock while the
CLI "core show channel" command is executing can slow down the
system. It could block the system if the console output is halted
or paused. * Made capture the CLI "core show channel" output into
a buffer to be output after the channel is unlocked. * Removed
use of C++ keyword as a variable name. out renamed to obuf. *
Checked allocation of obuf for failure so will not crash. (closes
issue ASTERISK-18571) Reported by: Pavel Troller Tested by:
rmudgett
2011-11-11 15:21 +0000 [r344608] Jonathan Rose <jrose@digium.com>
* main/pbx.c: Fix a segmentation fault when using an extension with
CID matching and no CID. Attempting to call an extension which
used Caller ID matching with a channel that has an empty caller
id string would result in a segmentation fault. (closes issue
ASTERISK-18392 Reported By: Ales Zelenik
2011-11-10 22:59 +0000 [r344536-344539] Richard Mudgett <rmudgett@digium.com>
* apps/app_queue.c: Fix potential deadlock calling ast_call() with
channel locks held. Fixed app_queue.c:ring_entry() calling
ast_call() with the channel locks held. Chan_local attempts to do
deadlock avoidance in its ast_call() callback and could deadlock
if a channel lock is already held.
* apps/app_queue.c: Make AMI event AgentCalled get
CallerID/ConnectedLine info from the incoming channel. It was
strange that the AgentCalled AMI event would get most of its
information from the incoming channel but then get the CallerID
information from the outgoing channel. Before connected line
support was added, this information was always the same at this
point. (closes issue ASTERISK-18152) Reported by: Thomas Farnham
Tested by: rmudgett
2011-11-10 21:14 +0000 [r344385-344439] Kinsey Moore <kmoore@digium.com>
* apps/app_meetme.c: Fix another incorrect case with meetme's PIN
logic and add documentation This fixes an issue where a user of a
dynamic conference was asked for a PIN twice. This also adds
documentation to assist in future modifications to the piece of
code responsible for PIN checking. (closes issue AST-670)
* channels/sip/include/sip.h, channels/chan_sip.c: Fix several bugs
with SDP parsing and well-formedness of responses Fix bug
ASTERISK-16558 which dealt with the order of responses to
incoming streams defined by SDP. Fix unreported bug where
offering multiple same-type streams would cause Asterisk to reply
with an incorrect SDP response missing one or more streams
without a proper declination. Fix bugs related to a single
non-audio stream being offered with responses requesting codecs
that were not offered in the initial invite along with an
additional audio stream that was not in the initial invite.
Review: https://reviewboard.asterisk.org/r/1516/
2011-11-10 16:18 +0000 [r344330] Matthew Nicholson <mnicholson@digium.com>
* res/res_rtp_asterisk.c: only attempt to do stun handling on ipv4
or ipv4 mapped to ipv6 addresses Patch by: jkonieczny (modified)
ASTERISK-18490
2011-11-09 20:37 +0000 [r344268] Richard Mudgett <rmudgett@digium.com>
* channels/chan_sip.c: Fix deadlock during dialplan reload. Another
deadlock between the conlock/hints and channels/channel locking
orders. * Don't hold the channel and private lock in sip_new()
when calling ast_exists_extension(). (closes issue
ASTERISK-18740) Reported by: Byron Clark Patches:
sip_exists_exten_dlock_3.diff (license #5041) patch uploaded by
Gregory Hinton Nietsky ASTERISK-18740.patch (license #6157) patch
uploaded by Byron Clark Tested by: Byron Clark
2011-11-09 19:57 +0000 [r344215] Terry Wilson <twilson@digium.com>
* channels/sip/include/sip.h,
channels/sip/include/reqresp_parser.h, channels/chan_sip.c,
channels/sip/reqresp_parser.c: Don't treat a host:port string as
a domain The domain matching code prior to 1.8 used to manually
remove the port from the host:port string when determining if an
incoming request matched the list of domains. When switching to
the new parsing functions, the documentation implied that the
"domain" was being returned by these functions, when instead it
was returning the "hostport" as defined by RFC 3261. This led to
confusion and resulted in 1.8+ rejecting an incoming request from
x.x.x.x:xxxxx when domain=x.x.x.x was set in sip.conf. This patch
renames the "domain" variables in the parsing functions to
"hostport" to more accurately describe what it is that they are
returning and also properly truncates the resulting hostport
strings when dealing with domain matching. Review:
https://reviewboard.asterisk.org/r/1574/
2011-11-09 18:42 +0000 [r344158] Alexandr Anikin <may@telecom-service.ru>
* addons/ooh323c/src/ootypes.h, addons/ooh323c/src/oochannels.c,
addons/ooh323c/src/ooq931.c, addons/ooh323c/src/ooh323.c,
addons/ooh323c/src/ooh245.c, addons/ooh323c/src/ooq931.h: (closes
issue ASTERISK-18748) Reported by: Fabrizio Lazzaretti Patches:
ASTERISK-18748-5.patch (License #5415) patch uploaded by may213
Tested by: Fabrizio Lazzaretti
2011-11-09 18:38 +0000 [r344157] Terry Wilson <twilson@digium.com>
* tests/test_netsock2.c: Add a unit test for
ast_sockaddr_split_hostport Review:
https://reviewboard.asterisk.org/r/1575/
2011-11-09 17:13 +0000 [r344102] Kinsey Moore <kmoore@digium.com>
* apps/app_meetme.c: Fix pin parameter behavior regression in
MeetMe The last time this code was touched (by me), a subtlety
was missed based on the difference between needing to check a
pin's validity and the need to prompt for a pin. (closes issue
ASTERISK-18488)
2011-11-09 15:25 +0000 [r344048] Matthew Nicholson <mnicholson@digium.com>
* formats/format_wav.c: don't call ltohl() twice on the same value
ASTERISK-18739 Patch by: pawel (modified)
2011-11-08 19:25 +0000 [r343936] Walter Doekes <walter+asterisk@wjd.nu>
* pbx/pbx_config.c: Fix crash when dialplan remove include is
called with too few arguments. "dialplan remove include x from y"
crashed when the amount of arguments was less than 6. (closes
issue ASTERISK-18762) Reported by: Andrey Solovyev Tested by:
Andrey Solovyev
2011-11-08 17:58 +0000 [r343851] Richard Mudgett <rmudgett@digium.com>
* channels/chan_sip.c, main/acl.c: Fixed reference to incorrect
variable if unknown host configured crash. * Fixed a LOG_ERROR
message referencing the config variable list v that had
previously been processed and became NULL. * Added error return
value set that was missing in an ast_append_ha() error return
path. (closes issue ASTERISK-18743) Reported by: Michele Patches:
issueA18743-fix_dynamic_exclude_static_bad_host_log.patch
(license #5674) patch uploaded by Walter Doekes Tested by:
Michele
2011-11-08 13:26 +0000 [r343791] Leif Madsen <lmadsen@digium.com>
* build_tools/prep_tarball: Fix boo-boo in prep_tarball script. A
hardcoded a branch number was in the prep_tarball which could not
work. Changed it to the variable.
2011-11-07 21:40 +0000 [r343690] Matthew Nicholson <mnicholson@digium.com>
* channels/chan_sip.c: respect case changes in peer names on sip
reload ASTERISK-18669
2011-11-07 21:13 +0000 [r343637] Richard Mudgett <rmudgett@digium.com>
* channels/chan_sip.c: Fix __sip_subscribe_mwi_do() incorectly
changing dialogs hash key callid. Changing an object value used
as a container key requires removing the object from the
container and reinserting it. * Created change_callid_pvt() to
call instead of build_callid_pvt(). The change_callid_pvt() will
correctly change the dialog callid so the ao2 conainter can
explicitly unlink it.
2011-11-07 20:27 +0000 [r343621] Kinsey Moore <kmoore@digium.com>
* channels/chan_sip.c: Prevent BLF subscriptions from causing
deadlocks Fix a locking inversion in sip_send_mwi_to_peer that
was causing deadlocks. This function now requires that both the
peer and associated pvt be unlocked before it is called for cases
where peer and peer->mwipvt form a circular reference. (closes
issue ASTERISK-18663) Review:
https://reviewboard.asterisk.org/r/1563/
2011-11-07 19:36 +0000 [r343577] Richard Mudgett <rmudgett@digium.com>
* channels/chan_sip.c: Fix deadlock if peer is destroyed while
sending MWI notice. A dialog cannot be destroyed by the
ao2_callback dialog_needdestroy because of a deadlock between the
dialogs container lock and the RWLOCK of the events subscription
list. * Create dialogs_to_destroy container to hold dialogs that
will be destroyed. * Ensure that the event subscription callback
will never happen with an invalid peer pointer by making the
event callback removal the first thing in the peer destructor
callback. (closes issue ASTERISK-18747) Reported by: Gregory
Hinton Nietsky Review: https://reviewboard.asterisk.org/r/1564/
2011-11-03 20:26 +0000 [r343375] Walter Doekes <walter+asterisk@wjd.nu>
* res/res_config_sqlite.c: Fix sqlite config driver segfault and
broken queries The sqlite realtime handler assumed you had a
static config configured as well. The realtime multientry handler
assumed that you weren't using dynamic realtime. (closes issue
ASTERISK-18354) (closes issue ASTERISK-18355) Review:
https://reviewboard.asterisk.org/r/1561
2011-11-03 19:56 +0000 [r343336] Richard Mudgett <rmudgett@digium.com>
* funcs/func_dialgroup.c: Remove invalid flag given to iterator in
func_dialgroup.c
2011-11-03 16:15 +0000 [r343281] Alexandr Anikin <may@telecom-service.ru>
* addons/ooh323c/src/printHandler.c, addons/ooh323c/src/ooq931.c,
addons/ooh323c/src/ooGkClient.c, addons/ooh323c/src/ooTimer.c,
addons/ooh323c/src/dlist.c, addons/ooh323c/src/dlist.h: Final fix
memleaks in GkClient codes, same for Timer codes. (these memleaks
stop development of gk codes, now i can continue) Fix
printHandler 'Unbalanced Structure' issues with locking
printHandler data for single thread.
2011-11-03 15:33 +0000 [r343220-343276] Terry Wilson <twilson@digium.com>
* channels/sip/include/sip.h: Make room for the fax detect flags
The original REGISTERTRYING flag, in addition to being impossible
to check, also encroached on the space for the flag above it.
This patch moves the flags that were below REGISTERTRYING back to
where they were as though we had just removed the REGISTERTRYING
option.
* channels/sip/include/sip.h, contrib/realtime/mysql/sippeers.sql,
channels/chan_sip.c: Remove registertrying option in chan_sip
This option is not only useless, but has been broken since
inception since the flag was never copied from the peer where it
is set to the pvt where it was checked. RFC 3261 specificially
states that you should not send a provisional response to a
non-INVITE request, and if we did fix the code so that it worked,
it would cause the same kind of user enumeration vulnerability
that we've discussed with the nat= setting. This patch removes
registertrying option and any code that would have sent a 100
response to a register. Review:
https://reviewboard.asterisk.org/r/1562/
2011-11-02 22:21 +0000 [r343157-343181] Walter Doekes <walter+asterisk@wjd.nu>
* channels/chan_sip.c: Fix improper warning introduced by r342927
and more tweaks Changeset r342927 introduced a warning which was
only supposed to be emitted when a found realtime peer had an
empty (or no) name. It turned out that there were some
inconsistencies left. Now found peers with an empty name are
explicitly ignored like before r342927 but better. Reviewed by:
Stefan Schmidts, Terry Wilson Review:
https://reviewboard.asterisk.org/r/1560
* main/utils.c, include/asterisk/stringfields.h,
include/asterisk/utils.h: Ensure that string field lengths are
properly aligned Integers should always be aligned. For some
platforms (ARM, SPARC) this is more important than for others.
This changeset ensures that the string field string lengths are
aligned on *all* platforms, not just on the SPARC for which there
was a workaround. It also fixes that the length integer can be
resized to 32 bits without problems if needed. (closes issue
ASTERISK-17310) Reported by: radael, S Adrian Reviewed by:
Tzafrir Cohen, Terry Wilson Tested by: S Adrian Review:
https://reviewboard.asterisk.org/r/1549
2011-11-02 19:32 +0000 [r343047-343102] Leif Madsen <lmadsen@digium.com>
* apps/app_authenticate.c: Add note about how Authenticate()
application with option 'd' works. (closes issue ASTERISK-17422)
Reported by: Leif Madsen
* configs/queues.conf.sample: Update documentation for leastrecent
strategy. In queues.conf.sample the leastrecent strategy was
incorrectly described. Now updated to reflect how the strategy
actually checks peers. (closes issue ASTERISK-17854) Reported by:
Sebastian Denz Patches: queues.conf-doc_issue.patch (License
#6139)
2011-11-02 13:44 +0000 [r342990] Kevin P. Fleming <kpfleming@digium.com>
* apps/app_meetme.c: Modify comments in MeetMe application
documentation about DAHDI. The MeetMe application documentation
has some comments about usage of DAHDI, and they were a bit
outdated relative to modern DAHDI releases. This patch changes
the comment to just tell the user that a functional DAHDI timing
source is required, and no longer mention 'dahdi_dummy', since
that module does not exist in current DAHDI releases.
2011-11-01 20:53 +0000 [r342869-342927] Walter Doekes <walter+asterisk@wjd.nu>
* main/config.c, channels/chan_sip.c,
configs/extconfig.conf.sample, include/asterisk/config.h: Several
fixes to the chan_sip dynamic realtime peer/user lookup There
were several problems with the dynamic realtime peer/user lookup
code. The lookup logic had become rather hard to read due to lots
of incremental changes to the realtime_peer function. And, during
the addition of the sipregs functionality, several possibilities
for memory leaks had been introduced. The insecure=port matching
has always been broken for anyone using the sipregs family. And,
related, the broken implementation forced those using sipregs to
*still* have an ipaddr column on their sippeers table. Thanks
Terry Wilson for comprehensive testing and finding and fixing
unexpected behaviour from the multientry realtime call which
caused the realtime_peer to have a completely unused code path.
This changeset fixes the leaks, the lookup inconsistenties and
that you won't need an ipaddr column on your sippeers table
anymore (when you're using sipregs). Beware that when you're
using sipregs, peers with insecure=port will now start matching!
(closes issue ASTERISK-17792) (closes issue ASTERISK-18356)
Reported by: marcelloceschia, Walter Doekes Reviewed by: Terry
Wilson Review: https://reviewboard.asterisk.org/r/1395
* UPGRADE.txt, configs/res_ldap.conf.sample, res/res_realtime.c,
configs/dbsep.conf.sample, main/config.c,
contrib/realtime/mysql/sipfriends.sql (removed),
contrib/realtime/mysql/sippeers.sql (added),
configs/res_config_mysql.conf.sample,
configs/extconfig.conf.sample: Cleanup references to sipusers and
sipfriends dynamic realtime families Somewhere between 1.4 and
1.8 the sipusers family has become completely unused. Before
that, the sipfriends family had been obsoleted in favor of
separate sipusers and sippeers families. Apparently, they have
been merged back again into a single family which is now called
"sippeers". Reviewed by: irroot, oej, pabelanger Review:
https://reviewboard.asterisk.org/r/1523
2011-10-31 15:58 +0000 [r342769] Matthew Jordan <mjordan@digium.com>
* channels/chan_iax2.c, main/pbx.c: Fixed invalid memory access
when adding extension to pattern match tree When an extension is
removed from a context, its entry in the pattern match tree is
not deleted. Instead, the extension is marked as deleted. When an
extension is removed and re-added, if that extension is also a
prefix of another extension, several log messages would report an
error and did not check whether or not the extension was deleted
before accessing the memory. Additionally, if the extension was
already in the tree but previously deleted, and the pattern was
at the end of a match, the findonly flag was not honored and the
extension would be erroneously undeleted. Additionaly, it was
discovered that an IAX2 peer could be unregistered via the CLI,
while at the same time it could be scheduled for unregistration
by Asterisk. The unregistration method now checks to see if the
peer was already unregistered before continuing with an
unregistration. (closes issue ASTERISK-18135) Reported by: Jaco
Kroon, Henry Fernandes, Kristijan Vrban Tested by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/1526
2011-10-29 04:19 +0000 [r342661] Richard Mudgett <rmudgett@digium.com>
* tests/test_linkedlists.c, include/asterisk/linkedlists.h: Fix
AST_LIST_INSERT_BEFORE_CURRENT() updating the wrong variable.
AST_LIST_INSERT_BEFORE_CURRENT() could not be used twice in an
iteration or before AST_LIST_REMOVE_CURRENT() without corrupting
the list. AST_LIST_INSERT_BEFORE_CURRENT() could also corrupt the
list if AST_LIST_INSERT_BEFORE_CURRENT() or
AST_LIST_REMOVE_CURRENT() is used on the next iteration. * Fixed
cut and paste error using the wrong variable in
AST_LIST_INSERT_BEFORE_CURRENT(). * Added linked list unit tests
for AST_LIST_INSERT_BEFORE_CURRENT(), AST_LIST_APPEND_LIST(), and
AST_LIST_INSERT_LIST_AFTER().
2011-10-27 19:34 +0000 [r342545-342602] Jonathan Rose <jrose@digium.com>
* res/res_rtp_multicast.c: Fix sequence number overflow over 16
bits causing codec change in RTP packets. Sequence number was
handled as an unsigned integer (usually 32 bits I think, more
depending on the architecture) and was put into the rtp packet
which is basically just a bunch of bits using an or operation.
Sequence number only has 16 bits allocated to it in an RTP packet
anyway, so it would add to the next field which just happened to
be the codec. This makes sure the sequence number is set to be a
16 bit integer regardless of architecture (hopefully) and also
makes it so the incrementing of the sequence number does bitwise
or at the peak of a 16 bit number so that the value will be set
back to 0 when going beyond 65535 anyway. (closes issue
ASTERISK-18291) Reported by: Will Schick Review:
https://reviewboard.asterisk.org/r/1542/
* res/res_jabber.c: Cleanup reference leaks in res_jabber
res_jabber.c had a number of places where astobjs would be
referenced and have their reference counts bumped without having
a dereference made before the object lost scope. This patch adds
a number of ASTOBJ_UNREFs to resolve that. Review:
https://reviewboard.asterisk.org/r/1478/
2011-10-25 22:04 +0000 [r342484-342487] Richard Mudgett <rmudgett@digium.com>
* main/astobj2.c: Check fopen return value for ao2 reference debug
output. Reported by: wdoekes Patched by: wdoekes Review:
https://reviewboard.asterisk.org/r/1539/
* channels/sig_pri.c: Change D-channel warning to be less confusing
on non-NFAS setups. The "No D-channels available! Using Primary
channel as D-channel anyway!" WARNING message has been confusing
on non-NFAS setups. The message refers to things that are NFAS
specific. * Changed the warning to several different warnings to
be more accurate for the situation and less confusing as a
result: "No D-channels up! Switching selected D-channel from X to
Y.", "No D-channels up!", and "D-channel is down!".
2011-10-25 21:08 +0000 [r342380-342435] Terry Wilson <twilson@digium.com>
* apps/app_queue.c: Use int for storing ao2_container_count instad
of size_t AST-676
* apps/app_queue.c: Simplify queue membercount code Despite an
ominous sounding comment stating that membercount was for "logged
in" members only and thus we couldn't use ao2_container_count(),
I could not find a single place in the code where that seemed to
be accurate. The only time we decremented membercount was when we
were marking something dead or actually removing it. The only
places we incremented it were either after ao2_link(), or trying
to correct for having set it to 0 during a reload. In every case
where we were correcting the value, it seemed that we were trying
to make the count actually match what ao2_container_count() would
return. The only place I could find where we made a determination
about something being "logged in" or not, we didn't trust the
membercount, but instead looked at devicestate, paused, etc. This
patch removes membercount, replaces its use with
ao2_container_count, and manually adds the results of
ao2_container_count to a "membercount" field for ast_data queue
query results. This patch also would fix AST-676, but as it is
slightly riskier than the previously committed fix, the two
commits have been made separately. Reivew:
https://reviewboard.asterisk.org/r/1541/
* apps/app_queue.c: Properly update membercount for reloaded
members Since q->membercount is set to 0 before reloading, it is
important to increment it again for reloaded members as well as
added. (closes issue AST-676) Review:
https://reviewboard.asterisk.org/r/1541/
2011-10-25 19:08 +0000 [r342276-342328] Kinsey Moore <kmoore@digium.com>
* pbx/pbx_spool.c: Fix compilation on Snow Leopard/FreeBSD for
pbx_spool.c One of the changes in the recent spool handling of
hardlinks patch was just outside a HAVE_INOTIFY block and caused
compilation to fail in some build environments. This has been
corrected.
* pbx/pbx_spool.c: Fix spool handling to allow call files to be
hardlinked into place This fixes the inotify code to handle call
files being hardlinked into the spool directory. The smsq utility
does this, instead of rename(), to ensure that it cannot
accidentally overwrite an existing spool file. A rename() might
do that, but link() will definitely not. The inotify code had
broken this, because it would wait for an IN_CLOSE_WRITE event on
the file... which was never forthcoming, since it was never
opened. Now we look for IN_OPEN events following the IN_CREATE
event, and only wait for an IN_CLOSE_WRITE if the file was
actually opened. Patch-by: dwmw2 (closes issue ASTERISK-18331)
Review: https://reviewboard.asterisk.org/r/1391/
2011-10-25 01:23 +0000 [r342223] Terry Wilson <twilson@digium.com>
* main/config.c, include/asterisk/config.h: Return NULL when no
results returned for realtime_multientry It was not documented
what the return value should be when no entries were returned
with the multientry realtime callback. This change forces
consistent behavior even if the backends return an empty
ast_config. Review: https://reviewboard.asterisk.org/r/1521/
2011-10-24 19:49 +0000 [r342061] Jonathan Rose <jrose@digium.com>
* channels/chan_sip.c: Outbound SIP OPTIONS messages will now
include fromuser of related peer. This behavior matches up more
closely with the way invite/register/etc are handled. This patch
also modifies some adjacent code for code style compliance.
Pretty minor. (closes issue ASTERISK-17616) Reported by: Jeremy
Kister Patches: chan_sip.c-options-fromuser-fix-v1.patch uploaded
by Jeremy Kister (license #6232)
2011-10-23 11:36 +0000 [r341906-341921] Gregory Nietsky <gregory@distrotech.co.za>
* apps/app_queue.c: Revert Janitor patch 341906 For now
* apps/app_queue.c: Whitespace Fixups / Add Braces This janitorial
patch is related to work on RB1538
2011-10-21 16:41 +0000 [r341806-341809] Matthew Nicholson <mnicholson@digium.com>
* pbx/pbx_lua.c: only process args that exist ASTERISK-18395
* pbx/pbx_lua.c: don't limit the length of app and function
arguments ASTERISK-18395
2011-10-20 21:54 +0000 [r341717] Richard Mudgett <rmudgett@digium.com>
* include/asterisk/features.h, main/features.c, res/res_agi.c: Fix
AGI exec Park to honor the Park application parameters. The fix
for ASTERISK-12715 and ASTERISK-12685 added a check for the Park
application because the channel needed to be masqueraded to
prevent a crash. Since the Park application now always
masquerades the channel into the parking lot, the special check
is no longer needed. The fix also resulted in AGI exec Park
attempting to double park the call and not honor the Park
application parameters. * Removed no longer necessary call to
ast_masq_park_call() by AGI exec for the Park application.
(Reverts -r146923) * Fix Park application to only return 0 or -1.
The AGI exec Park was causing broken pipe error messages because
the Park application returned 1 on successful park. (closes issue
ASTERISK-18737)
2011-10-20 21:26 +0000 [r341664-341704] Paul Belanger <pabelanger@digium.com>
* funcs/func_callerid.c: Fixed typo from previous commit
* funcs/func_callerid.c: Updated documentation for the optional CID
parameter with CALLERID
2011-10-20 15:11 +0000 [r341529] Terry Wilson <twilson@digium.com>
* include/asterisk/strings.h: Clean up ast_check_digits The code
was originally copied from the is_int() function in the AEL code.
wdoekes pointed out that the function should take a const char*
and that their was an unneeded variable. This is now fixed.
2011-10-19 18:59 +0000 [r341435] Paul Belanger <pabelanger@digium.com>
* channels/chan_gtalk.c: Outgoing calls with Google Voice Google
has recently make some changes (again) to their protocol. Rather
then patching asterisk to flip between the two different methods,
we now allow both. Lets hope this keeps Google Voice happy for a
while. (closes issue ASTERISK-18714) Reported by: Iordan Iordanov
Patches: chan_gtalk.patch uploaded by Iordan Iordanov (licenses
6311)
2011-10-19 07:38 +0000 [r341379] Terry Wilson <twilson@digium.com>
* include/asterisk/strings.h, channels/chan_sip.c: Don't use
is_int() since it doesn't link well on all platforms Just create
an normal API function in strings.h that does the same thing just
to be safe. ASTERISK-17146
2011-10-19 07:15 +0000 [r341366] Stefan Schmidt <sst@sil.at>
* channels/chan_sip.c: Don't sent in-dialog requests like UPDATE
when Asterisk has not yet received a Contact URI from a UAS
2011-10-18 23:37 +0000 [r341314] Terry Wilson <twilson@digium.com>
* channels/chan_sip.c: Don't resolve numeric hosts or contact
unresolved hosts If a SIP dial string contains a numeric hostname
that is not a peer name, don't try to resolve it as it is
unlikely that someone really means Dial(SIP/0.0.4.26) when
Dial(SIP/1050) is called. Also, make sure that create_addr
returns -1 if an address isn't resolved so that we don't attempt
to send SIP requests to an address that doesn't resolve. (closes
issue ASTERISK-17146, ASTERISK-17716) Review:
https://reviewboard.asterisk.org/r/1532/
2011-10-18 23:20 +0000 [r341312] Alexandr Anikin <may@telecom-service.ru>
* addons/chan_ooh323.c: fix issue on channel numbering (calls could
have same channel number on heavy loaded system)
2011-10-18 21:03 +0000 [r341254] Richard Mudgett <rmudgett@digium.com>
* channels/chan_iax2.c, channels/sip/include/sip.h,
channels/chan_mgcp.c, include/asterisk/features.h,
channels/chan_dahdi.c, channels/sig_analog.c,
channels/chan_sip.c, main/features.c: More parking issues. * Fix
potential deadlocks in SIP and IAX blind transfer to parking. *
Fix SIP, IAX, DAHDI analog, and MGCP channel drivers to respect
the parkext_exclusive option with transfers
(Park(,,,,,exclusive_lot) parameter). Created
ast_park_call_exten() and ast_masq_park_call_exten() to maintian
API compatibility. * Made masq_park_call() handle a failed
ast_channel_masquerade() setup. * Reduced excessive struct
parkeduser.peername[] size.
2011-10-17 17:35 +0000 [r341189] Terry Wilson <twilson@digium.com>
* channels/chan_sip.c: Initialize variables before calling
parse_uri If parse_uri was called with an empty URI, some
pointers would be modified and an invalid read could result. This
patch avoids calling parse_uri with an empty contact uri when
parsing REGISTER requests. AST-2011-012 (closes issue
ASTERISK-18668)
2011-10-17 16:23 +0000 [r341108-341112] Paul Belanger <pabelanger@digium.com>
* apps/app_voicemail.c: Fix previous commit
* apps/app_voicemail.c: Voicemail compiler flags are 'core' support
2011-10-17 15:35 +0000 [r341088] Terry Wilson <twilson@digium.com>
* channels/chan_sip.c: Don't try to remove peers without IPs from
peers_by_ip (closes issue ASTERISK-18696)
2011-10-17 15:08 +0000 [r341074] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
* pbx/pbx_realtime.c: Remove an unused include of md5.h Unused
include of asterisk/md5.h in pbx_realtime.c . A commit needed to
test the commit message.
2011-10-14 21:36 +0000 [r341022] Kevin P. Fleming <kpfleming@digium.com>
* build_tools/embed_modules.xml, Makefile.moddir_rules: Change the
internal name of the menuselect options that are used to control
whether modules are embedded or not; using just the bare category
name led to accidentally enabling these options when users used
the wrong "--enable" operation on the menuselect command line.
Now the internal option names are prefixed with "EMBED_", so they
won't be the same as the name of the category containing the
modules they control the embedding of.
2011-10-14 20:49 +0000 [r340970] Kinsey Moore <kmoore@digium.com>
* res/res_rtp_asterisk.c, channels/chan_sip.c: Quiet RTCP Receiver
Reports during fax transmission RTCP is now disabled for
"inactive" RTP audio streams during SIP T.38 sessions. The
ability to disable RTCP streams in res_rtp_asterisk was missing,
so this code was added to support the bug fix. (closes issue
ASTERISK-18400)
2011-10-14 16:33 +0000 [r340878] Terry Wilson <twilson@digium.com>
* main/channel.c: Avoid unnecessary WARNING message Add
AST_CONTROL_UPDATE_RTP_PEER frame to be ignored here to avoid
displaying a WARNING message. (closes issue ASTERISK-18610) Patch
by: Kristijan_Vrban
2011-10-14 15:58 +0000 [r340863] Jonathan Rose <jrose@digium.com>
* codecs/codec_dahdi.c, apps/app_system.c, res/res_curl.c,
funcs/func_realtime.c, build_tools/cflags.xml, utils/utils.xml,
res/res_fax.c, apps/app_celgenuserevent.c: Fixes some support
level info so that it can be read by menuselect. (issue
ASTERISK-18268) Review: https://reviewboard.asterisk.org/r/1525/
2011-10-13 22:48 +0000 [r340809] Richard Mudgett <rmudgett@digium.com>
* main/features.c: Fix DTMF blind transfer continuing to execute
dialplan after transfer. Party A calls Party B. Party A DTMF
blind transfers Party B to Party C. Party A channel continues to
execute dialplan. * Fixed the return value of
builtin_blindtransfer() to return the correct value after a
transfer so the dialplan will not keep executing. * Removed
unnecessary connected line update that did not really do
anything. * Made access to GOTO_ON_BLINDXFR thread safe in
check_goto_on_transfer(). * Fixed leak of xferchan for failure
cases in check_goto_on_transfer(). * Updated debug messages in
builtin_blindtransfer() and check_goto_on_transfer(). (closes
issue ASTERISK-18275) Reported by: rmudgett Tested by: rmudgett
2011-10-13 06:58 +0000 [r340717] Stefan Schmidt <sst@sil.at>
* channels/chan_sip.c: storing the route-set also on a 181 response
not only on 180,182 or 183.
2011-10-13 06:52 +0000 [r340662-340715] Terry Wilson <twilson@digium.com>
* channels/chan_sip.c: Initialize ast_sockaddr before calling
ast_sockaddr_resolve Avoid possible jump based on unitialized
value
* res/res_config_sqlite.c: Don't skip the query field on a realtime
multi query There is no documented reason to not add the query
field to the varlist returned by a realtime multi query, despite
the config category being set to its value. Of course, there is
no documentation that the category should be set to the value
either. There is lots of no documentation when it comes to
realtime. But, other engines do not skip this field so I am
forcing this backend to follow the convention, because not doing
so is very silly.
2011-10-12 20:30 +0000 [r340576] Stefan Schmidt <sst@sil.at>
* channels/chan_sip.c: Store route-set from provisional SIP
responses so early-dialog requests can be routed properly
2011-10-12 20:19 +0000 [r340534] Terry Wilson <twilson@digium.com>
* channels/chan_sip.c: Update SIP realtime fullcontact regardless
of caching We should update the fullcontact field in the realtime
table whether or not rtcachefriends is set. There is no reason to
treat a non-cached realtime entity differently than a cached in
this regard. (closes issue ASTERISK-18446) Reported by: wdoekes
2011-10-12 20:07 +0000 [r340470-340522] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c: Initialize the PRI channel alarms properly
on startup. The PRI channel alarms were initialized with an
inverted sense. (closes issue ASTERISK-18710) Reported by:
Tzafrir Cohen
* apps/app_meetme.c: Update MeetMe p and X option documentation
when interacting with the s option. ASTERISK-12175 changed the p
and X options to not interfere with the s option when they are
used together. It makes more sense for the s option to have
priority for the DTMF '*' key since it cannot change its
activation code. Otherwise, you could not use option s with the p
or X options. JIRA AST-671
2011-10-12 16:27 +0000 [r340418] Paul Belanger <pabelanger@digium.com>
* channels/chan_sip.c: Fix verbose messages when IPv6 logic was
added (closes issue ASTERISK-18612) Reported by: Tim Osman
2011-10-11 21:03 +0000 [r340279-340365] Richard Mudgett <rmudgett@digium.com>
* channels/sig_ss7.c, channels/chan_dahdi.c, channels/sig_ss7.h:
Add protection for SS7 channel allocation and better glare
handling. * Added a CLI "ss7 show channels" command that might
prove useful for future debugging. * Made the incoming SS7
channel event check and gripe message uniform. * Made sure that
the DNID string for an incoming call is always initialized.
(issue ASTERISK-17966) Reported by: Kenneth Van Velthoven
Patches: jira_asterisk_17966_v1.8_glare.patch (license #5621)
patch uploaded by rmudgett
* channels/sip/include/dialog.h, channels/chan_sip.c: Fix some
potential deadlocks pointed out by helgrind. * Fixed deadlock
potential calling dialog_unlink_all() in __sip_autodestruct().
Found by helgrind. * Fixed deadlock potential in
handle_request_invite() after calling sip_new(). Found by
helgrind. * The sip_new() function now returns with the created
channel already locked. * Removed the dead code that starts a PBX
in in sip_new(). No sip_new() callers caused that code to be
executed and it was a bad thing to do anyway. * Removed unused
parameters and return value from dialog_unlink_all(). * Made
dialog_unlink_all() and __sip_autodestruct() safely obtain the
owner and private channel locks without a deadlock avoidance
loop.
* include/asterisk/manager.h, main/manager.c: Convert registered
AMI actions to ao2 objects. * Fixed race between calling an AMI
action callback and unregistering that action. Refixes
ASTERISK-13784 broken by ASTERISK-17785 change. * Fixed potential
memory leak if an AMI action failed to get registered because is
already was registered. Part of the ao2 conversion. * Fixed AMI
ListCommands action not walking the actions list with a lock
held. * Fix usage of ast_strdupa() and alloca() in loops. Excess
stack usage. * Fix AMI Originate action Variable header requiring
a space after the header colon. Reported by Yaroslav Panych on
the asterisk-dev list. * Increased the number of listed variables
allowed per AMI Originate action Variable header to 64. * Fixed
AMI GetConfigJSON action output format. * Fixed usage of res
contents outside of scope in append_channel_vars(). * Fixed
inconsistency of config file channelvars option. The values no
longer accumulate with every channelvars option in the config
file. Only the last value is kept to be consistent with the CLI
"manager show settings" command. (closes issue ASTERISK-18479)
Reported by: Jaco Kroon
2011-10-11 00:43 +0000 [r340263] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
* include/asterisk/sha1.h, main/channel.c, main/sha1.c: Update SHA1
code to RFC 6234 RFC 6234 is an update to RFC 3174 from which the
code was originally taken. It has a slightly better code, and a
better phrased license (simple 3-clause BSD). * main/sha1.c is
sha1.c from RFC 6234 with formatting changes only. *
include/asterisk/sha1.h merges sha.h and sha-private.h from RFC
6234. * Removed unused include of asterisk/sha1.h from
main/channels.c Review: https://reviewboard.asterisk.org/r/1503/
2011-10-10 20:23 +0000 [r340164] Matthew Jordan <mjordan@digium.com>
* channels/chan_sip.c: Updated chan_sip to place calls on hold if
SDP address in INVITE is ANY This patch fixes the case where an
INVITE is received with c=0.0.0.0 or ::. In this case, the call
should be placed on hold. Previously, we checked for the address
being null; this patch keeps that behavior but also checks for
the ANY IP addresses. Review:
https://reviewboard.asterisk.org/r/1504/ (closes issue
ASTERISK-18086) Reported by: James Bottomley Tested by: Matt
Jordan
2011-10-10 14:14 +0000 [r340108] Matthew Nicholson <mnicholson@digium.com>
* doc/appdocsxml.dtd, main/loader.c, main/xmldoc.c, main/pbx.c,
main/manager.c, res/res_fax.c, apps/app_fax.c,
include/asterisk/module.h, res/res_agi.c,
include/asterisk/xmldoc.h: Load the proper XML documentation when
multiple modules document the same application. This patch adds
an optional "module" attribute to the XML documentation spec that
allows the documentation processor to match apps with identical
names from different modules to their documentation. This patch
also fixes a number of bugs with the documentation processor and
should make it a little more efficient. Support for multiple
languages has also been properly implemented. ASTERISK-18130
Review: https://reviewboard.asterisk.org/r/1485/
2011-10-09 01:16 +0000 [r339830-339938] Igor Goncharovskiy <igor.goncharovsky@gmail.com>
* channels/chan_unistim.c: Fix compilation issue, caused by missed
session structure (closes issue ASTERISK-18694) Reported by:
alex70
* channels/chan_unistim.c: Fix segfault in Unistim channel (closes
issue ASTERISK-18638) Reported by: jonnt
* channels/chan_unistim.c: Fix char array cast as short array in
send_client() function (for ARM platform) (closes issue
ASTERISK-17314) Reported by: jjoshua
2011-10-07 19:34 +0000 [r339625-339776] Richard Mudgett <rmudgett@digium.com>
* apps/app_url.c: Initialize option flags for SendURL application.
(closes issue ASTERISK-18574) Reported by: marcelloceschia
* autoconf/ast_ext_lib.m4, configure,
include/asterisk/autoconfig.h.in, configure.ac: Fix regression in
configure script for libpri capability checks. JIRA AST-598 added
the PRI_L2_PERSISTENCE option to fix BRI PTMP TE layer 2
persistence issues with some telcos. ASTERISK-18535 attempted to
fix the unexpected requirement that libpri *must* have that
feature to work with Asterisk. The AST_EXT_LIB_SETUP_DEPENDENT
lines made the PRI optional features required. Unfortunately, I
thought AST_EXT_LIB_SETUP_DEPENDENT didn't do anything useful for
libpri and deleted those lines for libpri. The result was the
HAVE_PRI_xxx defines that control the ability to use optional
libpri features were also deleted. * Created
AST_EXT_LIB_SETUP_OPTIONAL configuration macro to allow optional
features in a library that the source code could take advantage
of if the code supports the feature. (closes issue
ASTERISK-18687) Reported by: Norbert Tested by: rmudgett
* main/udptl.c, channels/chan_sip.c: Fix debugging messages
generated by 'udptl debug'. * Makes chan_sip set the tag to the
channel name. * Fixes received debug message sequence number. *
Removed tx/rx debug message type since it was hard coded to 0. *
Made udptl.c logged message header consistent if possible: "UDPTL
(%s): ". * Removed unused rx_expected_seq_no from struct
ast_udptl. (closes issue ASTERISK-18401) Reported by: Kevin P.
Fleming Patches: jira_asterisk_18401_v1.8.patch (license #5621)
patch uploaded by rmudgett Tested by: Matthew Nicholson
2011-10-05 21:30 +0000 [r339566] Leif Madsen <lmadsen@digium.com>
* build_tools/prep_tarball: Update prep_tarball script to download
pre-exported documentation. I've updated the prep_tarball script
to now download the pre-exported documentation from the Asterisk
wiki. This will give us more control over what is being included
in the tarball releases, and will make both the PDF and HTML
exported documentation look much better (especially when viewing
from a console). (Closes issue ASTERISK-18677)
2011-12-15 Asterisk Development Team <asteriskteam@digium.com>
* Asterisk 1.8.8.0 Released.
2011-12-09 Asterisk Development Team <asteriskteam@digium.com>
* Asterisk 1.8.8.0-rc5 Released.
* Fixed crash from orphaned MWI subscriptions in chan_sip
This patch resolves the issue where MWI subscriptions are orphaned
by subsequent SIP SUBSCRIBE messages. When a peer is removed, either
by pruning realtime SIP peers or by unloading / loading chan_sip, the
MWI subscriptions that were orphaned would still be on the event engine
list of valid subscriptions but have a pointer to a peer that no longer
was valid. When an MWI event would occur, this would cause a seg fault.
(closes issue ASTERISK-18663)
Review: https://reviewboard.asterisk.org/r/1610/
* Don't crash on INFO automon request with no channel
AST-2011-014. When automon was enabled in features.conf, it was possible
to crash Asterisk by sending an INFO request if no channel had been
created yet.
(closes issue ASTERISK-18805)
* Default to nat=yes; warn when nat in general and peer differ
AST-2011-013. It is possible to enumerate SIP usernames when the general and
user/peer nat settings differ in whether to respond to the port a request is
sent from or the port listed for responses in the Via header. In 1.4 and
1.6.2, this would mean if one setting was nat=yes or nat=route and the other
was either nat=no or nat=never. In 1.8 and 10, this would mean when one
was nat=force_rport and the other was nat=no.
In order to address this problem, it was decided to switch the default
behavior to nat=yes/force_rport as it is the most commonly used option
and to strongly discourage setting nat per-peer/user when at all
possible.
For more discussion of the issue, please see:
http://lists.digium.com/pipermail/asterisk-dev/2011-November/052191.html
(closes issue ASTERISK-18862)
Review: https://reviewboard.asterisk.org/r/1591/
2011-11-16 Asterisk Development Team <asteriskteam@digium.com>
* Asterisk 1.8.8.0-rc4 Released.
* Ensure that a null vmexten does not cause a segfault.
When sip_send_mwi_to_peer was modified recently to avoid deadlocks,
vmexten was not expected to be null. This change handles that
situation to avoid a segfault.
(closes issue ASTERISK-18663)
2011-11-09 Asterisk Development Team <asteriskteam@digium.com>
* Asterisk 1.8.8.0-rc3 Released.
* Prevent BLF subscriptions from causing deadlocks
Fix a locking inversion in sip_send_mwi_to_peer that was causing
deadlocks.
This function now requires that both the peer and associated pvt be
unlocked
before it is called for cases where peer and peer->mwipvt form a
circular
reference.
(closes issue ASTERISK-18663)
Review: https://reviewboard.asterisk.org/r/1563/
* Fix deadlock if peer is destroyed while sending MWI notice.
A dialog cannot be destroyed by the ao2_callback dialog_needdestroy
because of a deadlock between the dialogs container lock and the
RWLOCK of the events subscription list.
* Create dialogs_to_destroy container to hold dialogs that will be
destroyed.
* Ensure that the event subscription callback will never happen with
an invalid peer pointer by making the event callback removal the first
thing in the peer destructor callback.
(closes issue ASTERISK-18747)
Reported by: Gregory Hinton Nietsky
Review: https://reviewboard.asterisk.org/r/1564/
* Fix issue with setting defaultenabled on categories that are already
enabled by default.
(closes issue ASTERISK-18738)
Reported by: Paul Belanger
2011-10-18 Asterisk Development Team <asteriskteam@digium.com>
* Asterisk 1.8.8.0-rc2 Released.
* AST-2011-012
* menuselect/menuselect.c: Fix --enable/--enable-category.
------------------------------------------------------------------------
r339719 | rmudgett | 2011-10-06 17:47:50 -0500 (Thu, 06 Oct 2011) | 20 lines
Fix regression in configure script for libpri capability checks.
JIRA AST-598 added the PRI_L2_PERSISTENCE option to fix BRI PTMP TE layer
2 persistence issues with some telcos. ASTERISK-18535 attempted to fix
the unexpected requirement that libpri *must* have that feature to work
with Asterisk. The AST_EXT_LIB_SETUP_DEPENDENT lines made the PRI
optional features required. Unfortunately, I thought
AST_EXT_LIB_SETUP_DEPENDENT didn't do anything useful for libpri and
deleted those lines for libpri. The result was the HAVE_PRI_xxx defines
that control the ability to use optional libpri features were also
deleted.
* Created AST_EXT_LIB_SETUP_OPTIONAL configuration macro to allow optional
features in a library that the source code could take advantage of if the
code supports the feature.
(closes issue ASTERISK-18687)
Reported by: Norbert
Tested by: rmudgett
------------------------------------------------------------------------
r340878 | twilson | 2011-10-14 11:33:28 -0500 (Fri, 14 Oct 2011) | 8 lines
Avoid unnecessary WARNING message
Add AST_CONTROL_UPDATE_RTP_PEER frame to be ignored here to avoid
displaying a WARNING message.
(closes issue ASTERISK-18610)
Patch by: Kristijan_Vrban
------------------------------------------------------------------------
r341088 | twilson | 2011-10-17 10:35:05 -0500 (Mon, 17 Oct 2011) | 4 lines
Don't try to remove peers without IPs from peers_by_ip
(closes issue ASTERISK-18696)
------------------------------------------------------------------------
2011-10-05 Asterisk Development Team <asteriskteam@digium.com>
* Asterisk 1.8.8.0-rc1 Released.
2011-10-05 21:30 +0000 [r339566] Leif Madsen <lmadsen@digium.com>
* build_tools/prep_tarball: Update prep_tarball script to download
pre-exported documentation. I've updated the prep_tarball script
to now download the pre-exported documentation from the Asterisk
wiki. This will give us more control over what is being included
in the tarball releases, and will make both the PDF and HTML
exported documentation look much better (especially when viewing
from a console). (Closes issue ASTERISK-18677)
2011-10-05 17:01 +0000 [r339506-339511] Richard Mudgett <rmudgett@digium.com>
* apps/app_dial.c: Fix Dial F option notes formatting.
* main/manager.c: Fix XML error in AMI action Challenge.
2011-10-05 16:31 +0000 [r339505] Matthew Nicholson <mnicholson@digium.com>
* res/res_fax.c: The app name in the documentation must match what
we register the application as.
2011-10-05 16:26 +0000 [r339406-339504] Richard Mudgett <rmudgett@digium.com>
* main/manager.c: Add missing documentation of required AMI action
Challenge AuthType header. (closes issue ASTERISK-18554) Reported
by: Vlad Povorozniuc Patches:
__20110919-manager-challenge-docs.patch.txt (license #4999) patch
uploaded by Leif Madsen
* Makefile: Make always create the MOH directory
(/var/lib/asterisk/moh). (closes issue ASTERISK-18409) Reported
by: abelbeck Patches: asterisk-1.8-makefile-moh.patch (license
#5903) patch uploaded by abelbeck Tested by: abelbeck, Michael
Keuter
2011-10-04 19:33 +0000 [r339297-339352] Jonathan Rose <jrose@digium.com>
* main/say.c: Removes improper use of sound 'and' in German
language mode from application saynumber Asterisk would say 'Five
hundert und sechs und zwanzig' instead of 'Five hundert sechs und
zwanzig'... which is both weird sounding and wrong. This patch
makes sure Asterisk will only say the 'and' word between the
single digit and double digit places. (closes issue
ASTERISK-18212) Reported By: Lionel Elie Mamane Patches:
upstream_germand_no_and.diff (License #5402) uploaded by Lionel
Elie Mamane
* res/res_jabber.c: Reverting revision 333265 due to component
connection problems it introduces. I'm going to attempt some
generic res_jabber cleanup and come up with a new fix for this
problem, but first it seems prudent to remove this rather broad
attempt to fix it and instead approach this problem either from
the same angle but looking only at canceling (or possibly
rescheduling) the send when we absolutely know it will cause a
segfault or, if that can't be easily accomplished, strictly from
the devstate side of things. Also, I'm pretty sure a lot of the
code in res_jabber isn't thread safe. (issue ASTERISK-18626)
(issue ASTERISK-18078)
2011-10-04 11:44 +0000 [r339244] Alexandr Anikin <may@telecom-service.ru>
* addons/ooh323c/src/memheap.c: fix forget declaration in previous
change
2011-10-03 20:12 +0000 [r339144-339147] Leif Madsen <lmadsen@digium.com>
* channels/chan_sip.c: Remove duplicated Maxforwards line in AMI
output. (Closes issue ASTERISK-18637) Reported by: Jacek
Konieczny Patches: asterisk-sipshowpeer.patch (License #6298)
uploaded by Jacek Konieczny
* apps/app_dial.c: Make documentation for Dial() options 'F' and
'F()' more clear. (Closes issue ASTERISK-18646) Reported by:
Physis Heckman Tested by: Richard Mudgett
2011-10-03 18:42 +0000 [r339087] Alexandr Anikin <may@telecom-service.ru>
* addons/ooh323c/src/memheap.c: destroy memheap mutex properly
before memheap deleted (fix memory leak occured after r304950
changes with DEBUG_THREAD compile option)
2011-10-03 18:40 +0000 [r339086] Terry Wilson <twilson@digium.com>
* channels/chan_sip.c, main/file.c: Properly ignore
AST_CONTROL_UPDATE_RTP_PEER in more places After the change in
r336294, the new AST_CONTROL_UPDATE_RTP_PEER frame is sent when a
re-invite happens. If we receive a re-invite from a device the
waitstream_core was not aware of the new control frame and would
drop the call. (closes issue ASTERISK-18610) Reported by:
Kristijan_Vrban
2011-09-30 22:05 +0000 [r338800] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c: Fix segfault in analog_ss_thread() not
checking ast_read() for NULL. NOTE: The problem was reported
against v1.6.2. It is unlikely to ever happen on v1.8 and above
since chan_dahdi.c:analog_ss_thread() is unlikely to be used. The
version in sig_analog.c has largely replaced it. (closes issue
ASTERISK-18648) Reported by: Stephan Bosch Patches:
jira_asterisk_18648_v1.8.patch (license #5621) patch uploaded by
rmudgett Tested by: Stephan Bosch
2011-09-30 18:54 +0000 [r338718] Jonathan Rose <jrose@digium.com>
* configs/queues.conf.sample: Adds documentation for
QueueMemberStatus event generation
2011-09-30 16:27 +0000 [r338663] Richard Mudgett <rmudgett@digium.com>
* channels/chan_sip.c: Fix formatting of AMI header for SIP show
peer. ASTERISK-17486 exposed the problem for AMI parsers. (closes
issue ASTERISK-18649) Reported by: Jacek Konieczny Patches:
asterisk-sipshowpeer_response_end.patch (license #6298) patch
uploaded by Jacek Konieczny
2011-09-30 09:31 +0000 [r338609] TransNexus OSP Development <support@transnexus.com>
* apps/app_osplookup.c, configure.ac: Remove r338137 and r338138.
2011-09-29 21:12 +0000 [r338555] Paul Belanger <pabelanger@digium.com>
* tests/test_linkedlists.c, tests/test_amihooks.c,
tests/test_security_events.c, tests/test_locale.c,
tests/test_logger.c, tests/test_dlinklists.c: Test modules should
depend on the TEST_FRAMEWORK flag
2011-09-29 20:54 +0000 [r338551] Jason Parker <jparker@digium.com>
* tests/test_db.c, tests/test_netsock2.c: Test modules have a
support level of core.
2011-09-29 18:31 +0000 [r338492] Leif Madsen <lmadsen@digium.com>
* channels/chan_sip.c: Update documentation for SIP_HEADER. The
SIP_HEADER function only works on the the initial SIP INVITE. The
documentation was updated in trunk, but not in 1.8 or 10, so I'm
making them match. (Closes issue ASTERISK-18640)
2011-09-29 12:13 +0000 [r338416] Gregory Nietsky <gregory@distrotech.co.za>
* channels/sip/include/sip.h, channels/chan_sip.c: The rtptimeout
setting is ignored on a per peer basis. Not only is the
rtptimeout ignored in some cases but rtpkeepalive and
rtpholdtimeout is affected. this commit also removes
rtptimeout/rtpholdtimeout on text rtp. (closes issue
ASTERISK-18559) Review: https://reviewboard.asterisk.org/r/1452
2011-09-28 22:35 +0000 [r338235-338322] Richard Mudgett <rmudgett@digium.com>
* channels/sig_pri.c: Make duplicate call ptr warning message more
helpful. * Adds the value of the call ptr to the duplicate call
ptr message to help trace why there is a duplicate call ptr.
* include/asterisk/logger.h: Fix inconsistency in
LOG_VERBOSE/AST_LOG_VERBOSE declaration. (closes issue
ASTERISK-17973) Reported by: Luke H Patches: logger_h.patch
(license #6278) patch uploaded by Luke H
2011-09-28 20:52 +0000 [r338227] Jason Parker <jparker@digium.com>
* tests/test_db.c, tests/test_netsock2.c, build_tools/cflags.xml,
channels/chan_usbradio.c, build_tools/cflags-devmode.xml,
agi/agi.xml, utils/utils.xml, build_tools/embed_modules.xml: Add
support levels to non-module sections of menuselect (cflags,
utils, etc).
2011-09-28 20:24 +0000 [r338224] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c: Fix chan_dahd compiling with gcc 4.6 when
PRI and SS7 not present. (closes issue ASTERISK-18357) Reported
by: Matthew Nicholson
2011-09-28 07:28 +0000 [r338137-338138] TransNexus OSP Development <support@transnexus.com>
* configure.ac: Updated for checking OSP Toolkit version 4.0.0.
* apps/app_osplookup.c: Updated for OSP Toolkit 4.0.0.
2011-09-27 20:10 +0000 [r338084] Paul Belanger <pabelanger@digium.com>
* apps/app_macro.c: Upgrade app_macro to core
2011-09-26 19:30 +0000 [r337973] Richard Mudgett <rmudgett@digium.com>
* include/asterisk/channel.h, main/cel.c, main/manager.c,
funcs/func_odbc.c, cel/cel_custom.c, apps/app_minivm.c,
main/logger.c, cel/cel_sqlite3_custom.c, cdr/cdr_manager.c,
cdr/cdr_custom.c, apps/app_voicemail.c, apps/app_dial.c,
main/pbx.c, cdr/cdr_sqlite3_custom.c, cdr/cdr_syslog.c,
tests/test_gosub.c, include/asterisk/cel.h: Fix deadlock when
using dummy channels. Dummy channels created by
ast_dummy_channel_alloc() should be destoyed by
ast_channel_unref(). Using ast_channel_release() needlessly grabs
the channel container lock and can cause a deadlock as a result.
* Analyzed use of ast_dummy_channel_alloc() and made use
ast_channel_unref() when done with the dummy channel. (Primary
reason for the reported deadlock.) * Made
app_dial.c:dial_exec_full() not call ast_call() holding any
channel locks. Chan_local could not perform deadlock avoidance
correctly. (Potential deadlock exposed by this issue. Secondary
reason for the reported deadlock since the held lock was part of
the deadlock chain.) * Fixed some uses of
ast_dummy_channel_alloc() not checking the returned channel
pointer for failure. * Fixed some potential chan=NULL pointer
usage in func_odbc.c. Protected by testing the bogus_chan value.
* Fixed needlessly clearing a 1024 char auto array when setting
the first char to zero is enough in manager.c:action_getvar().
(closes issue ASTERISK-18613) Reported by: Thomas Arimont
Patches: jira_asterisk_18613_v1.8.patch (license #5621) patch
uploaded by rmudgett Tested by: Thomas Arimont
2011-09-23 19:14 +0000 [r337839-337898] Gregory Nietsky <gregory@distrotech.co.za>
* contrib/init.d/rc.archlinux.asterisk: Spelling fix
* apps/app_queue.c: Make sure a CDR is on the stack for call in the
Queue. Only let update_cdr act on the last CDR in the stack. In
some circumstances [Attended transfer to queue] a CDR record is
not inserted for this call where it should. (closes issue
ASTERISK-18567) Review: https://reviewboard.asterisk.org/r/1266
2011-09-23 00:44 +0000 [r337774] Russell Bryant <russell@digium.com>
* configs/res_pktccops.conf.sample: Comment out entries in sample
res_pktccops.conf. With these options enabled, they can cause
Asterisk to freak out by SYN flooding a network and eating the
CPU. Obviously it would be good to fix the code so that this
can't happen, but we can at least change the default
configuration so it doesn't happen. This was reported downstream
to the Fedora issue tracker:
https://bugzilla.redhat.com/show_bug.cgi?id=658431
2011-09-22 21:29 +0000 [r337720] Richard Mudgett <rmudgett@digium.com>
* channels/sig_pri.c: Made ISDN not add numbering plan prefix
strings to empty numbers. When the Caller-ID is restricted, the
expected behavior is for the Caller-ID to be blank. In
chan_dahdi, the national prefix is placed onto the Caller-ID
number even if it is restricted (empty) causing the Caller-ID to
be the national prefix rather than blank. This behavior was lost
when sig_pri was extracted from chan_dahdi. * Made not add prefix
strings to empty connected line, calling, and ANI number strings.
(closes issue ASTERISK-18577) Reported by: Kris Shaw Patches:
jira_asterisk_18577_v1.8.patch (license #5621) patch uploaded by
rmudgett Tested by: Kris Shaw
2011-09-22 11:39 +0000 [r337430-337541] Gregory Nietsky <gregory@distrotech.co.za>
* res/res_srtp.c: Add warned to ast_srtp to prevent errors on each
frame from libsrtp The first 9 frames are not reported as some
devices dont use srtp from first frame these are suppresed. the
warning is then output only once every 100 frames.
* channels/chan_h323.c: If IP address is used in chan_h323 host
parameter of peer configuration. module tries to resolve IP
address to IP address and fails. Simple fix to set family of
socket this is a hangover from ipv6 changes. (closes issue
ASTERISK-18237) (issue ASTERISK-17278) (issue ASTERISK-17500)
* main/channel.c: Its possible to loose audio on ast_write when the
channel is not transcoded correctly. in the case of DAHDI the
channel is hungup. This patch tries to "fix" the problem and make
the channel compatiable and warn the user of this problem. Please
note there is a underlying problem with codec negotion this does
not fix the problem it does try to rectify it and prevent loss of
service. Review: https://reviewboard.asterisk.org/r/1442/ (closes
issue ASTERISK-17541) (closes issue ASTERISK-18063) (issue
ASTERISK-14384) (issue ASTERISK-17502) (issue ASTERISK-18325)
(issue ASTERISK-18422)
2011-09-21 21:18 +0000 [r337325-337353] Tilghman Lesher <tilghman@meg.abyt.es>
* apps/app_voicemail.c: More silly spacing changes
* apps/app_voicemail.c: Dumb little spacing fix.
* funcs/func_curl.c: Escape commas in keys and values, when keys
and values are enumerated by commas. Review:
https://reviewboard.asterisk.org/r/1433
2011-09-20 22:38 +0000 [r337118] Matthew Jordan <mjordan@digium.com>
* main/app.c, apps/app_followme.c, apps/app_voicemail.c,
apps/app_dial.c, include/asterisk/app.h, apps/app_meetme.c,
apps/app_minivm.c: Fix for incorrect voicemail duration in
external notifications This patch fixes an issue where the
voicemail duration was being reported with a duration
significantly less than the actual sound file duration.
Voicemails that contained mostly silence were reporting the
duration of only the sound in the file, as opposed to the
duration of the file with the silence. This patch fixes this by
having two durations reported in the __ast_play_and_record family
of functions - the sound_duration and the actual duration of the
file. The sound_duration, which is optional, now reports the
duration of the sound in the file, while the actual full duration
of the file is reported in the duration parameter. This allows
the voicemail applications to use the sound_duration for minimum
duration checking, while reporting the full duration to external
parties if the voicemail is kept. (issue ASTERISK-2234) (closes
issue ASTERISK-16981) Reported by: Mary Ciuciu, Byron Clark, Brad
House, Karsten Wemheuer, KevinH Tested by: Matt Jordan Review:
https://reviewboard.asterisk.org/r/1443
2011-09-20 22:18 +0000 [r337115] Leif Madsen <lmadsen@digium.com>
* contrib/init.d/rc.redhat.asterisk: Update RedHat Init script to
work with Heartbeat. The current RedHat init script was not LSB
compatible. This change will make it LSB compatible so that it
can work correctly with Heartbeat. (Closes issue ASTERISK-18253)
Reported by: c0rnoTa
2011-09-20 21:04 +0000 [r337061] Kinsey Moore <kmoore@digium.com>
* tests/test_pbx.c, main/pbx.c: Make CANMATCH with the new pattern
match engine behave more like the old one When checking an
extension for E_CANMATCH using the new extension matching
algorithm, an exact match was not returned as a possible match
resulting in the queue failing to allow a caller to exit on DTMF.
This removes the requirement that an extension be longer than
acquired digits for an E_CANMATCH operation to succeed. (closes
issue ASTERISK-18044) Review:
https://reviewboard.asterisk.org/r/1367/
2011-09-20 19:10 +0000 [r336977-337007] Richard Mudgett <rmudgett@digium.com>
* channels/sig_ss7.c: Check if a channel was created before using
the pointer in sig_ss7_new_ast_channel(). Fixes the crash in
ASTERISK-17955 gdb-11918.txt backtrace. * Added some missing
libss7 access lock protection. * Prevent cancelling the
ss7_linkset() thread at inoportune times just like the
pri_dchannel() thread. (issue ASTERISK-17955) Reported by: Ian M
Sherman Patches: jira_asterisk_17955_v1.8.patch (license #5621)
patch uploaded by rmudgett (attached to related ASTERISK-17966)
* channels/sig_ss7.c: Fix deadlock from not releasing SS7 linkset
lock. sig_ss7_hangup() failed to release the SS7 linkset lock if
the call had the alreadyhungup flag set. * Made unlock the SS7
linkset lock in sig_ss7_hangup() if the alreadyhungup flag is
set. * Made ss7_start_call() not hold any locks while creating
the channel for an incoming call to prevent deadlock. * Made
ss7_grab() a void function, since it could never fail, to
simplify calling code. * Made obtain the channel lock to do
softhangup in some places. Patches: jira_ast_668_v1.8.patch
(license #5621) patch uploaded by rmudgett JIRA AST-668
2011-09-20 00:56 +0000 [r336877] Russell Bryant <russell@digium.com>
* res/res_rtp_asterisk.c: Fix crashes in ast_rtcp_write(). This
patch addresses crashes related to RTCP handling. The backtraces
just show a crash in ast_rtcp_write() where it appears that the
RTP instance is no longer valid. There is a race condition with
scheduled RTCP transmissions and the destruction of the RTP
instance. This patch utilizes the fact that ast_rtp_instance is a
reference counted object and ensures that it will not get
destroyed while a reference is still around due to scheduled RTCP
transmissions. RTCP transmissions are scheduled and executed from
the chan_sip scheduler context. This scheduler context is
processed in the SIP monitor thread. The destruction of an RTP
instance occurs when the associated sip_pvt gets destroyed (which
happens when the sip_pvt reference count reaches 0). However, the
SIP monitor thread is not the only thread that can cause a
sip_pvt to get destroyed. The sip_hangup function, executed from
a channel thread, also decrements the reference count on a
sip_pvt and could cause it to get destroyed. While this is being
changed anyway, the patch also removes calling ast_sched_del()
from within the RTCP scheduler callback. It's not helpful. Simply
returning 0 prevents the callback from being rescheduled. (closes
issue ASTERISK-18570) Related issues that look like they are the
same problem: (issue ASTERISK-17560) (issue ASTERISK-15406)
(issue ASTERISK-15257) (issue ASTERISK-13334) (issue
ASTERISK-9977) (issue ASTERISK-9716) Review:
https://reviewboard.asterisk.org/r/1444/
2011-09-19 22:07 +0000 [r336791] Terry Wilson <twilson@digium.com>
* channels/chan_sip.c: Don't interfere with T.38 reinvites This is
an update to the fix for ASTERISK-18340 and ASTERISK-17725
2011-09-19 20:27 +0000 [r336733] Tilghman Lesher <tilghman@meg.abyt.es>
* Makefile.rules, include/asterisk/optional_api.h, Makefile,
configure, include/asterisk/autoconfig.h.in, main/Makefile,
codecs/gsm/Makefile, configure.ac: Various changes to allow 1.8
to compile on Mac OS X Lion (10.7) * Makefile workaround for 10.6
extended to work on 10.7 and later. * Now uses the 'weak' symbol
for Lion systems, which no longer support 'weak_import' Closes
ASTERISK-17612. Closes ASTERISK-18213. Tested by: tilghman, oej.
2011-09-19 20:07 +0000 [r336716] Jonathan Rose <jrose@digium.com>
* res/res_musiconhold.c, apps/app_queue.c, apps/app_mixmonitor.c,
apps/app_echo.c, apps/app_saycounted.c, apps/app_mp3.c,
apps/app_morsecode.c: Document applications that play audio and
do not answer unanswered calls. This patch is part of an effort
to document early media and its usage. If you are interested in
contributing to this documentation effort, there are probably
other applications worth documenting as well as an Asterisk wiki
article at
https://wiki.asterisk.org/wiki/display/AST/Early+Media+and+the+Progress+Application
2011-09-19 18:46 +0000 [r336658] Richard Mudgett <rmudgett@digium.com>
* UPGRADE.txt, apps/app_dial.c: Made Dial d and H options no longer
immediately auto-answer the calling leg. The Dial d and H options
break DTMF attended transfer atxferdropcall option. 1) Party A
calls party B. 2) Party B does a DTMF attended transfer to Party
C. If the dialplan uses the Dial d or H options to call Party C
then the Dial application answers the call immediately before
initiating the call leg to Party C. The premature answer causes
the transfer code to not invoke the atxferdropcall=no behavior
for a blonde transfer since Party C has "answered". The transfer
code thinks that Party B has "consulted" with Party C when Party
B hangs up and completes the transfer to Party A. Party A now
hears ringback until Party C actually answers. ASTERISK-13294
Dial d option. ASTERISK-11067 Dial H option to disconnect before
answer. The referenced issues made Dial answer with the d and H
options because many SIP and ISDN phones cannot send DTMF before
the call is connected. * Made require the dialplan to control
when or if the call needs to be answered to use the Dial
application d and H options. (The call is no longer surprise
answered when using the Dial d or H options.) Review:
https://reviewboard.asterisk.org/r/1381/ JIRA AST-623 JIRA
AST-666
2011-09-19 16:21 +0000 [r336591] Jason Parker <jparker@digium.com>
* contrib/realtime/postgresql/realtime.sql,
configs/cel_odbc.conf.sample, sounds/Makefile,
contrib/realtime/mysql/sipfriends.sql,
contrib/realtime/mysql/voicemail.sql, cel/cel_odbc.c, /,
contrib/realtime/mysql/iaxfriends.sql,
contrib/realtime/mysql/meetme.sql: Remove weird mergeinfo props
that make merges annoying sometimes.
2011-09-19 15:41 +0000 [r336572] Leif Madsen <lmadsen@digium.com>
* contrib/scripts/get_ilbc_source.sh: Update get_ilbc_source.sh
script to work again. Recently iLBC support in Asterisk has
changed after the acquisition of GIPS by Google. More information
about how this may affect you is available in a blog post at:
http://blogs.asterisk.org/2011/09/19/ilbc-support-in-asterisk-after-googles-acquisition-of-gips/
2011-09-19 15:25 +0000 [r336569] Richard Mudgett <rmudgett@digium.com>
* channels/sig_pri.c: Rework sig_pri_hangup() to be simpler and
clearer. JIRA AST-675
2011-09-19 13:33 +0000 [r336501] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: Add diversion header to a 302 redirect
response if we have diversion data (closes issue ASTERISK-18143)
patch by oej
2011-09-19 13:27 +0000 [r336499] Gregory Nietsky <gregory@distrotech.co.za>
* channels/chan_h323.c: A long time ago in a galaxy far far away a
IPv6 update was made, chan_h323 was not updated causeing all to
flee to chan_ooh323. the brave Jedi [asterisk developers]
pondered this miscarrige of justice and restored order to the
force for the sake of closing out 2 old issues. (closes issue
ASTERISK-17278) (closes issue ASTERISK-17500) Reported by: dread,
sybasesql Tested by: irroot Reviewed by: IRC (russellb,
kpfleming)
2011-09-19 12:06 +0000 [r336378-336440] Olle Johansson <oej@edvina.net>
* main/manager.c: Make sure manager_debug option is reset at reload
* Makefile: Revert accidental change that fixes OS/X Lion support
* Makefile, channels/chan_sip.c: Add missing unlock at MWI message
sending time (closes issue ASTERISK-18573) Patches:
sip_mwi_lock.patch (license #5041) by Gregory Hinton Nietsky
Thanks to irrot for the reminder, to Gregory for the patch!
2011-09-16 22:10 +0000 [r336312-336314] Terry Wilson <twilson@digium.com>
* funcs/func_frame_trace.c: Whitespace fix
* funcs/func_frame_trace.c: Add missing frame types to
func_frame_trace Also casts control frames to the proper enum so
that the compile will catch new additions.
2011-09-16 19:53 +0000 [r336294] Jonathan Rose <jrose@digium.com>
* include/asterisk/frame.h, main/channel.c, main/rtp_engine.c,
channels/chan_sip.c: Fix bad RTP media bridges in directmedia
calls on peers separated by multiple Asterisk nodes. In a
situation involving devices on separate Asterisk trunks, the
remote RTP bridge would break when starting a call with
directmedia. This patch queues a new type of control frame so
that our RTP bridge loop can properly detect when these
situations occur and check to see if peers need to be updated in
order to send their media to the proper location. (Closes issue
ASTERISK-18340) Reported by: Thomas Arimont (Closes issue
ASTERISK-17725) Reported by: kwk Tested by: twilson, jrose
2011-09-16 19:06 +0000 [r336234] Sean Bright <sean@malleable.com>
* UPGRADE.txt: Make a note that inotify won't work with an NFS
mounted spooler directory.
2011-09-16 10:09 +0000 [r335978-336166] Gregory Nietsky <gregory@distrotech.co.za>
* channels/chan_misdn.c: The round robin routing routine in
chan_misdn.c is broken. it rotates between ports but never checks
the channels in the ports. i have extensivly tested it and
verified it works on 1 upto 4 ports. before the patch only 1 out
of each port was used now all are used as expected. (closes issue
ASTERISK-18413) Reported by: irroot Tested by: irroot Reviewed
by: irroot Review: https://reviewboard.asterisk.org/r/1410/
* apps/app_queue.c: Locking order in app_queue.c causes deadlocks.
a channel lock must never be held with the queues container lock
held. the deadlock occured on masquerade. the queues container
lock is a relic of the past the old queue module lock. with ao2
there is no need to hold this lock when dealing with members this
patch removes unneeded locks. (closes issue ASTERISK-18101)
(closes issue ASTERISK-18487) Reported by: Paul Rolfe, Jason
Legault Tested by: irroot, Jason Legault, Paul Rolfe Reviewed by:
Matthew Nicholson Review:
https://reviewboard.asterisk.org/r/1402/
* channels/chan_agent.c: lock the channel before calling
ast_bridged_channel() to prevent a seg fault. AMI agents list
called on shutdown causes a segfault, introducing proper locking
will prevent this. (closes issue ASTERISK-18092) Reported by:
agustina Patches: chan_agent.patch (License #5041) patch uploaded
by irroot
2011-09-14 18:21 +0000 [r335851-335911] Richard Mudgett <rmudgett@digium.com>
* configure, include/asterisk/autoconfig.h.in, configure.ac: Remove
unnecessary libpri dependency checks in the configure script.
Using the --with-pri option with the configure script generated
an error about not having PRI_L2_PERSISTENCE if you did not have
the absolute latest libpri SVN checkout installed. The
AST_EXT_LIB_SETUP_DEPENDENT macro in the configure.ac script
seems to be for libraries that are dependent upon other libraries
and not necessarily for optional/added features within a library.
(closes issue ASTERISK-18535) Reported by: Michael Keuter
* channels/chan_dahdi.c: Fixed cut-n-paste regression using the
wrong variable. Fixes the missing DAHDI channels when using the
newer chan_dahdi.conf sections for channel configuration. (closes
issue ASTERISK-18496) Reported by: Sean Darcy Patches:
jira_asterisk_18496_v1.8.patch (license #5621) patch uploaded by
rmudgett Tested by: Sean Darcy, rmudgett
2011-09-14 13:28 +0000 [r335790] Matthew Nicholson <mnicholson@digium.com>
* main/manager.c: The tech and data members of
fast_originate_helper are not string fields. ASTERISK-17709
2011-09-13 22:10 +0000 [r335720] Richard Mudgett <rmudgett@digium.com>
* apps/app_directed_pickup.c: Remove obsolete todo comment about
PICKUPRESULT.
2011-09-13 21:33 +0000 [r335716] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
* main/asterisk.c: do parse defaultlanguage from asterisk.conf Do
parse the option "defaultlanguage" from the [options] section of
asterisk.conf, as in the sample config file. Otherwise the
build-time default language (normally "en") is always the default
one. Review: https://reviewboard.asterisk.org/r/1342/
Signed-off-by: Tzafrir Cohen (License #5035)
<tzafrir.cohen@xorcom.com>
2011-09-13 21:30 +0000 [r335714] Paul Belanger <pabelanger@digium.com>
* apps/app_meetme.c: Meetme should have 'core' support level
(closes issue ASTERISK-18542)
2011-09-13 18:52 +0000 [r335655] Tilghman Lesher <tilghman@meg.abyt.es>
* configure, configure.ac: Move mandatory checks closer to the
beginning of the file. If these are going to fail, they should
fail as quickly as possible.
2011-09-13 18:20 +0000 [r335618] Matthew Nicholson <mnicholson@digium.com>
* main/pbx.c, main/manager.c: Don't limit the size of appdata for
manager originate actions. ASTERISK-17709 Patch by: tilghman
(with modifications)
2011-09-13 07:11 +0000 [r335497] Russell Bryant <russell@digium.com>
* main/event.c, include/asterisk/event.h, res/ais/evt.c: Fix a
crash in res_ais. This patch resolves a crash observed in a load
testing environment that involved the use of the res_ais module.
I observed some crashes where the event delivery callback would
get called, but the length parameter incidcating how much data
there was to read was 0. The code assumed (with good reason I
would think) that if this callback got called, there was an event
available to read. However, if the rare case that there's nothing
there, catch it and return instead of blowing up. More
specifically, the change always ensure that the size of the
received event in the cluster is always big enough to be a real
ast_event. Review: https://reviewboard.asterisk.org/r/1423/
2011-09-12 15:54 +0000 [r335431-335433] Matthew Nicholson <mnicholson@digium.com>
* main/channel.c: Properly set caller_warning and callee_warning
before we try to use them. ASTERISK-18199 Patch by: elguero
Testing by: rtang
* bridges/bridge_multiplexed.c: Prevent a race condition when the
bridge technology changes. This change was ported from asterisk
10. ASTERISK-18155
2011-09-12 14:21 +0000 [r335320-335341] Kinsey Moore <kmoore@digium.com>
* apps/app_dial.c: Ensure frames are not written to dialed channel
if ringback is requested When a single channel was dialed and
there was media to be forwarded to the calling channel, the media
was written without regard for ringback causing silence to be
heard in some circumstances. This regression was introduced when
the meaning of "single" changed to mean only the number of
channels dialed. (closes issue ASTERISK-18083)
* channels/chan_iax2.c: Prevent IAX2 from getting IPv6 addresses
via DNS IAX2 does not support IPv6 and getting such addresses
from DNS can cause error messages on the remote end involving bad
IPv4 address casts in the presence of IPv6/IPv4 tunnels. This
patch ensures that IAX2 will not encounter IPv6 addresses via DNS
queries. (closes issue ASTERISK-18090)
2011-09-12 13:25 +0000 [r335319] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: Lock the peer->mvipvt to avoid crashes with
SIP history enabled After the launch of 1.6 event-based MWI we
have two threads handling the peer->mwipvt, which cause issues
with SIP history additions in combination with the max limit for
number of history entries. Review:
https://reviewboard.asterisk.org/r/1373/ (closes issue
ASTERISK-18288) Thanks to irrot for peer review. Work with this
bug funded by IPvision AS
2011-09-12 11:09 +0000 [r335259] Stefan Schmidt <sst@sil.at>
* channels/chan_sip.c: build_peer doesnt unlink a peer object from
peers_by_ip container which leads to a wrong refcounter value.
adding an ao2_unlink from the peers_by_ip container fix it.
Review: https://reviewboard.asterisk.org/r/1428/
2011-09-09 16:09 +0000 [r335064] Matthew Jordan <mjordan@digium.com>
* channels/chan_console.c, channels/sig_pri.c, channels/chan_oss.c,
main/channel.c, channels/chan_usbradio.c, main/dial.c,
channels/chan_dahdi.c, channels/chan_misdn.c,
channels/chan_skinny.c, funcs/func_frame_trace.c,
main/features.c, channels/chan_h323.c, channels/chan_alsa.c,
include/asterisk/frame.h, channels/sig_ss7.c,
channels/chan_mgcp.c, apps/app_dial.c, channels/chan_unistim.c,
main/pbx.c, addons/chan_ooh323.c, channels/chan_sip.c: Updated
SIP 484 handling; added Incomplete control frame When a SIP phone
uses the dial application and receives a 484 Address Incomplete
response, if overlapped dialing is enabled for SIP, then the 484
Address Incomplete is forwarded back to the SIP phone and the
HANGUPCAUSE channel variable is set to 28. Previously, the
Incomplete application dialplan logic was automatically
triggered; now, explicit dialplan usage of the application is
required. Additionally, this patch adds a new AST_CONTOL_FRAME
type called AST_CONTROL_INCOMPLETE. If a channel driver receives
this control frame, it is an indication that the dialplan expects
more digits back from the device. If the device supports overlap
dialing it should attempt to notify the device that the dialplan
is waiting for more digits; otherwise, it can handle the frame in
a manner appropriate to the channel driver. (closes issue
ASTERISK-17288) Reported by: Mikael Carlsson Tested by: Matthew
Jordan Review: https://reviewboard.asterisk.org/r/1416/
2011-09-08 22:27 +0000 [r334953] Richard Mudgett <rmudgett@digium.com>
* main/logger.c: Fix crash with res_fax when MALLOC_DEBUG and "core
stop gracefully" are used. Asterisk crashes if MALLOC_DEBUG is
enabled when res_fax tries to unregister its logger level. * Make
ast_logger_unregister_level() use ast_free() instead of free().
When MALLOC_DEBUG is enabled, ast_free() does not degenerate into
a call to free(). Therefore, if you allocated memory with a form
of ast_malloc you must free it with ast_free.
2011-09-07 19:35 +0000 [r334843] Paul Belanger <pabelanger@digium.com>
* channels/chan_iax2.c: Cleanup chan_iax2.c log messages Review:
https://code.asterisk.org/code/cru/CR-AST-11
2011-09-07 19:31 +0000 [r334840] Richard Mudgett <rmudgett@digium.com>
* main/features.c: Fix AMI action Park crash. * Made AMI action
Park not say anything to the parker channel (AMI header Channel2)
since the AMI action is a third party parking the call. (This is
a change in behavior that cannot be preserved without a lot of
effort.) * Made not play pbx-parkingfailed if the Park 's' option
is used. JIRA AST-660
2011-09-07 13:26 +0000 [r334682] Stefan Schmidt <sst@sil.at>
* main/features.c: Adding the Feature to sent a Reason Header in a
SIP Cancel message by set the flag AST_FLAG_ANSWERED_ELSEWHERE
before doing a masquerade in the pickup function.
2011-09-07 08:12 +0000 [r334616-334620] Alec L Davis <sivad.a@paradise.net.nz>
* CHANGES, apps/app_queue.c: peroid typo
* main/pbx.c: Prevent segfault if call arrives before Asterisk is
fully booted. Prevent ast_pbx_start and ast_run_start from
starting a new thread unless asterisk is fully booted. alecdavis
(license 585) Tested by: alecdavis Review:
https://reviewboard.asterisk.org/r/1407/
2011-09-06 13:48 +0000 [r334453] Gregory Nietsky <gregory@distrotech.co.za>
* apps/app_voicemail.c: Make SQL query in app_voicemail.c portable
LIMIT is not portable. Regression from r312212 (closes issue
ASTERISK-18255) Reported by: Leif Madsen Tested by: Leif Madsen
Review: https://reviewboard.asterisk.org/r/1415/
2011-09-23 Asterisk Development Team <asteriskteam@digium.com>
* Asterisk 1.8.7.0 Released.
2011-09-19 Asterisk Development Team <asteriskteam@digium.com>
* Asterisk 1.8.7.0-rc2 Released.
* r335851 | rmudgett | 2011-09-14 10:53:25 -0500 (Wed, 14 Sep 2011) |
11 lines
Fixed cut-n-paste regression using the wrong variable.
Fixes the missing DAHDI channels when using the newer chan_dahdi.conf
sections for channel configuration.
(closes issue ASTERISK-18496)
* r335911 | rmudgett | 2011-09-14 13:21:35 -0500 (Wed, 14 Sep 2011) |
13 lines
Remove unnecessary libpri dependency checks in the configure script.
Using the --with-pri option with the configure script generated an
error
about not having PRI_L2_PERSISTENCE if you did not have the absolute
latest libpri SVN checkout installed.
The AST_EXT_LIB_SETUP_DEPENDENT macro in the configure.ac script seems
to
be for libraries that are dependent upon other libraries and not
necessarily for optional/added features within a library.
(closes issue ASTERISK-18535)
* r336572 | lmadsen | 2011-09-19 10:41:16 -0500 (Mon, 19 Sep 2011) | 7
lines
Update get_ilbc_source.sh script to work again.
Recently iLBC support in Asterisk has changed after the acquisition of
GIPS
by Google. More information about how this may affect you is available
in a
blog post at:
http://blogs.asterisk.org/2011/09/19/ilbc-support-in-asterisk-after-googles-acquisition-of-gips/
* r335714 | pabelanger | 2011-09-13 16:30:18 -0500 (Tue, 13 Sep 2011)
| 4 lines
Meetme should have 'core' support level
(closes issue ASTERISK-18542)
2011-09-07 Asterisk Development Team <asteriskteam@digium.com>
* Asterisk 1.8.7.0-rc1 Released.
2011-09-06 13:48 +0000 [r334453] Gregory Nietsky <gregory@distrotech.co.za>
* apps/app_voicemail.c: Make SQL query in app_voicemail.c portable
LIMIT is not portable. Regression from r312212 (closes issue
ASTERISK-18255) Reported by: Leif Madsen Tested by: Leif Madsen
Review: https://reviewboard.asterisk.org/r/1415/
2011-09-02 20:59 +0000 [r334296-334355] Richard Mudgett <rmudgett@digium.com>
* res/res_musiconhold.c: MusicOnHold has extra unref which may lead
to memory corruption and crash. The problem happens when a call
is disconnected and you had started a MOH class that does not use
the files mode. If you define REF_DEBUG and recreate the problem,
it will announce itself with the following warning: Attempt to
unref mohclass 0xb70722e0 (default) when only 1 ref remained, and
class is still in a container! * Fixed moh_alloc() and
moh_release() functions not handling the state->class reference
consistently. (closes issue ASTERISK-18346) Reported by: Mark
Murawski Patches: jira_asterisk_18346_v1.8.patch (license #5621)
patch uploaded by rmudgett Tested by: rmudgett, Mark Murawski
Review: https://reviewboard.asterisk.org/r/1404/
* main/config.c, include/asterisk/config.h: Fix potential memory
allocation failure crashes in config.c. * Added required checks
to the returned memory allocation pointers to prevent crashes. *
Made ast_include_rename() create a replacement ast_variable list
node if the new filename is longer than the available space.
Fixes potential crash and memory leak. * Factored out
ast_variable_move() from ast_variable_update() so
ast_include_rename() can also use it when creating a replacement
ast_variable list node. * Made the filename stuffed at the end of
the struct a minimum allocated size in ast_variable_new() in case
ast_include_rename() changes the stored filename. * Constify
struct char pointers pointing to strings stuffed at the end of
the struct for: ast_variable, cache_file_mtime, and
ast_config_map. * Factored out cfmtime_new() to remove inlined
code and allow some struct pointers to become const. * Removed
the list lock from struct cache_file_mtime that was never used. *
Added doxygen comments to several structure elements and better
documented what strings are stuffed at the struct end char array.
* Reworked ast_config_text_file_save() and set_fn() to handle
allocation failure of the include file scratch pad object
tracking blank lines. * Made ast_config_text_file_save() fn[]
declared with PATH_MAX to ensure it is long enough for any
filename with path. Also reduced the number of container fileset
buckets from a rediculus 180,000 to 1023. JIRA AST-618 Review:
https://reviewboard.asterisk.org/r/1378/
2011-09-01 17:38 +0000 [r334229-334234] Tilghman Lesher <tilghman@meg.abyt.es>
* main/pbx.c: Remove 1.6 compatibility documentation from 1.8, as
it no longer applies.
* res/res_config_odbc.c: Create a local alias for
ast_odbc_clear_cache. As a function pointer, the reference has to
be resolved at load time irrespective of the RTLD_LAZY flag.
Creating a local alias solves this problem, because the structure
is initialized with that local function pointer, while the actual
function can remain lazily linked until runtime. The reason why
this is important is because we lazily load function references
during the module loading process, in order to obtain priority
values for each module, ensuring that modules are loaded in the
correct order. Previous to this change, when this module was
initially loaded, the module loader would emit a symbol
resolution error, because of the above requirement. Closes
ASTERISK-18399 (reported by Mikael Carlsson, fix suggested by
Walter Doekes, patch by me)
2011-08-31 18:50 +0000 [r334156] Matthew Nicholson <mnicholson@digium.com>
* channels/chan_sip.c: Disable T.38 when we get a invite with image
media port set to 0 ASTERISK-17678
2011-08-31 15:57 +0000 [r334009-334012] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c: No DAHDI channel available for conference,
user introduction disabled. The following error will consistently
occur when trying to dial into a MeetMe conference when the
server does not have DAHDI hardware installed: app_meetme.c: No
DAHDI channel available for conference, user introduction
disabled (is chan_dahdi loaded?) While chan_dahdi is loaded
correctly during compilation and install of Asterisk/Dahdi,
including associated modules, etc., a chan_dahdi.conf
configuration file in /etc/asterisk is not created by FreePBX if
hardware does not exist, causing MeetMe to be unable to open a
DAHDI pseudo channel. * Allow chan_dahdi to create a pseudo
channel when there is no chan_dahdi.conf file to load. (closes
issue ASTERISK-17398) Reported by: Preston Edwards Patches:
jira_asterisk_17398_v1.8.patch (license #5621) patch uploaded by
rmudgett Tested by: rmudgett
* main/channel.c, channels/chan_agent.c: Call pickup race leaves
orphaned channels or crashes. Multiple users attempting to pickup
a call that has been forked to multiple extensions either crashes
or fails a masquerade with a "bad things may happen" message.
This is the scenario that is causing all the grief: 1) Pickup
target is selected 2) target is marked as being picked up in
ast_do_pickup() 3) target is unlocked by ast_do_pickup() 4) app
dial or queue gets a chance to hang up losing calls and calls
ast_hangup() on target 5) SINCE A MASQUERADE HAS NOT BEEN SETUP
YET BY ast_do_pickup() with ast_channel_masquerade(),
ast_hangup() completes successfully and the channel is no longer
in the channels container. 6) ast_do_pickup() then calls
ast_channel_masquerade() to schedule the masquerade on the dead
channel. 7) ast_do_pickup() then calls ast_do_masquerade() on the
dead channel 8) bad things happen while doing the masquerade and
in the process ast_do_masquerade() puts the dead channel back
into the channels container 9) The "orphaned" channel is visible
in the channels list if a crash does not happen. This patch does
the following: * Made ast_hangup() set AST_FLAG_ZOMBIE on a
successfully hung-up channel and not release the channel lock
until that has happened. * Made __ast_channel_masquerade() not
setup a masquerade if either channel has AST_FLAG_ZOMBIE set. *
Fix chan_agent misuse of AST_FLAG_ZOMBIE since it would no longer
work. (closes issue ASTERISK-18222) Reported by: Alec Davis
Tested by: rmudgett, Alec Davis, irroot, Karsten Wemheuer (closes
issue ASTERISK-18273) Reported by: Karsten Wemheuer Tested by:
rmudgett, Alec Davis, irroot, Karsten Wemheuer Review:
https://reviewboard.asterisk.org/r/1400/
2011-08-31 15:18 +0000 [r334006] Kinsey Moore <kmoore@digium.com>
* channels/chan_sip.c: Correct an AMI protocol violation with
SIPshowpeer The response of SIPshowpeer ends with "\r\n\r\n".
Since other commands are ended by using \r\n this confuses any
interfacing script. (closes issue ASTERISK-17486)
2011-08-30 21:16 +0000 [r333947] Alexandr Anikin <may@telecom-service.ru>
* addons/ooh323c/src/oochannels.c, addons/ooh323c/src/ooq931.c,
addons/ooh323c/src/ooCalls.c, addons/ooh323c/src/ooh323.c,
addons/ooh323c/src/ooGkClient.c, addons/ooh323c/src/ooCalls.h:
cleanups in ACF/ARJ GK replies processing fixed long (24 sec)
pause if acf/arj proccessed before ast_cond_wait called to wait
this
2011-08-29 21:38 +0000 [r333836] Terry Wilson <twilson@digium.com>
* channels/chan_sip.c: Refresh peer address if DNS unavailable at
peer creation If Asterisk starts and no DNS is available,
outbound registrations will fail indefinitely. This patch copies
the address from the sip_registry struct, which will be updated,
to the peer->addr when necessary. If dnsmgr is enabled, the
registration fails without the patch because even though the
address on the registry is updated via dnsmgr, the address is
just copied on the first try. Since we use ast_sockaddr_copy,
dnsmgr can't update the address that is copied to the sip_pvt or
peers. Closes issue ASTERISK-18000 Review:
https://reviewboard.asterisk.org/r/1335/
2011-08-29 21:06 +0000 [r333784-333785] Richard Mudgett <rmudgett@digium.com>
* include/asterisk/channel.h: Add some do not hold locks notes to
channel.h
* addons/chan_mobile.c: Fix deadlock potential of
chan_mobile.c:mbl_ast_hangup().
2011-08-29 17:11 +0000 [r333630] Matthew Jordan <mjordan@digium.com>
* apps/app_voicemail.c: Fixed improperly formatted TestEvent AMI
message in app_voicemail
2011-08-29 15:55 +0000 [r333569] Jonathan Rose <jrose@digium.com>
* res/res_jabber.c: Accidental use of variable client->status
instead of client->state in from ASTERISK-18078 (issue
ASTERISK-18078)
2011-08-28 09:49 +0000 [r333507] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
* channels/chan_vpb.cc: chan_vpb: remove unused variables (gcc4.6)
GCC 4.6 detects variables that get assined to, but never used
later. Also removes some remmed-out lines that become invalid.
(closes issue ASTERISK-18336) Signed-off-by: Tzafrir Cohen
(License #5035) <tzafrir.cohen@xorcom.com>,
2011-08-26 16:19 +0000 [r333378] Jonathan Rose <jrose@digium.com>
* res/res_jabber.c: [patch] Buddies are always auto-registered when
processing the roster Reporter said autoregister flag was ignored
for registering 'buddies' which had a subscription to us.
Verified that this was the case and observed how the patch
addressed this and made sure it didn't break anything. (closes
issue ASTERISK-14233) Reported by: Simon Arlott Patches:
asterisk-0015229.patch (license #5756) patch uploaded by Simon
Arlott Tested by: Jonathan Rose
2011-08-26 14:36 +0000 [r333339-333354] Matthew Jordan <mjordan@digium.com>
* apps/app_voicemail.c: Fixed incorrect pointer copy to structure
copy in revision 333339
* apps/app_voicemail.c: Bug fixes for voicemail user emailsubject /
emailbody. This code change fixes a few issues with the voicemail
user override of emailbody and emailsubject, including escaping
the strings, potential memory leaks, and not overriding the
voicemail defaults. Revision 325877 fixed this for
ASTERISK-16795, but did not fix it for ASTERISK-16781. A
subsequent check-in prevented 325877 from being applied to 10.
This check-in resolves both issues, and applies the changes to
1.8, 10, and trunk. (closes issue ASTERISK-16781) Reported by:
Sebastien Couture Tested by: mjordan (closes issue
ASTERISK-16795) Reported by: mdeneen Tested by: mjordan Review:
https://reviewboard.asterisk.org/r/1374
2011-08-25 19:00 +0000 [r333267] Jason Parker <jparker@digium.com>
* Makefile: Fix for DESTDIR spaces patch.
2011-08-25 18:47 +0000 [r333265] Jonathan Rose <jrose@digium.com>
* res/res_jabber.c: Segfault when publishing device states via XMPP
and not connected When using publishing device state with
res_jabber, Asterisk will attempt to send a device state using
the unconnected client using iks_send_raw and crash. This patch
checks the validity of the connection before attempting to send
the device state. (closes issue ASTERISK-18078) Reported by:
Michael L. Young Patches:
res_jabber-segfault-pubsub-not-connected2.patch (license #5026)
patch uploaded by Michael L. Young Tested by: Jonathan Rose
2011-08-25 15:27 +0000 [r333201] Jason Parker <jparker@digium.com>
* makeopts.in, sounds/Makefile, Makefile, build_tools/mkpkgconfig,
configure, configure.ac: Fix installation into directories
containing spaces. This also vastly simplifies the logic in
sounds/Makefile (Closes issue ASTERISK-18290) Reported by: Paul
Belanger Review: https://reviewboard.asterisk.org/r/1379/
2011-08-23 18:14 +0000 [r333010] Richard Mudgett <rmudgett@digium.com>
* apps/app_queue.c: Memory Leak in app_queue The patch that was
committed in the 1.6.x versions of Asterisk for ASTERISK-15862
actually fixed two issues. One was not applicable to 1.8 but the
other is. queue_leak.patch fixes the portion applicable to 1.8.
(closes issue ASTERISK-18265) Reported by: Fred Schroeder
Patches: queue_leak.patch (license #5049) patch uploaded by
mmichelson Tested by: Thomas Arimont
2011-08-23 18:11 +0000 [r333009] Matthew Nicholson <mnicholson@digium.com>
* UPGRADE.txt, configs/sip.conf.sample, CHANGES,
channels/sip/include/sip.h: default 'sipstorecause' to no We've
decided to disable this feature by default in future 1.8
versions. This would be an unexpected behavior change for anyone
depending on that SIP_CAUSE update in their dialplan. Please
refer to the asterisk-dev mailing list more information:
http://lists.digium.com/pipermail/asterisk-dev/2011-August/050626.html
(issue AST-580)
2011-08-22 22:11 +0000 [r332939-332945] Richard Mudgett <rmudgett@digium.com>
* apps/app_queue.c, main/config.c, include/asterisk/config.h:
Revert previous commit. Not ready yet.
* apps/app_queue.c, main/config.c, include/asterisk/config.h:
Signed
* main/config.c: Minor code optimizations. * Simplify
ast_category_browse() logic for easier understanding. * Remove
dead code in ast_variable_delete() and simplify some of its
logic.
2011-08-22 19:41 +0000 [r332876] Paul Belanger <pabelanger@digium.com>
* channels/chan_gtalk.c: Revert previous commit It seems google is
still making changes to the protocol. (issue ASTERISK-18301)
2011-08-22 19:32 +0000 [r332874] Richard Mudgett <rmudgett@digium.com>
* apps/app_queue.c: Reference leaks in app_queue. * Fixed
load_realtime_queue() leaking a queue reference when it
overwrites q when processing a realtime queue. (issue
ASTERISK-18265) * Make join_queue() unreference the queue
returned by load_realtime_queue() when it is done with the
pointer. The load_realtime_queue() returns a reference to the
just loaded realtime queue. * Fixed queues container reference
leak in queues_data_provider_get(). * queue_unref() should not
return q that was just unreferenced. * Made logic in
__queues_show() and queues_data_provider_get() when calling
load_realtime_queue() easier to understand.
2011-08-22 18:15 +0000 [r332817] Matthew Jordan <mjordan@digium.com>
* main/app.c, configs/manager.conf.sample,
include/asterisk/manager.h, apps/app_voicemail.c,
include/asterisk/test.h, main/manager.c, main/file.c,
main/test.c: Review: https://reviewboard.asterisk.org/r/1364/
This update adds a new AMI event, TestEvent, which is enabled
when the TEST_FRAMEWORK compiler flag is defined. It also adds
initial usage of this event to app_voicemail. The TestEvent AMI
event is used extensively by the voicemail tests in the Asterisk
Test Suite.
2011-08-22 18:14 +0000 [r332759-332816] Richard Mudgett <rmudgett@digium.com>
* res/res_config_pgsql.c, res/res_config_odbc.c: Memory leaks in
realtime_multi_xxx() when database access returns error. * Fix
realtime_multi_pgsql() configuration memory leak when the
database access returns an error. * Fix realtime_multi_odbc()
configuration category use after free when the database access
returns an error.
* main/config.c: Memory leak reading realtime database variable
list. Calling ast_load_realtime() can leak the last list node if
the read list only contains empty variable value items. * Fixed
list filter loop in ast_load_realtime() to delete the list node
immediately instead of the next time through the loop. The next
time through the loop may not happen if the node to delete is the
last in the list. (issue ASTERISK-18277) (issue ASTERISK-18265)
Patches: jira_asterisk_18265_v1.8_config.patch (license #5621)
patch uploaded by rmudgett
2011-08-21 14:31 +0000 [r332699] Paul Belanger <pabelanger@digium.com>
* channels/chan_gtalk.c: Fix outgoing calls in chan_gtalk (closes
issue ASTERISK-18301) Reported by: az1324
2011-08-18 21:26 +0000 [r332559] Terry Wilson <twilson@digium.com>
* main/netsock2.c: Fix possible error on stringification of
IPv4-mapped addrs The FreeBSD netsock2 test has been failing for
a while. We were pasing sa->len to getnameinfo instead of
sa_tmp->len. ASTERISK-18289
2011-08-18 19:28 +0000 [r332503] Kinsey Moore <kmoore@digium.com>
* channels/chan_dahdi.c: CRC4 in "dahdi show status" gives wrong
impression to T1 users Change CRC4 to CRC in the output of "dahdi
show status" so that it can apply in more situations without
confusing users, especially since T1 lines use CRC6 instead of
CRC4. (closes issue AST-471)
2011-08-18 14:46 +0000 [r332355-332446] Tilghman Lesher <tilghman@meg.abyt.es>
* build_tools/cflags.xml, build_tools/cflags-devmode.xml: Move
BETTER_BACKTRACES out of development mode, as it's useful when
DEBUG_THREADS is enabled.
* makeopts.in, sounds/Makefile, Makefile, agi/Makefile,
utils/Makefile, configure, include/asterisk/autoconfig.h.in,
configure.ac, Makefile.moddir_rules: Re-add support for spaces in
pathnames, including now spaces in DESTDIR. This was initially
added to 1.8 prior to release, primarily to support the standard
paths on Mac OS X, but was partially reverted recently in
Subversion, due to the lack of support for spaces in DESTDIR.
This commit restores support for the standard paths on Mac OS X,
and also includes support for spaces in DESTDIR. (closes issue
ASTERISK-18290) Reported by: pabelanger Review:
https://reviewboard.asterisk.org/r/1326/
2011-08-17 17:35 +0000 [r332320] Terry Wilson <twilson@digium.com>
* res/res_timing_timerfd.c: Don't read from a disarmed or invalid
timerfd Numerous isues have been reported for deadlocks that are
caused by a blocking read in res_timing_timerfd on a file
descriptor that will never be written to. This patch adds some
checks to make sure that the timerfd is both valid and armed
before calling read(). Should fix: ASTERISK-1842, ASTERISK-18197,
ASTERISK-18166, AST-486 AST-495, AST-507 and possibly others.
2011-08-17 15:51 +0000 [r332264] Richard Mudgett <rmudgett@digium.com>
* channels/sig_pri.c, channels/sig_pri.h, channels/chan_dahdi.c,
configs/chan_dahdi.conf.sample, configure,
include/asterisk/autoconfig.h.in, configure.ac: Outgoing BRI
calls fail when using Asterisk 1.8 with HA8, HB8, and B410P
cards. France Telecom brings layer 2 and layer 1 down on BRI
lines when the line is idle. When layer 1 goes down Asterisk
cannot make outgoing calls and the HA8 and HB8 cards also get IRQ
misses. The inability to make outgoing calls is because the line
is in red alarm and Asterisk will not make calls over a line it
considers unavailable. The IRQ misses for the HA8 and HB8 card
are because the hardware is switching clock sources from the line
which just brought layer 1 down to internal timing. There is a
DAHDI option for the B410P card to not tell Asterisk that layer 1
went down so Asterisk will allow outgoing calls: "modprobe
wcb4xxp teignored=1". There is a similar DAHDI option for the HA8
and HB8 cards: "modprobe wctdm24xxp bri_teignored=1".
Unfortunately that will not clear up the IRQ misses when the
telco brings layer 1 down. * Add layer 2 persistence option to
customize the layer 2 behavior on BRI PTMP lines. The new option
has three settings: 1) Use libpri default layer 2 setting. 2)
Keep layer 2 up. Bring layer 2 back up when the peer brings it
down. 3) Leave layer 2 down when the peer brings it down. Layer 2
will be brought up as needed for outgoing calls. JIRA AST-598
2011-08-17 14:31 +0000 [r332234] Matthew Nicholson <mnicholson@digium.com>
* channels/chan_sip.c: print a warning instructing the user to
disable storesipcause if we process 100 or more scheduler entries
at a time AST-580
2011-08-16 20:10 +0000 [r332176] Paul Belanger <pabelanger@digium.com>
* tests/test_db.c, tests/test_linkedlists.c, tests/test_sched.c,
tests/test_netsock2.c, tests/test_strings.c, tests/test_pbx.c,
tests/test_func_file.c, tests/test_security_events.c,
tests/test_stringfields.c, tests/test_time.c, tests/test_skel.c,
tests/test_locale.c, tests/test_acl.c, tests/test_devicestate.c,
tests/test_utils.c, tests/test_aoc.c, tests/test_astobj2.c,
tests/test_poll.c, tests/test_amihooks.c,
tests/test_substitution.c, tests/test_heap.c,
tests/test_ast_format_str_reduce.c, tests/test_expr.c,
tests/test_logger.c, tests/test_gosub.c, tests/test_app.c,
tests/test_dlinklists.c, tests/test_event.c: Flag test modules as
'core' Review: https://reviewboard.asterisk.org/r/1369/
2011-08-16 17:38 +0000 [r332118] Jonathan Rose <jrose@digium.com>
* channels/chan_sip.c: ASTERISK-18067 ASTERISK-15479 - White Space
affects mailbox value, multiple MWI subs Before, having multiple
subscriptions to mailboxes on a sip peer set via the mailbox
setting in sip.conf would only result in updates being sent on
whichever mailbox triggered the mwi event. Now all of them get
counted regardless. Also fixes a bug involving parsing of the
mailbox option in sip.conf so that trailing and leading spaces
before/after commas are trimmed. (closes issue ASTERISK-18067)
Reported by: aragon (closes issue ASTERISK-15479) Reported by:
Ben Winslow Patches:
chan_sip.c-mwi_multi_mailbox_fix-1.6.2.13.diff (License #5288)
patch uploaded by Ben Winslow
2011-08-16 16:31 +0000 [r332100] Richard Mudgett <rmudgett@digium.com>
* CHANGES, configs/features.conf.sample, main/asterisk.c,
main/features.c: Fix multiple parking issues. JIRA ASTERISK-17183
Multi-parkinglot directs calls to wrong parkinglot. JIRA
ASTERISK-17870 Cannot retrieve parked calls. JIRA ASTERISK-17430
ParkedCall() with no extension should pickup first available call
and does not. JIRA AST-576 Issues with parking lots * Removed
searching for parking lots by extension. Parking lots can only be
found by the parking lot name since parking lot access extensions
and spaces are not guaranteed to be unique. * Added
parking_lot_name option to the Park and ParkedCall applications.
Updated documentation for Park and ParkedCall applications. * Add
parkext_exclusive configuration option to make parking entry
extensions specify which parking lot they access. (closes issue
ASTERISK-17183) Reported by: David Cabrejos Tested by: rmudgett,
David Cabrejos (closes issue ASTERISK-17870) Reported by: Remi
Quezada (closes issue ASTERISK-17430) Reported by: Philippe
Lindheimer JIRA ASTERISK-17452 Parking_offset not used JIRA
AST-624 'next' setting for findslot does nothing * Reimplemented
since findslot feature option broken by -r114655. (closes issue
ASTERISK-17452) Reported by: David Woolley Tested by: rmudgett
JIRA ASTERISK-15792 Dialplan continues execution after transfer
to park. This happens for DTMF attended transfer, DTMF blind
transfer, and DTMF one-touch-parking if the party initiating
these features also initiated the call. * Fixed the return code
from the affected builtin features when parking a call. (closes
issue ASTERISK-15792) Reported by: Mat Murdock Tested by:
rmudgett, twilson JIRA AST-607 The courtesytone is not playing to
the expected call when picking up a parked call. This is mostly a
documentation problem. However, the option is not reset to the
default when features.conf is reloaded. * Updated
features.conf.sample documentation for courtesytone and
parkedplay options. * Reset the parkedplay option to default when
features.conf is reloaded. JIRA AST-615 AMI Park action followed
by features reload results in orphaned channels in parking lot. *
Reloading features.conf will not touch parking lots that have
calls still parked in them. Reload again at a later time. Misc
additional fixes: * Added unit test for parking lot dialplan
usage checking. * Made update connected line when a parked call
is retrieved from a parking lot. * Made retrieved parked call
stop ringing or MOH depending upon how the call was waiting in
the parking lot. * Made CLI "features show" indicate if the
parking lot is enabled for use. * Added PARKINGDYNEXTEN channel
variable to allow dynamic parking lots to specify the parking lot
access extension. * Made AMI ParkedCalls action ParkedCall events
have a Parkinglot header. * Made AMI ParkedCalls action
ParkedCallsComplete event have a Total header. * Fixed potential
deadlock from AMI Park action holding channel locks while calling
masq_park_call(). * Fixed several places where ast_strdupa() were
used inside of loops. (Mostly fixed by refactoring the loop body
into its own function.) * Fixed copy_parkinglot() copying too
much from the source parking lot. Extracted the parking lot
configuration settings into struct parkinglot_cfg. * Refactored
courtesytone playing code to put the channel not playing the tone
in autoservice. * Fix when pbx-parkingfailed is played that the
other channel is put in autoservice if it exists. * Fixed
parkinglot reference leak in parked_call_exec() error paths. *
Fixed parkinglot_unref() use of parkinglot after it was unreffed.
* Made destroy the struct ast_parkinglot parkings lock when done.
* Refactored the features.conf parking lot configuration code to
eliminate redundancy. * Fixed feature reload to better protect
parking lots. * Fixed parking lot container reference leak in
handle_parkedcalls(). * Fixed the total count in
handle_parkedcalls(). Review:
https://reviewboard.asterisk.org/r/1358/
2011-08-16 15:06 +0000 [r332021-332026] Matthew Nicholson <mnicholson@digium.com>
* channels/sip/include/sip.h, channels/chan_sip.c: use
DEFAULT_STORE_SIP_CAUSE to set the default value for the
'storesipcause' option AST-580
* configs/sip.conf.sample, CHANGES, channels/chan_sip.c: Added the
'storesipcause' option to sip.conf to allow the user to disable
the setting of HASH(SIP_CAUSE,<chan name>) on the channel. Having
chan_sip set HASH(SIP_CAUSE,<chan name>) on the channel carries a
significant performance penalty because of the usage of the
MASTER_CHANNEL() dialplan function. AST-580
2011-08-15 17:24 +0000 [r331955] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c: Fix some minor chan_dahdi config load
issues. * Address chan_dahdi.conf dahdichan option todo item
about needing line number. * Make ignore_failed_channels option
also apply to dahdichan option. * Don't attempt to create a
default pseudo channel if the chan_dahdi.conf channel/channels
option is not allowed. * Add a similar check for dahdichan in
normal chan_dahdi.conf sections as is done in users.conf.
2011-08-15 15:21 +0000 [r331886] Paul Belanger <pabelanger@digium.com>
* main/rtp_engine.c: Fix noisy message when briding channels
(closes issue ASTERISK-18270) Reported by: Federico Alves
2011-08-15 15:12 +0000 [r331867] David Vossel <dvossel@digium.com>
* channels/chan_sip.c: Fixes locking inversion issues present in
the handling of the sip REFER method. (closes issue
ASTERISK-18082) Reported by: James Van Vleet
2011-08-12 19:01 +0000 [r331774] Matthew Nicholson <mnicholson@digium.com>
* apps/app_queue.c: Unlock the channel before calling update_queue.
Holding the channel lock when calling update_queue which attempts
to lock the queue lock can cause a deadlock. This deadlock
involves the following chain: 1. hold chan lock -> wait queue
lock 2. hold queue lock -> wait agent list lock 3. hold agent
list lock -> wait chan list lock 4. hold chan list lock -> wait
chan lock
2011-08-12 18:58 +0000 [r331714-331771] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c: Suppress warning message when using
DAHDITransfer or DAHDIHangup. * The fake event should only be
processed by the channel that currently owns the private and not
the associated call waiting or 3-way channel. JIRA AST-620 JIRA
SWP-3616
* channels/chan_dahdi.c: AMI actions DAHDIHangup and DAHDITransfer
have no effect. The AMI actions DAHDIHangup and DAHDITransfer
have no effect on a DAHDI channel. These two AMI actions are
highly specialized to analog channels and appear to make the
channel behave like a jack port for headsets. * Made the faked
DAHDI event get processed before a normal media stream read in
dahdi_read() instead of trying to trigger an exception read by
setting the AST_FLAG_EXCEPTION flag. Apparently a change was made
long ago that changed how AST_FLAG_EXCEPTION is processed in the
core. Unfortunately, the faked DAHDI events no longer worked when
that happened. * Updated the DAHDI AMI action documentation for
the following actions: DAHDITransfer, DAHDIHangup,
DAHDIDialOffhook, DAHDIDNDon, DAHDIDNDoff, DAHDIShowChannels, and
DAHDIRestart. * Made use sscanf() instead of atoi() for better
error checking of the DAHDIChannel header string. JIRA AST-620
JIRA SWP-3616
2011-08-12 16:30 +0000 [r331658] Terry Wilson <twilson@digium.com>
* tests/test_netsock2.c: Fix netsock2 multiple zero-expansion test
Remove erroneous single bracket.
2011-08-12 16:20 +0000 [r331649] Kinsey Moore <kmoore@digium.com>
* main/logger.c: Logger does not warn of failure to open logging
channels Currently, logger only prints an error message to stderr
when it fails to open a logger channel where many users will not
see it because the logger lock is held. The alternative provided
by this patch is to log the error to all attached consoles in the
hopes that it will be easier to see. Additionally, this patch
prevents the failed logger channel from being added to the list
where it would silently fail on each call to the Asterisk logger.
(closes issue ASTERISK-16231) Review:
https://reviewboard.asterisk.org/r/1338
2011-08-12 15:49 +0000 [r331635] Jonathan Rose <jrose@digium.com>
* apps/app_dial.c, apps/app_meetme.c: Fixes 32bit compilation
warnings brought on by 331634 in app_dial and app_meetme
2011-08-11 21:46 +0000 [r331578] Jason Parker <jparker@digium.com>
* apps/app_dial.c, apps/app_meetme.c: Use proper values for 64-bit
option flags. Also, reusing bits es no bueno, so change the value
of a duplicate. (issue ASTERISK-18239)
2011-08-11 21:39 +0000 [r331575] Richard Mudgett <rmudgett@digium.com>
* funcs/func_shell.c: Segfault in shell_helper in func_shell.c. The
return value of popen() was not checked for failure to open.
(closes issue ASTERISK-18109) JIRA SWP-3633 Reported by: Michael
Myles Tested by: rmudgett
2011-08-10 22:23 +0000 [r331517] Kinsey Moore <kmoore@digium.com>
* channels/chan_sip.c: SIP Notify via AMI or CLI leaks SIP PVTs Any
SIP notify sent via AMI or CLI leaks a SIP PVT with ref count +2.
Removing the additional ref just before the invite and adding an
unref following it corrects the issue as seen via REF_DEBUG. The
unref existed in a distant revision and it appears as though the
wrong ref operation was removed. (closes issue ASTERISK-18091)
Review: https://reviewboard.asterisk.org/r/1332/
2011-08-10 20:29 +0000 [r331461] Richard Mudgett <rmudgett@digium.com>
* main/logger.c: Output of queue log not started until logger
reloaded. ASTERISK-15863 caused a regression with queue logging.
The output of the queue log is not started until the logger
configuration is reloaded. * Queue log initialization is
completely delayed until the first message is posted to the queue
log system. Including the initial opening of the queue log file.
* Fixed rotate_file() ROTATE strategy to give the file just
rotated out to the configured exec function after rotate. Just
like the other strategies. * Fixed logger reload to always post
the queue reload entry instead of just if there is a queue log
file. * Refactored some code to eliminate some redundancy and to
reduce stack utilization. (closes issue ASTERISK-17036) JIRA
SWP-2952 Reported by: Juan Carlos Valero Patches:
jira_asterisk_17036_v1.8.patch (license #5621) patch uploaded by
rmudgett Tested by: rmudgett (closes issue ASTERISK-18208)
Reported by: Christian Pinedo Review:
https://reviewboard.asterisk.org/r/1333/
2011-08-31 Asterisk Development Team <asteriskteam@digium.com>
* Asterisk 1.8.6.0 Released.
2011-08-25 Asterisk Development Team <asteriskteam@digium.com>
* Asterisk 1.8.6.0-rc3 Released.
------------------------------------------------------------------------
r333201 | qwell | 2011-08-25 10:27:06 -0500 (Thu, 25 Aug 2011) | 8 lines
Fix installation into directories containing spaces.
This also vastly simplifies the logic in sounds/Makefile
(Closes issue ASTERISK-18290)
Reported by: Paul Belanger
Review: https://reviewboard.asterisk.org/r/1379/
------------------------------------------------------------------------
2011-08-22 Asterisk Development Team <asteriskteam@digium.com>
* Asterisk 1.8.6.0-rc2 Released.
------------------------------------------------------------------------
r331575 | rmudgett | 2011-08-11 16:39:58 -0500 (Thu, 11 Aug 2011) | 9 lines
Segfault in shell_helper in func_shell.c.
The return value of popen() was not checked for failure to open.
(closes issue ASTERISK-18109)
JIRA SWP-3633
Reported by: Michael Myles
Tested by: rmudgett
------------------------------------------------------------------------
r332355 | tilghman | 2011-08-17 14:21:36 -0500 (Wed, 17 Aug 2011) | 13 lines
Re-add support for spaces in pathnames, including now spaces in DESTDIR.
This was initially added to 1.8 prior to release, primarily to support the
standard paths on Mac OS X, but was partially reverted recently in Subversion,
due to the lack of support for spaces in DESTDIR. This commit restores support
for the standard paths on Mac OS X, and also includes support for spaces in
DESTDIR.
(closes issue ASTERISK-18290)
Reported by: pabelanger
Review: https://reviewboard.asterisk.org/r/1326/
------------------------------------------------------------------------
r332559 | twilson | 2011-08-18 16:26:01 -0500 (Thu, 18 Aug 2011) | 7 lines
Fix possible error on stringification of IPv4-mapped addrs
The FreeBSD netsock2 test has been failing for a while. We were
pasing sa->len to getnameinfo instead of sa_tmp->len.
ASTERISK-18289
------------------------------------------------------------------------
2011-08-10 Asterisk Development Team <asteriskteam@digium.com>
* Asterisk 1.8.6.0-rc1 Released.
2011-08-10 13:47 +0000 [r331315] Kinsey Moore <kmoore@digium.com>
* main/manager.c: AMI action ModuleReload returns Error if Module:
missing or empty An empty string was not being checked for
properly causing identification of the module to be reloaded to
fail and return an Error with message "No such module." (closes
issue AST-616)
2011-08-09 22:12 +0000 [r331248] Richard Mudgett <rmudgett@digium.com>
* channels/chan_iax2.c, apps/app_parkandannounce.c, main/pbx.c,
channels/chan_sip.c, main/features.c: Misc minor items found in
code. * Add some reentrancy protection in pbx.c when creating the
contexts_table hash table. * Fix inverted test in chan_sip.c
conditional code. * Fix uninitialized variable and use of the
wrong variable in chan_iax2.c. * Fix test of return value in
app_parkandannounce.c. Explicitly testing for -1 is bad if the
function does not actually return that value when it fails. *
Fixup some comments and add some curly braces in features.c.
2011-08-09 16:13 +0000 [r331146] Alexandr Anikin <may@telecom-service.ru>
* addons/ooh323c/src/ooLogChan.c, addons/ooh323c/src/ooGkClient.c,
addons/chan_ooh323.c: move ast_cond_signal for admitted call
after all data filled/freed clear all log channels by pointed
number not only first free allocated callToken in ooh323_answer
2011-08-09 15:58 +0000 [r331142] Jason Parker <jparker@digium.com>
* doc/asterisk.8: Regenerate asterisk man page from sgml.
2011-08-08 20:52 +0000 [r331038] Kinsey Moore <kmoore@digium.com>
* res/res_musiconhold.c: In-queue MOH stops after a periodic
announcement If the seek value is past the end of file when
resuming G.722 MOH, MOH will cease to function for the duration
of the MOH session through all starts and stops until saved state
is cleared. Adjusting the code to guarantee a single valid read
(which is already assumed) fixes the bug. (closes issue
ASTERISK-18077) Review: https://reviewboard.asterisk.org/r/1328/
Tested-by: Jonathan Rose <jrose@digium.com>
2011-08-04 20:29 +0000 [r330843] Terry Wilson <twilson@digium.com>
* configure, configure.ac: Make libsrtp instructions more explicit
when linking fails (closes issue ASTERISK-18139)
2011-08-04 19:37 +0000 [r330827] Alexandr Anikin <may@telecom-service.ru>
* addons/ooh323c/src/ooCmdChannel.c,
addons/ooh323c/src/ooGkClient.c: change gk client behaivour on
rrq/grq failures to setup timers and next tries after timeout
instead of complete failure in the ooh323 stack
2011-08-03 15:14 +0000 [r330705-330762] Kinsey Moore <kmoore@digium.com>
* main/Makefile: editing files in main/editline does not ensure
rebuild of libedit.a When editing a source file in main/editline,
the build system does not rebuild libedit.a and uses the already
existing one instead. Adding a PHONY to CHECK_SUBDIR fixes this
problem. (closes issue ASTERISK-16221) Patch-by: Walter Doekes
* channels/chan_dahdi.c, channels/sig_analog.c: Call pickup broken
for DAHDI channels when beginning with # The call pickup feature
did not work on DAHDI devices for anything other than feature
codes beginning with * since all feature codes in chan_dahdi were
originally hard-coded to begin with *. This patch is also applied
to chan_dahdi.c to fix this bug with radio modes. (closes issue
AST-621) Review: https://reviewboard.asterisk.org/r/1336/
2011-08-02 20:51 +0000 [r330648] Kevin P. Fleming <kpfleming@digium.com>
* res/res_jabber.c: Convert an error message to actually be
helpful.
2011-08-02 16:15 +0000 [r330575-330581] David Vossel <dvossel@digium.com>
* channels/chan_iax2.c: Fixes crash in chan_iax2. Fixes crash in
chan_iax2 resulting from an edge case in the way control frames
are queued during calltoken negotiation is complete. (closes
issue ASTERISK-17610) Reported by: mgrobecker
* channels/chan_sip.c: Optimization to buffer initialization fix.
* channels/chan_sip.c: Fixes uninitialized string buffer in log
message. (closes issue ASTERISK-17200) Reported by: lmadsen
2011-08-01 15:22 +0000 [r330433] Kinsey Moore <kmoore@digium.com>
* main/say.c: Incorrect playback for Spanish in some circumstances
When you say the time in spanish and it is 01:00 - 01:59 or 13:00
- 13:59 you must use female pronunciation "1F". The function
"say_date_with_format_es" does not take this in account. (closes
ASTERISK-15016) Patch-by: Luis Jimenez
2011-07-30 23:56 +0000 [r330368] Richard Mudgett <rmudgett@digium.com>
* main/channel.c: Remove some redundant locking code in
ast_do_masquerade(). Also updated some comments.
2011-07-30 15:25 +0000 [r330311] Gregory Nietsky <gregory@distrotech.co.za>
* main/channel.c: prevent double masqurading channels when one is
been hung up and deadlock avoidance is used. There is a race
condition in ast_do_masquerade / ast_hangup (at least) Reported
by me signed off by schmidts with input from David Vossel Review:
https://reviewboard.asterisk.org/r/1323/
2011-07-29 17:18 +0000 [r330203-330213] Sean Bright <sean@malleable.com>
* formats/format_wav.c: Correct the check for O_RDONLY.
* formats/format_wav.c: Only write to wav files that were opened to
be written to.
2011-07-28 21:42 +0000 [r330107] Terry Wilson <twilson@digium.com>
* main/term.c: Make console colors work for TERM=xterm-256color
2011-07-28 17:04 +0000 [r330050] Richard Mudgett <rmudgett@digium.com>
* channels/sig_pri.c: Merged revisions 330033 from
https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
.......... r330033 | rmudgett | 2011-07-28 11:26:38 -0500 (Thu,
28 Jul 2011) | 15 lines Datacalls with B410P fail. Incoming and
outgoing call legs of a data call are using different formats:
a-law, u-law. When the call is bridged, the media stream is run
through translation to convert the media formats. The translation
is bad for data calls. * Make incoming call that does not
explicitly specify u-law or a-law use the DAHDI channel's default
law. The outgoing call always uses the default law from the DAHDI
channel. (closes issue ABE-2800) Patches:
jira_abe_2800_companding.patch (license #5621) patch uploaded by
rmudgett ..........
2011-07-28 15:45 +0000 [r329994] Jason Parker <jparker@digium.com>
* channels/chan_sip.c: Fix a SIP transfer deadlock. The locking in
this function is very scary. There are like 6 structs involved.
(closes issue AST-470)
2011-07-28 15:26 +0000 [r329991] Matthew Nicholson <mnicholson@digium.com>
* res/res_fax.c: check for CONFIG_STATUS_FILE_INVALID when loading
the res_fax config file Patch by: tzafrir Reported by: tzafrir
(closes issue ASTERISK-18161)
2011-07-28 11:34 +0000 [r329895] Sean Bright <sean@malleable.com>
* channels/chan_sip.c: Make the output of Externhost in 'sip show
settings' more consistent.
2011-07-27 19:27 +0000 [r329782] Leif Madsen <lmadsen@digium.com>
* apps/app_confbridge.c: Change support for ConfBridge() in 1.8 to
Extended.
2011-07-27 19:17 +0000 [r329767] Sean Bright <sean@malleable.com>
* Makefile.moddir_rules: Explicitly sort the module list so that
the menuselect lists are sorted. (closes issue ASTERISK-18141)
Reported by: Richard Miller Patches: sort-order.diff uploaded by
seanbright (License #5060) Tested by: leifmadsen
2011-07-27 18:10 +0000 [r329709] Jonathan Rose <jrose@digium.com>
* configs/indications.conf.sample: Fix New Zealand indications
profile based on http://www.telepermit.co.nz/TNA102.pdf (closes
issue ASTERISK-16263) Reported by: richardf Patches:
nz-indications.patch uploaded by richardf (License #6015)
2011-07-27 04:23 +0000 [r329613] Tilghman Lesher <tilghman@meg.abyt.es>
* cdr/cdr_odbc.c: Duration and billsec are swapped in high
resolution time. Closes ASTERISK-18024 Patches:
20110726__ASTERISK-18024.diff by Tilghman Lesher (License 5003)
2011-07-26 14:04 +0000 [r329527-329529] Jonathan Rose <jrose@digium.com>
* apps/app_voicemail.c: Changes sound file for prepend
"then-press-pound" to "vm-then-pound" which is the same prompt,
only it turned out "then-press-pound" was part of extra sounds.
Also, vm is more appropriate anyway.
* main/app.c, apps/app_voicemail.c, include/asterisk/app.h,
configs/voicemail.conf.sample: Fixes some voicemail forwarding
behavior based around prepend mode. Formerly, prepend forwarding
would have the user record a message with no useful prompt and an
expectation for the user to push a button on the phone when
finished recording. If a length of silence was detected instead,
the recording would be canceled and the user would re-enter the
voicemail forwarding menu. Subsequent time-outs in prepend
recording would also bug out in the sense that they would write
over the original message and get sent to the recipient
regardless of whether they timed out or were accepted. This patch
fixes this issue and adds a prompt which will be played after a
timeout informing the user that they needed to press a button.
Currently, the sound files that we have are somewhat inadquate
for this, so after the call we simply have Allison say "Please
try again. Then press pound." which actually relies on two
separate sound files. Just one would be more appropriate.
reporter: Vlad Povorozniuc Review:
https://reviewboard.asterisk.org/r/1327/
2011-07-25 19:49 +0000 [r329471] Paul Belanger <pabelanger@digium.com>
* main/enum.c: Decrease verbose messages to debug, to help clean up
CLI.
2011-07-22 21:10 +0000 [r329144-329333] Richard Mudgett <rmudgett@digium.com>
* main/pbx.c: Fix memory leak in an allocation error path of
handle_statechange(). * Make use buffer accessor function in
handle_statechange() rather than directly accessing the struct
member. * Make use less redundant loop construct for iterating
over hints.
* main/pbx.c: Deadlocks dealing with dialplan hints during reload.
There are two remaining different deadlocks reported dealing with
dialplan hints. The deadlock in ASTERISK-17666 is caused by
invalid locking order in ast_remove_hint(). The hints container
must be locked before the hint object. The deadlock in
ASTERISK-17760 is caused by a catch-22 situation in
handle_statechange(). The deadlock is caused by not having the
conlock before calling the watcher callbacks. Unfortunately,
having that lock causes a different deadlock as reported in
ASTERISK-16961. * Fixed ast_remove_hint() locking order. * Made
handle_statechange() no longer call the watcher callbacks holding
any locks that matter. * Made hint ao2 destructor do the watcher
callbacks for extension deactivation to guarantee that they get
called. * Fixed hint reference leak in ast_add_hint() if the
callback container constructor failed. * Fixed hint reference
leak in complete_core_show_hint() for every hint it found for CLI
tab completion. * Adjusted locking in
ast_merge_contexts_and_delete() for safety. * Added
context_merge_lock to prevent ast_merge_contexts_and_delete() and
handle_statechange() from interfering with each other. * Fixed
ast_change_hint() not taking into account that the extension is
used for the hash key. (closes issue ASTERISK-17666) Reported by:
irroot Tested by: irroot JIRA SWP-3318 (closes issue
ASTERISK-17760) Reported by: Byron Clark Tested by: irroot JIRA
SWP-3393 Review: https://reviewboard.asterisk.org/r/1313/
* channels/chan_dahdi.c, configs/chan_dahdi.conf.sample: Document
parkinglot in chan_dahdi.conf.sample. * Document existing feature
in chan_dahdi.conf.sample. * Remove some dead code related to the
parkinglot option.
* apps/app_directed_pickup.c: Update PickupChan documentation. The
PickupChan uses the ampersand as the argument separator. Was
documented as: PickupChan(channel[,channel2[,...][,options]])
Fixed documentation to:
PickupChan(Technology/Resource[&Technology2/Resource2[&...]][,options])
This is a continuation of ASTERISK-17494 for v1.8 and later.
(closes issue ASTERISK-18144) Reported by: Erik Smith Patches:
pickupchan_ducumentation-v2.patch (License #6263) patch uploaded
by Erik Smith Tested by: Erik Smith
* main/features.c: Dialplan bridge() app mutex 'current_dest_chan'
freed more times than we've locked! This appears to be a leftover
from when ast_channel was converted to ao2 objects. Simply
removed the extraneous unlock. (closes issue ASTERISK-17772)
2011-07-20 21:20 +0000 [r329027] Paul Belanger <pabelanger@digium.com>
* UPGRADE.txt: Asterisk now requires libpri 1.4.11+ for PRI
support.
2011-07-20 20:52 +0000 [r329012] Richard Mudgett <rmudgett@digium.com>
* channels/sig_pri.c, channels/sig_pri.h, channels/chan_dahdi.c:
Backport useful CLI "pri show channels" command to v1.8. The "pri
show channels" command is useful for debuging to see if there are
any stuck B channels. .......... r307964 | rmudgett | 2011-02-15
15:42:55 -0600 (Tue, 15 Feb 2011) | 9 lines Add CLI "pri show
channels" command. List the current mapping of DAHDI B channels
to Asterisk channel names and which calls are on hold or
call-waiting. Calls on hold or call-waiting are not associated
with any B channel. JIRA LIBPRI-27 JIRA SWP-2547 ..........
r308205 | rmudgett | 2011-02-17 14:21:56 -0600 (Thu, 17 Feb 2011)
| 1 line Add more verbage to CLI command 'pri show channels'
usage. .......... r312579 | rmudgett | 2011-04-04 11:17:58 -0500
(Mon, 04 Apr 2011) | 59 lines Change also updates 'pri show
channels' command with the "chan idle" column to report if a
channel is available for use.
2011-07-20 20:16 +0000 [r328987] Terry Wilson <twilson@digium.com>
* tests/test_netsock2.c: We can't guarantee an eth0 is present
FreeBSD test fails on this case presumably because there is no
eth0 on the test machine. Better to just remove this test for
now.
2011-07-20 19:00 +0000 [r328935] Kinsey Moore <kmoore@digium.com>
* channels/chan_sip.c: Inband DTMF regression The functionality of
inband DTMF in chan_sip relied upon
ast_rtp_instance_dtmf_mode_get/set not working properly to avoid
calling ast_rtp_instance_dtmf_begin/end on RTP streams with
inband DTMF. According to documentation,
ast_rtp_instance_dtmf_begin/end is meant only for RFC2833 DTMF,
never inband. This fixes the regression introduced in revision
328823.
2011-07-19 21:29 +0000 [r328878] Kevin P. Fleming <kpfleming@digium.com>
* sounds/Makefile, Makefile, Makefile.moddir_rules: Revert partial
attempt at handling pathnames with spaces. Revision 299794
attempted to improve the build system to be able to handle
pathnames (primarily DESTDIR) with spaces in them, since this is
common on some platforms (including Mac OSX). Unfortunately, the
changes were incomplete and did not actually provide the desired
behavior, and as a side effect the functionality that ensured
that stale headers in the Asterisk 'include' directory were
removed got broken. In addition, the check for stale (and
possibly incompatible) modules in the Asterisk 'modules'
directory also got broken, and would never report any stale
modules. Users upgrading to this version or later versions would
then see unexpected module load errors. Since there are few users
who actually want to install Asterisk into paths that contain
spaces, and a proper fix for the build system would take many
hours, the best solution for now is to just revert the partial
solution.
2011-07-19 17:57 +0000 [r328770-328823] Kinsey Moore <kmoore@digium.com>
* include/asterisk/rtp_engine.h, res/res_rtp_asterisk.c,
main/rtp_engine.c, channels/chan_sip.c: RTP bridge away with
inband DTMF and feature detection When deciding whether Asterisk
was allowed to bridge the call away from the core, chan_sip did
not take into account the usage of features on dialed channels
that require monitoring of DTMF on channels utilizing inband
DTMF. This would cause Asterisk to allow the call to be locally
or remotely bridged, preventing access to the data required to
detect activations of such features. (closes 17237) Review:
https://reviewboard.asterisk.org/r/1302/
* apps/app_meetme.c: MeetMe requests a PIN twice in some
circumstances If a call to MeetMe includes both the dynamic(D)
and always request PIN(P) options, MeetMe will ask for the PIN
two times: once for creating the conference and once for entering
the conference. This behavior was introduced in rev 311616 when
adding the CONFFLAG_ALWAYSPROMPT option to the logic branch
controlling PIN entry for joining a conference. (closes AST-601)
Review: https://reviewboard.asterisk.org/r/1305/
2011-07-19 01:35 +0000 [r328716] Terry Wilson <twilson@digium.com>
* tests/test_linkedlists.c (added), include/asterisk/linkedlists.h:
Make AST_LIST_REMOVE safer AST_LIST_REMOVE shouldn't modify the
element passed in if it isn't found. This commit also adds linked
list unit tests. Review: https://reviewboard.asterisk.org/r/1321/
2011-07-18 20:47 +0000 [r328593-328663] Mark Murawki <markm@intellasoft.net>
* apps/app_dial.c: app_dial may double free a channel datastore
When starting a call with originate, and having the callee
channel run Bridge() on pickup, we will double free the
dialed_interface_info datastore, causing a crash. Make sure to
check if the datastore still exists before trying to free it.
(closes issue ASTERISK-17917) Reported by: Mark Murawski Tested
by: Mark Murawski
* channels/chan_sip.c: If the sip private structure is null,
sip_setoption() will defref the null pointer and crash. Ideally,
sip_setoption shouldn't be called if there is a lack of a sip
private structure. But this will fix a crash. (closes issue
ASTERISK-17909) Reported by: Mark Murawski Tested by: Mark
Murawski
* main/asterisk.c: Fixed invalid read and null pointer deref on
asterisk shutdown. In some cases when starting asterisk with -c
and hitting control-c to shutdown, there will be an invalid read
and null pointer deref causing a crash. (closes issue
ASTERISK-17927) Reported by: Mark Murawski Tested by: Mark
Murawski, Kinsey Moore, Tilghman Lesher
2011-07-18 07:10 +0000 [r328540] Tilghman Lesher <tilghman@meg.abyt.es>
* funcs/func_odbc.c: Typo
2011-07-15 20:41 +0000 [r328446] Leif Madsen <lmadsen@digium.com>
* apps/app_macro.c, channels/chan_jingle.c, apps/app_dahdibarge.c,
apps/app_readfile.c, apps/app_setcallerid.c,
channels/chan_vpb.cc, apps/app_meetme.c, cdr/cdr_sqlite.c,
channels/chan_h323.c: Revert changes to defaultenabled state for
modules in Asterisk 1.8
2011-07-15 19:22 +0000 [r328427] Alexandr Anikin <may@telecom-service.ru>
* addons/ooh323c/src/ooGkClient.c: small gk processing fixes: -
decrease for 1 second registration ttl for very low expirations
(some providers expire few earlier than TTL) - delete rrq and
registration expire timers on URQ received as we make new
registration.
2011-07-14 23:12 +0000 [r328302] Richard Mudgett <rmudgett@digium.com>
* channels/chan_sip.c: Missing SIP pvt and channel unlock in
sip_set_rtp_peer(). Regression introduced by -r326144. Add
missing SIP pvt and channel unlock in sip_set_rtp_peer().
2011-07-14 20:13 +0000 [r328209] Leif Madsen <lmadsen@digium.com>
* apps/app_image.c, res/res_http_post.c, formats/format_wav_gsm.c,
utils/stereorize.c, pbx/pbx_loopback.c, funcs/func_shell.c,
main/features.c, channels/chan_alsa.c, apps/app_externalivr.c,
formats/format_jpeg.c, res/res_speech.c, formats/format_gsm.c,
apps/app_milliwatt.c, formats/format_g719.c,
apps/app_saycounted.c, apps/app_fax.c, apps/app_echo.c,
funcs/func_math.c, channels/chan_agent.c, apps/app_dahdiras.c,
utils/astman.c, res/res_ael_share.c, apps/app_transfer.c,
apps/app_playback.c, res/res_config_curl.c, funcs/func_curl.c,
apps/app_waitforring.c, channels/chan_misdn.c, tests/test_skel.c,
addons/cdr_mysql.c, codecs/codec_ilbc.c, apps/app_zapateller.c,
apps/app_chanspy.c, apps/app_cdr.c, tests/test_substitution.c,
funcs/func_md5.c, utils/muted.c, tests/test_gosub.c,
funcs/func_sysinfo.c, funcs/func_vmcount.c, funcs/func_sha1.c,
cdr/cdr_radius.c, formats/format_siren7.c,
apps/app_controlplayback.c, funcs/func_config.c, main/manager.c,
bridges/bridge_builtin_features.c, funcs/func_volume.c,
cdr/cdr_sqlite.c, funcs/func_aes.c, funcs/func_frame_trace.c,
tests/test_devicestate.c, res/res_agi.c, tests/test_astobj2.c,
apps/app_confbridge.c, apps/app_ivrdemo.c,
res/res_clioriginate.c, res/res_calendar_icalendar.c,
funcs/func_dialplan.c, funcs/func_db.c,
tests/test_ast_format_str_reduce.c, res/res_fax.c,
res/res_limit.c, apps/app_festival.c, apps/app_waitforsilence.c,
apps/app_waituntil.c, channels/chan_console.c,
apps/app_getcpeid.c, apps/app_queue.c, funcs/func_global.c,
funcs/func_extstate.c, channels/chan_usbradio.c,
apps/app_flash.c, codecs/codec_ulaw.c, channels/chan_nbs.c,
formats/format_g729.c, funcs/func_dialgroup.c, funcs/func_env.c,
res/res_timing_dahdi.c, funcs/func_strings.c,
res/res_calendar_caldav.c, apps/app_chanisavail.c,
formats/format_sln16.c, apps/app_ices.c, apps/app_exec.c,
bridges/bridge_multiplexed.c, cel/cel_odbc.c,
formats/format_pcm.c, pbx/pbx_ael.c, formats/format_h263.c,
cdr/cdr_manager.c, res/res_clialiases.c, funcs/func_sprintf.c,
tests/test_app.c, apps/app_softhangup.c, codecs/codec_g726.c,
apps/app_morsecode.c, utils/smsq.c, bridges/bridge_simple.c,
tests/test_sched.c, apps/app_talkdetect.c, apps/app_db.c,
res/res_calendar_ews.c, funcs/func_callcompletion.c,
tests/test_acl.c, funcs/func_cdr.c, utils/ael_main.c,
utils/streamplayer.c, res/res_calendar.c, cel/cel_radius.c,
channels/chan_vpb.cc, res/res_snmp.c, apps/app_dictate.c,
apps/app_authenticate.c, res/res_phoneprov.c, funcs/func_logic.c,
res/res_jabber.c, funcs/func_uri.c,
funcs/func_audiohookinherit.c, res/res_config_odbc.c,
funcs/func_odbc.c, res/res_realtime.c, codecs/codec_resample.c,
formats/format_h264.c, apps/app_rpt.c, channels/chan_mgcp.c,
tests/test_amihooks.c, codecs/codec_lpc10.c, channels/chan_sip.c,
cdr/cdr_syslog.c, funcs/func_lock.c, res/res_adsi.c,
utils/conf2ael.c, tests/test_pbx.c, apps/app_channelredirect.c,
formats/format_vox.c, res/res_stun_monitor.c, tests/test_aoc.c,
formats/format_g723.c, utils/extconf.c, tests/test_poll.c,
addons/chan_ooh323.c, cdr/cdr_sqlite3_custom.c,
funcs/func_module.c, apps/app_sayunixtime.c,
cdr/cdr_adaptive_odbc.c, res/res_smdi.c, tests/test_time.c,
apps/app_skel.c, funcs/func_srv.c, apps/app_amd.c,
pbx/pbx_realtime.c, apps/app_url.c, apps/app_dial.c,
apps/app_page.c, channels/chan_bridge.c, apps/app_privacy.c,
codecs/codec_speex.c, apps/app_disa.c, res/res_mutestream.c,
res/res_monitor.c, apps/app_macro.c, res/res_timing_kqueue.c,
res/res_fax_spandsp.c, channels/chan_unistim.c,
funcs/func_base64.c, addons/app_mysql.c,
channels/chan_multicast_rtp.c, apps/app_meetme.c,
utils/hashtest.c, res/res_musiconhold.c, apps/app_followme.c,
res/res_config_sqlite.c, cdr/cdr_csv.c,
tests/test_security_events.c, formats/format_ilbc.c,
funcs/func_enum.c, channels/chan_phone.c,
tests/test_stringfields.c, funcs/func_groupcount.c,
tests/test_locale.c, addons/chan_mobile.c, cdr/cdr_custom.c,
res/res_security_log.c, apps/app_parkandannounce.c,
apps/app_while.c, apps/app_jack.c, res/res_rtp_asterisk.c,
apps/app_nbscat.c, codecs/codec_a_mu.c, tests/test_dlinklists.c,
res/res_convert.c, pbx/pbx_lua.c, utils/astcanary.c,
channels/chan_oss.c, tests/test_strings.c, res/res_srtp.c,
cdr/cdr_tds.c, res/res_timing_pthread.c,
apps/app_directed_pickup.c, channels/chan_h323.c,
cel/cel_sqlite3_custom.c, apps/app_senddtmf.c,
funcs/func_callerid.c, addons/app_saycountpl.c, cel/cel_pgsql.c,
funcs/func_speex.c, apps/app_dahdibarge.c, channels/chan_local.c,
tests/test_logger.c, apps/app_record.c, apps/app_playtones.c,
bridges/bridge_softmix.c, apps/app_alarmreceiver.c,
channels/chan_iax2.c, res/res_pktccops.c,
res/res_rtp_multicast.c, channels/chan_dahdi.c, pbx/pbx_spool.c,
funcs/func_pitchshift.c, channels/chan_skinny.c,
apps/app_dumpchan.c, main/http.c, cdr/cdr_odbc.c,
utils/refcounter.c, res/res_calendar_exchange.c, res/res_ais.c,
codecs/codec_g722.c, tests/test_expr.c, funcs/func_timeout.c,
cel/cel_tds.c, formats/format_wav.c, formats/format_ogg_vorbis.c,
funcs/func_cut.c, apps/app_speech_utils.c, apps/app_sendtext.c,
funcs/func_channel.c, utils/hashtest2.c, pbx/pbx_config.c,
funcs/func_iconv.c, apps/app_mixmonitor.c, formats/format_g726.c,
res/res_odbc.c, apps/app_voicemail.c, tests/test_heap.c,
addons/format_mp3.c, formats/format_sln.c, apps/app_readexten.c,
apps/app_userevent.c, codecs/codec_gsm.c, channels/chan_gtalk.c,
cdr/cdr_pgsql.c, tests/test_func_file.c, apps/app_setcallerid.c,
apps/app_osplookup.c, cel/cel_manager.c, cel/cel_custom.c,
tests/test_utils.c, apps/app_minivm.c, apps/app_mp3.c,
res/res_timing_timerfd.c, apps/app_directory.c,
res/res_config_ldap.c, formats/format_siren14.c,
apps/app_adsiprog.c, res/res_config_pgsql.c, apps/app_read.c,
funcs/func_version.c, codecs/codec_alaw.c, agi/eagi-test.c,
res/res_crypto.c, apps/app_originate.c, channels/chan_jingle.c,
apps/app_forkcdr.c, funcs/func_blacklist.c, pbx/pbx_dundi.c,
apps/app_sms.c, apps/app_stack.c, funcs/func_devstate.c,
apps/app_verbose.c, addons/res_config_mysql.c,
utils/check_expr.c, funcs/func_rand.c, apps/app_readfile.c,
codecs/codec_adpcm.c, apps/app_test.c, tests/test_event.c:
Introduce <support_level> tags in MODULEINFO. This change
introduces MODULEINFO into many modules in Asterisk in order to
show the community support level for those modules. This is used
by changes committed to menuselect by Russell Bryant recently
(r917 in menuselect). More information about the support level
types and what they mean is available on the wiki at
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Module+Support+States
2011-07-14 19:21 +0000 [r328205] Jonathan Rose <jrose@digium.com>
* res/res_monitor.c: Monitor application arguments requirements
fixed. Monitor was requiring options in spite of no individual
option on Monitor being required. Review:
https://reviewboard.asterisk.org/r/1320/
2011-07-13 18:46 +0000 [r328014] Richard Mudgett <rmudgett@digium.com>
* configs/features.conf.sample: Add ATXFER_NULL_TECH note in
features.conf.sample.
2011-07-12 22:53 +0000 [r327950] Kevin P. Fleming <kpfleming@digium.com>
* main/manager.c: Correct double-free situation in manager output
processing. The process_output() function calls ast_str_append()
and xml_translate() on its 'out' parameter, which is a pointer to
an ast_str buffer. If either of these functions need to
reallocate the ast_str so it will have more space, they will free
the existing buffer and allocate a new one, returning the address
of the new one. However, because process_output only receives a
pointer to the ast_str, not a pointer to its caller's variable
holding the pointer, if the original ast_str is freed, the caller
will not know, and will continue to use it (and later attempt to
free it). (reported by jkroon on #asterisk-dev)
2011-07-12 20:07 +0000 [r327890] Matthew Nicholson <mnicholson@digium.com>
* apps/app_directory.c: search in the current context for 'a' and
'o' instead of 'default'
2011-07-12 19:38 +0000 [r327888] Jason Parker <jparker@digium.com>
* Makefile: Fix uninstall target, so that modules dir gets cleared
again.
2011-07-12 19:10 +0000 [r327852] Matthew Jordan <mjordan@digium.com>
* apps/app_voicemail.c: Added additional checks for mailbox /
password beginning with '*' character A bug existed such that if
a user entered a password with '*', and the extension 'a' did not
exist, an invalid mailbox would be created and the user
authenticated. The code was changed to prevent this from
occurring, and to prevent users from having mailboxes or
passwords defined that begin with the '*' character. (closes
issue ASTERISK-17443) Reported by: Kevin Scott Adams Tested by:
Matt Jordan Review: https://reviewboard.asterisk.org/r/1316/
2011-07-12 15:35 +0000 [r327793] Tilghman Lesher <tilghman@meg.abyt.es>
* tests/test_substitution.c: Use 'printf' (POSIX issue 4) instead
of 'echo -n', for portability. The problem with using 'echo -n'
is that it is not portable. While BSD systems required that the
'-n' option be removed and interpreted, System V required that
all strings should be echoed with no interpretation of options.
This fundamental difference of behavior means that it is never
possible to use the '-n' flag to echo in tests which are meant to
be portable. In this case, on Mac OS X 10.6, the /bin/sh shell
builtin 'echo' uses the System V semantics of the command, and
thus the SHELL test failed on that platform.
http://pubs.opengroup.org/onlinepubs/009695399/utilities/echo.html#tag_04_41_16
2011-07-11 19:41 +0000 [r327682] Terry Wilson <twilson@digium.com>
* include/asterisk/jingle.h, channels/chan_gtalk.c: Update
chan_gtalk to work with changed GMail-based calls The messages
sent by the GMail client have changed, but include the old-style
messages as well. This patch checks for this case and uses the
old-style offer. (closes issue ASTERISK-18084) Review:
https://reviewboard.asterisk.org/r/1312/
2011-07-11 13:53 +0000 [r327512] Matthew Nicholson <mnicholson@digium.com>
* main/pbx.c, tests/test_substitution.c: reset our buffer each
iteration when doing variable substitution
2011-07-11 10:56 +0000 [r327411-327412] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
* main/Makefile: Properly building the Debian armhf (HardFloat)
port. Remove the line that should have been removed in r327411.
* main/Makefile: fix building the Debian armhf (HardFloat) port
Fixes
http://buildd.debian-ports.org/status/fetch.php?pkg=asterisk&arch=armhf&ver=1%3A1.8.4.4~dfsg-2&stamp=1309935385
(Missing pthreads)
2011-07-08 22:27 +0000 [r327258] Jason Parker <jparker@digium.com>
* main/db1-ast/mpool, addons, cdr, formats, codecs/gsm/src, funcs,
addons/ooh323c/src, bridges, codecs/lpc10, main/db1-ast/btree,
codecs/g722, main, main/db1-ast/recno, channels/sip, res, pbx,
res/ael, channels, main/stdtime, addons/ooh323c/src/h323, codecs,
utils, main/db1-ast/hash, cel, apps, main/db1-ast/db: Add .o
files to svn:ignore property, since it's only ignored if locally
configured to do so.
2011-07-08 21:41 +0000 [r327211] Richard Mudgett <rmudgett@digium.com>
* channels/chan_sip.c: INVITE 403 Forbidden response always
retransmits the maximum times. Asterisk sends a 403 Forbidden
response if authentication fails for an INVITE as required.
However, it ignores the ACK and keeps retransmitting the
response. * Made not delete the to-tag in the dialog so the
expected ACK can be matched with the dialog and stop the
retransmissions.
2011-07-08 19:52 +0000 [r327106] Matthew Nicholson <mnicholson@digium.com>
* main/pbx.c, tests/test_substitution.c: Reset our ast_str before
passing it on to dialplan function backends. It is possible for a
dialplan backend to not modify the given buffer or ast_str and
still return success. This causes any previous value stored in
the buffer to be used as if the new function call provided it.
Some functions also append to the given buffer assuming it is
empty. The test_substitution unit test has also been modified to
detect this problem. (closes issue ASTERISK-17878)
2011-07-08 16:00 +0000 [r327044-327046] Russell Bryant <russell@digium.com>
* tests/test_netsock2.c: Fix an error and add more log message info
to help see why this fails on FreeBSD.
* channels/chan_dahdi.c: Resolve some set-but-unused-variable
warnings.
2011-07-08 01:08 +0000 [r326985] Richard Mudgett <rmudgett@digium.com>
* main/pbx.c: Some code cleanup in pbx.c * Mostly comment and
format changes. * ast_context_remove_extension_callerid() and
ast_add_extension_nolock() will write lock the found specific
context. * ast_context_find() will now tolerate a NULL name. *
Eliminated some inlined versions of find_context() and
find_context_locked().
2011-07-07 19:17 +0000 [r326830] Tilghman Lesher <tilghman@meg.abyt.es>
* res/res_http_post.c: libgen.h is also needed on Darwin for
basename(3)
2011-07-07 16:04 +0000 [r326689] Jonathan Rose <jrose@digium.com>
* res/res_config_odbc.c: res_odbc patch by tilghman to fix integers
with null values Addresses some improper sql statements in
res_odbc that would cause an update to fail on realtime peers due
to trying to set as "(NULL)" rather than an actual NULL. (closes
issue #1922STERISK-17791) Reported by: marcelloceschia Patches:
20110505__issue19223.diff.txt uploaded by tilghman (license 14)
2011-07-07 15:28 +0000 [r326681-326683] Matthew Nicholson <mnicholson@digium.com>
* channels/chan_sip.c: use sips: or sip: depending on the transport
in use when building reply digest URIs
* channels/chan_sip.c: make the uri parameter used in reply digests
more standards compliant in certain cases by prepending "sip:" or
"sips:" to it
2011-07-06 15:26 +0000 [r326484] David Vossel <dvossel@digium.com>
* res/res_timing_timerfd.c: Reverts fix for timerfd locking issue.
jrose discovered a performance issue with this fix that prevents
his analog phones from working when using timerfd as a timing
source. Until it is understood what is causing this performance
problem, this patch is being reverted.
2011-07-06 14:35 +0000 [r326411-326469] Tilghman Lesher <tilghman@meg.abyt.es>
* pbx/pbx_dundi.c, channels/chan_gtalk.c, apps/app_queue.c,
channels/chan_iax2.c, res/res_jabber.c, apps/app_stack.c,
channels/chan_mgcp.c, apps/app_voicemail.c,
channels/chan_jingle.c, channels/chan_dahdi.c,
funcs/func_speex.c, channels/chan_sip.c, codecs/codec_speex.c,
funcs/func_aes.c: Removing type attributes, as a change to
menuselect makes them no longer necessary.
* pbx/pbx_dundi.c, channels/chan_gtalk.c, apps/app_queue.c,
channels/chan_iax2.c, res/res_jabber.c, apps/app_stack.c,
channels/chan_mgcp.c, apps/app_voicemail.c,
channels/chan_jingle.c, channels/chan_dahdi.c,
funcs/func_speex.c, channels/chan_sip.c, codecs/codec_speex.c,
funcs/func_aes.c: Add the attribute "type" to each "<use>" for
menuselect. This matters only when autoconf fails to detect that
weak linking is supported. External optional dependencies will
become optional in both cases, as they are removed at compile
time when not detected. However, runtime-optional modules are
made mandatory when weak linking is not found. This change
affects only the external optional dependencies; previously, they
were incorrectly required when weak linking support was not
detected. Patches: 20110702__issue18062__asterisk_trunk.diff.txt
by tilghman (License #5003) Tested by: iasgoscouk
2011-07-05 17:22 +0000 [r326291] Richard Mudgett <rmudgett@digium.com>
* channels/sip/include/sip.h, channels/chan_sip.c: Used auth=
parameter freed during "sip reload" causes crash. If you use the
auth= parameter and do a "sip reload" while there is an ongoing
call. The peer->auth data points to free'd memory. The patch does
several things: 1) Puts the authentication list into an ao2
object for reference counting to fix the reported crash during a
SIP reload. 2) Converts the authentication list from open coding
to AST list macros. 3) Adds display of the global authentication
list in "sip show settings". (closes issue ASTERISK-17939)
Reported by: wdoekes Patches: jira_asterisk_17939_v1.8.patch
(license #5621) patch uploaded by rmudgett Review:
https://reviewboard.asterisk.org/r/1303/ JIRA SWP-3526
2011-07-05 13:23 +0000 [r326209] Matthew Jordan <mjordan@digium.com>
* main/file.c: Updated filestream destructor to block until move is
complete when cache is used When a cache directory is used, the
process is forked and a mv command is executed to move the
temporary file to the permanent location. This caused issues with
voicemail, where a race condition occurred when the parent
expected the file to be in the permanent location prior to the mv
command completing. The parent process is now blocked until the
mv command completes. (closes issue ASTERISK-17724) Reported by:
Adiren P. Tested by: mjordan
2011-07-01 21:07 +0000 [r326144] Richard Mudgett <rmudgett@digium.com>
* channels/chan_sip.c: Better way to get chan and pvt lock for
issue ASTERISK-17431. Redoes -r308945 for issue ASTERISK-17431
deadlock fix for sip_set_udptl_peer() and sip_set_rtp_peer(). *
Lock the channels in the defined order and avoid the need for a
deadlock avoidance loop. * Lock the channel before getting the
pointer to the private structure to be sure that the pointer will
not change due to a masquerade or channel hangup. * To preserve
sanity, check that chan and p->owner are the same. (Pointer
rearangements should not happen without the protection of locks
because bad things tend to happen otherwise.)
2011-06-30 20:39 +0000 [r325935] Richard Mudgett <rmudgett@digium.com>
* configs/sip.conf.sample, channels/chan_sip.c: Misc minor changes
in chan_sip. * Add load failure exit if primary SIP container(s)
could not get created in chan_sip.c:load_module(). * Removed a
redundant static prototype. * Some typos. * Some whitespace.
2011-06-30 20:09 +0000 [r325877] Matthew Jordan <mjordan@digium.com>
* apps/app_voicemail.c: Patched voicemail user option for emailbody
/ emailsubject Incorporated changes per ASTERISK-16795; updated
unit tests to check for vmu->emailbody / vmu->emailsubject
(closes issue ASTERISK-16795) Reported by: mdeneen Tested by:
mjordan
2011-06-30 19:17 +0000 [r325821] Jonathan Rose <jrose@digium.com>
* res/res_musiconhold.c: Fixes an issue with Music on Hold classes
losing files in playlist when realtime is used. The bug occurs
rather intermittently and I relied on the reporters to test the
patch. After a sanity check and some testing, I'm giving it an
OK. (closes issue ASTERISK-17875) Reported by: David Cunningham
Patches: res_musiconhold.c.mohrt17875_v1 uploaded by Igor
Goncharovsky (license #5009)
2011-06-29 21:49 +0000 [r325740] Kinsey Moore <kmoore@digium.com>
* channels/sip/include/sip.h, channels/chan_sip.c: chan_sip:
cleanup from the introduction of ast_str Remove the length field
from sip_req and sip_pkt in chan_sip since they are redundant
(ast_str holds its own length) and refactor the necessary
functions. Review: https://reviewboard.asterisk.org/r/1281/
2011-07-11 Leif Madsen <lmadsen@digium.com>
* Asterisk 1.8.5.0 Released.
* r326484 | dvossel | 2011-07-06 10:26:49 -0500 (Wed, 06 Jul 2011)
Reverts fix for timerfd locking issue.
jrose discovered a performance issue with this
fix that prevents his analog phones from working
when using timerfd as a timing source. Until
it is understood what is causing this performance
problem, this patch is being reverted.
2011-06-29 Leif Madsen <lmadsen@digium.com>
* Asterisk 1.8.5-rc1 Released.
2011-06-29 18:59 +0000 [r325673] David Vossel <dvossel@digium.com>
* res/res_timing_timerfd.c: Fixes timerfd locking issue. (closes
ASTERISK-17867, ASTERISK-17415) Patches: fix uploaded by kobaz
https://reviewboard.asterisk.org/r/1255/
2011-06-29 18:16 +0000 [r325610-325614] Richard Mudgett <rmudgett@digium.com>
* apps/app_queue.c: Fixed some error exit cleanup in app_queue.c. *
Fixed error exit cleanup in app_queue.c copy_rules() and
reload_queue_rules().
* apps/app_queue.c: Response to QueueRule manager command does not
contain ActionID if it was specified. * Add ActionID support as
documented for the QueueRule AMI action. * Remove documentation
for ActionID with the Queues AMI action. The output does not
follow normal AMI response output and there is no place to put an
ActionID header. (closes issue AST-602) Reported by: Vlad
Povorozniuc Patches: jira_ast_602_v1.8.patch (license #5621)
patch uploaded by rmudgett Tested by: Vlad Povorozniuc, rmudgett
Review: https://reviewboard.asterisk.org/r/1295/ JIRA SWP-3575
2011-06-29 16:18 +0000 [r325537-325545] Matthew Nicholson <mnicholson@digium.com>
* main/channel.c: make framehooks prevent native bridging (for real
this time)
* apps/app_dial.c, main/rtp_engine.c: don't do native/remote
bridging if a framehook is active on the channel
2011-06-28 21:50 +0000 [r325416] Kevin P. Fleming <kpfleming@digium.com>
* channels/chan_sip.c: Fix random misspelling noticed on
asterisk-users.
2011-06-28 20:31 +0000 [r325339] David Vossel <dvossel@digium.com>
* channels/chan_sip.c: Fixes locking inversion caused by holding
sip pvt lock during async_goto. (closes ASTERISK-17352)
2011-06-28 20:07 +0000 [r325279] Terry Wilson <twilson@digium.com>
* /, channels/chan_sip.c: Merged revisions 325277 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r325277 | twilson | 2011-06-28 15:06:16 -0500
(Tue, 28 Jun 2011) | 9 lines Merged revisions 325275 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r325275 | twilson | 2011-06-28 15:03:19 -0500 (Tue, 28
Jun 2011) | 2 lines Don't leak SIP username information ........
................
2011-06-28 17:30 +0000 [r325212] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c: Use the device name and not the channel
name to initialize the device state. Correct ASTERISK-11323
implementation as I don't see how it ever worked as claimed when
it used the channel name and not the device name. (issue
ASTERISK-11323)
2011-06-28 15:46 +0000 [r325152] Jonathan Rose <jrose@digium.com>
* res/res_musiconhold.c: Fixes moh reload breaking custom mode moh
classes when the config file is untouched (closes issue
ASTERISK-17730) Reported by: sdolloff
2011-06-28 15:12 +0000 [r325091] Leif Madsen <lmadsen@digium.com>
* build_tools/prep_tarball: Remove line from prep_tarball that
kills mkrelease.
2011-06-27 16:30 +0000 [r324955] Tilghman Lesher <tilghman@meg.abyt.es>
* main/asterisk.c: Save and restore errno from within signal
handlers. This is recommended by the POSIX standard, as well as
by the sigaction(2) manpage for various platforms that we support
(e.g. Mac OS X).
2011-06-27 15:37 +0000 [r324914] Richard Mudgett <rmudgett@digium.com>
* channels/chan_sip.c: When subscribing MWI to an unsolicited
mailbox the first notification is incorrect. A remote peer
subscribed to MWI with the unsolicited option and a local phone
subscribed to the remote mailbox. The notify message-summary
events are sent correctly except for the first one when
subscribing, which will always be 0. This means the phone MWI
indicator will be wrong until the mailbox read/unread count
changes and the event is fired. Looks like this is a regression
from ASTERISK-16149. * Fix the logic to check the cache and if
allowed then fallback to manually counting mailbox messages.
(closes issue ASTERISK-17997) Reported by: rsw686 Patches:
jira_asterisk_17997_v1.8.patch (license #5621) uploaded by
rmudgett Tested by: rsw686 JIRA SWP-3551
2011-06-24 20:46 +0000 [r324849] Richard Mudgett <rmudgett@digium.com>
* pbx/pbx_config.c: Syntax errors in dialplan do not display the
file name. When issuing the CLI command "dialplan reload" syntax
errors and warnings are displayed on the console. The offending
line number is displayed on the console, but the file name is not
displayed. Errors caught in main/config.c do display the file
name. (closes issue ASTERISK-17985) Reported by: ulogic Patches:
pbx_config.patch uploaded by ulogic (License #5685) modified
format Tested by: rmudgett JIRA SWP-3554
2011-06-24 16:48 +0000 [r324768] Jonathan Rose <jrose@digium.com>
* include/asterisk/logger.h: DTMF wasn't being logged on connected
consoles when enabled in logger.conf Previously in order for DTMF
to be logged in a connected console session, the user would have
to do logger set channel DTMF on. This corrects that so that it
is on by default. This issue was caused by an off by one error
incurred by a logger level count of 6 in logger.h where it should
have been 7. (closes issue: ASTERISK-17974) Reported by: Luke H
2011-06-23 18:31 +0000 [r324685] David Vossel <dvossel@digium.com>
* channels/sip/reqresp_parser.c: Fixes sip crash when calling
remove_uri_parameters with NULL AST-2011-009 (closes issue
ASTERISK-18017) Reported by: jaredmauch
2011-06-23 18:29 +0000 [r324678] Kinsey Moore <kmoore@digium.com>
* /, channels/chan_sip.c: Merged revisions 324643 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r324643 | kmoore | 2011-06-23 13:21:12 -0500 (Thu, 23 Jun 2011) |
4 lines Addresses AST-2011-008, memory corruption and remote
crash in SIP driver. AST-2011-008 ........
2011-06-23 18:23 +0000 [r324652] David Vossel <dvossel@digium.com>
* channels/chan_iax2.c, include/asterisk/frame.h, /,
main/features.c: Merged revisions 324634 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r324634 | dvossel | 2011-06-23 13:18:46 -0500
(Thu, 23 Jun 2011) | 13 lines Merged revisions 324627 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r324627 | dvossel | 2011-06-23 13:16:52 -0500 (Thu, 23 Jun 2011)
| 7 lines Addresses AST-2011-010, remote crash in IAX2 driver
Thanks to twilson for identifying the issue and providing the
patches. AST-2011-010 ........ ................
2011-06-23 03:10 +0000 [r324557] Terry Wilson <twilson@digium.com>
* tests/test_netsock2.c: Remove tests for parsing address with
invalid port getaddrinfo on OS X returns with EAI_NONAME error
when passed a port greater than 65535. Linux throws no error, so
remove the tests for now.
2011-06-22 19:16 +0000 [r324491] Richard Mudgett <rmudgett@digium.com>
* channels/chan_sip.c: Use correct variable for text SRTP media.
2011-06-22 18:52 +0000 [r324484] Terry Wilson <twilson@digium.com>
* include/asterisk/netsock2.h, tests/test_netsock2.c (added),
main/netsock2.c, channels/chan_sip.c: Stop sending IPv6
link-local scope-ids in SIP messages The idea behind the patch
listed below was used, but in a more targeted manner. There are
now address stringification functions for addresses that are
meant to be sent to a remote party. Link-local scope-ids only
make sense on the machine from which they originate and so are
stripped in the new functions. There is also a host sanitization
function added to chan_sip which is used for when peer and dialog
tohost fields or sip_registry hostnames are used to craft a SIP
message. Also added are some basic unit tests for netsock2
address parsing. (closes issue ASTERISK-17711) Reported by:
ch_djalel Patches: asterisk-1.8.3.2-ipv6_ll_scope.patch uploaded
by ch_djalel (license 1251) Review:
https://reviewboard.asterisk.org/r/1278/
2011-06-22 18:41 +0000 [r324479-324481] Richard Mudgett <rmudgett@digium.com>
* channels/chan_sip.c: Timout or error on INFO or MESSAGE
transaction causes call to be lost. When exchanging INFO messages
within a call, 4xx error causes the call to be disconnected
although RFC 2976 explicitly states that such transactions do not
modify the state of the dialog. When exchanging MESSAGE messages
within a call, 4xx error causes the call to be disconnected. To
provide least surprise, we should not disconnect the call since a
MESSAGE is like INFO in this case. (Implied by RFC 3428 Section
2) (closes issue ASTERISK-17901) Reported by: neutrino88 Review:
https://reviewboard.asterisk.org/r/1257/ Review:
https://reviewboard.asterisk.org/r/1258/ JIRA SWP-3486
* channels/chan_sip.c: Comments and whitespace in chan_sip.c
2011-06-21 20:11 +0000 [r324364] David Vossel <dvossel@digium.com>
* include/asterisk/pbx.h, main/pbx.c: Fixes locking inversion issue
in ast_async_goto() During this function we can not hold the
"chan" lock while doing the masquerade, the explicit goto on the
tmp chan, or the channel alloc. Instead we need to get the
channel lock, store off information about the channel that we
need, and then let the channel lock go for the remainder of the
function. Review: https://reviewboard.asterisk.org/r/1275/
2011-06-21 16:09 +0000 [r324305] Kinsey Moore <kmoore@digium.com>
* apps/app_confbridge.c: ConfBridge does not handle hangup properly
When playing back a prompt to a channel, confbridge neglects to
check for hangup events causing lockup condititions for hangups
that occur before actually joining the conference. This change
ensures that the user is removed from the conference in the event
of a premature hangup. Review:
https://reviewboard.asterisk.org/r/1277/
2011-06-20 18:12 +0000 [r324239-324241] Leif Madsen <lmadsen@digium.com>
* configs/queuerules.conf.sample: Remove extra 'the'. Reported by
Vlad Povorozniuc
* configs/queuerules.conf.sample,
contrib/scripts/asterisk.logrotate: Revert previous merge which
had extra changes.
* configs/queuerules.conf.sample,
contrib/scripts/asterisk.logrotate: Remove extra 'the'. Reported
by Vlad Povorozniuc
2011-06-20 17:33 +0000 [r324237] Terry Wilson <twilson@digium.com>
* channels/chan_sip.c: Ignore media offers with a port of 0 Section
5.1 of RFC3264 states: A port number of zero in the offer
indicates that the stream is offered but MUST NOT be used.
(closes issue ASTERISK-17845) Reported by: jacco Patches:
issue19281_2.patch uploaded by jacco (license 1277) Tested by:
jacco, twilson
2011-06-17 18:51 +0000 [r324176-324178] Leif Madsen <lmadsen@digium.com>
* main/manager.c: Add Username and Secret fields to manager Login
action. Pointed out by Vlad Povorozniuc
* apps/app_meetme.c: Fix typo in documentation. Pointed out by Vlad
Povorozniuc
2011-06-17 18:23 +0000 [r324174] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c: Add header string to libpri debug output.
Add header string to libpri debug output so the libpri output can
be found/extracted easier from huge debug trace files.
2011-06-17 15:14 +0000 [r324115] Leif Madsen <lmadsen@digium.com>
* main/pbx.c: Fix grammar in documentation for Goto() and GotoIf()
(closes issue ASTERISK-18023) Reported by: Tim Osman
2011-06-16 22:41 +0000 [r324048-324049] Terry Wilson <twilson@digium.com>
* channels/chan_local.c: Shame on me
* include/asterisk/channel.h, main/channel.c,
channels/chan_local.c, channels/chan_sip.c: Lock the channel
before calling the setoption callback The channel needs to be
locked before calling these callback functions. Also,
sip_setoption needs to lock the pvt and a check p->rtp is
non-null before using it. Review:
https://reviewboard.asterisk.org/r/1220/
2011-06-16 18:12 +0000 [r323990] Richard Mudgett <rmudgett@digium.com>
* tests/test_event.c: The test_event unit test is occasionally
failing. Wait for the special posted event to process before
adding a new subscription.
2011-06-16 15:58 +0000 [r323754-323932] Terry Wilson <twilson@digium.com>
* Makefile: Don't assume ASTDBDIR exists It most likely doesn't on
FreeBSD
* tests/test_db.c: Remove now-useless cast of ARRAY_LEN
* include/asterisk/utils.h: Make ARRAY_LEN() return the same type
on x86 and x86_64 systems
* tests/test_db.c: Fix more ARRAY_LEN format string issues
* /, main/features.c: Merged revisions 323733 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r323733 | twilson | 2011-06-15 13:13:00 -0500
(Wed, 15 Jun 2011) | 16 lines Merged revisions 323732 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r323732 | twilson | 2011-06-15 13:06:24 -0500 (Wed, 15 Jun 2011)
| 9 lines Fix DYNAMIC_FEATURES DYNAMIC_FEATURES were broken by a
recent DTMF change. This patch makes sure that dynamic features
are also checked when deciding whether or not to pass DTMF
through or store it for interpreting. (closes issue
ASTERISK-17914) Reported by: vrban ........ ................
2011-06-15 17:42 +0000 [r323730] Jonathan Rose <jrose@digium.com>
* res/res_config_pgsql.c: Adds locking to find_table in
res_configure_pgsql to prevent a crash. Bryonclark described the
problem as occuring during this function because of multiple
simultaneous database operations causing corruption against a
pgsqlConn object. (closes issue ASTERISK-17811) Reported by:
byronclark Patches: pgsql_find_table_locking.patch uploaded by
byronclark (license 1200)
2011-06-15 17:09 +0000 [r323672] Terry Wilson <twilson@digium.com>
* tests/test_db.c: Cast ARRAY_LEN to size_t for ast_logging 32-bit
and 64-bit machines return different types for ARRAY_LEN(), so
cast it before using in a format string.
2011-06-15 16:43 +0000 [r323669-323670] Richard Mudgett <rmudgett@digium.com>
* tests/test_event.c: Add a test to the event unit tests to catch
ASTERISK-18002. The new tests check to see if there are ANY
subscribers to the event type when ast_event_check_subscriber()
is not passed any specific ie values. (issue ASTERISK-18002)
* main/event.c: [regression] Voicemail MWI is no longer sent. When
leaving a voicemail, the MWI message is never sent. The same
thing happens when checking a voicemail and marking it as read.
If you restart Asterisk, everything comes up at that state
correctly, but changes to the messages in voicemail causes the
light to not be set appropriately. Very easy to reproduce. * Made
ast_event_check_subscriber() return TRUE if there are ANY
subscribers to an event type when there are no restricting ie
values passed. This allows an event being queued to be queued.
(closes issue ASTERISK-18002) Reported by: lmadsen Tested by:
lmadsen, irroot Patches: jira_asterisk_18002_v1.8.patch uploaded
by rmudgett (License #5621) (closes issue ASTERISK-18019)
2011-06-15 16:09 +0000 [r323610] Jonathan Rose <jrose@digium.com>
* res/res_config_pgsql.c: Adds PQclear calls on result to various
parts of res_conf_pgsql (closes issue ASTERISK-17812) Reported
by: byronclark Patches: pgsql_pqclear.patch uploaded by
byronclark (license 1200)
2011-06-15 15:31 +0000 [r323608] Sean Bright <sean@malleable.com>
* main/manager.c, /: Merged revisions 323579 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r323579 | seanbright | 2011-06-15 11:22:50 -0400
(Wed, 15 Jun 2011) | 32 lines Merged revisions 323559 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r323559 | seanbright | 2011-06-15 11:15:30 -0400 (Wed, 15 Jun
2011) | 25 lines Resolve a segfault/bus error when we try to map
memory that falls on a page boundary. The fix for ASTERISK-15359
was incorrect in that it added 1 to the length of the mmap'd
region. The problem with this is that reading/writing to that
extra byte outside of the bounds of the underlying fd causes a
bus error. The real issue is that we are working with both a FILE
* and the raw fd underneath it and not synchronizing between
them. The code that was removed in ASTERISK-15359 was correct,
but we weren't flushing the FILE * before mapping the fd. Looking
at the manager code in 1.4 reveals that the FILE * in 'struct
mansession' is never used except to create a temporary file that
we immediately fdopen. This means we just need to write a 0 byte
to the fd and everything will just work. The other branches
require a call to fflush() which, while not a guaranteed fix,
should reduce the likelihood of a crash. This all makes sense in
my head. (closes issue ASTERISK-16460) Reported by:
Ravelomanantsoa Hoby (hoby) Patches:
issue17747_1.4_svn_markII.patch uploaded by Sean Bright (license
#5060) ........ ................
2011-06-15 00:50 +0000 [r323392-323456] Richard Mudgett <rmudgett@digium.com>
* main/event.c: Add missing break in ast_event_get_cached().
* main/netsock2.c: Made ast_sockaddr_split_hostport() port warning
msgs more meaningful.
* main/dnsmgr.c: Add more strict hostname checking to
ast_dnsmgr_lookup(). Change suggested in review. Review:
https://reviewboard.asterisk.org/r/1240/
2011-06-14 16:38 +0000 [r323371] Jonathan Rose <jrose@digium.com>
* channels/chan_sip.c: Changes contact use in build_peer to use the
FORCE_RPORT flag instead of RPORT_PRESENT It turned out that this
was causing NAT=Yes to always use rport when present which was
against 1.6.2 behavior and the check itself was redundant since
the only way this segment of code could be reached was if
RPORT_PRESENT was already evaluated as true earlier. (closes
issue ASTERISK-17789) Reported by: byronclark Patches:
use_sip_nat_force_rport.patch uploaded by byronclark (license
1200)
2011-06-14 16:33 +0000 [r323370] Terry Wilson <twilson@digium.com>
* include/asterisk/rtp_engine.h, res/res_rtp_asterisk.c,
main/rtp_engine.c, channels/chan_sip.c: Add rtpkeepalives back to
1.8 The RTP-engine conversion left out support for handling
rtpkeepalives. This patch adds them back. (closes issue
ASTERISK-17304) Reported by: lmadsen Review:
https://reviewboard.asterisk.org/r/1226/
2011-06-13 20:22 +0000 [r323154-323234] Leif Madsen <lmadsen@digium.com>
* configs/sip.conf.sample: Additional documentation for bindaddr.
Note that bindaddr will only enable UDP instead of both UDP and
TCP which is what I would expect for backwards compatibility with
systems being upgraded which only support UDP transportation.
(closes issue ASTERISK-17976) Reported by: Sean Darcy
* main/channel.c: Avoid dividing by zero with L() option to Dial()
Reported by: nicolasom Patches: issue-17995.patch - nicolasom
(License #5994)
* res/res_agi.c: Tweak documentation for AGI Hangup command.
(closes issue ASTERISK-17999) Reported by: Ben Klang Patches:
hangup-doc.diff - uploaded by Ben Klang (License #5876)
2011-06-10 19:20 +0000 [r323040] Matthew Nicholson <mnicholson@digium.com>
* channels/chan_sip.c: Unlock the sip channel during fax detection
like chan_dahdi does to prevent a deadlock with
ast_autoservice_stop. (closes issue ASTERISK-17798) tested by
mnicholson
2011-06-10 15:29 +0000 [r322865-322981] Terry Wilson <twilson@digium.com>
* main/db.c: Avoid a DB1 infinite loop bug Explicity check the last
entry in the DB and make sure that we don't iterate past it.
Since there can be no duplicates, this just makes sure that we
stop after matching the last key. This patch also refactors the
code to get away from some code duplication. A previous patch
added many astdb tests and this patch passed them. Review:
https://reviewboard.asterisk.org/r/1259/
* tests/test_db.c (added): Add some astdb unit tests
* include/asterisk/astdb.h: Correct ast_db_deltree documentation
ast_db_deltree returns -1 on error, otherwise the number of
deletions
2011-06-09 17:37 +0000 [r322807] Matthew Nicholson <mnicholson@digium.com>
* channels/chan_sip.c: don't drop any voice frames when checking
for T.38 during early media (closes issue ASTERISK-17705) Review:
https://reviewboard.asterisk.org/r/1186/ patch by oej reported by
oej
2011-06-09 16:31 +0000 [r322749] Richard Mudgett <rmudgett@digium.com>
* include/asterisk/features.h, apps/app_directed_pickup.c,
main/features.c: Remove potential deadlock in call pickup race.
Deadlock is possible in ast_do_pickup() when holding the target
channel lock and trying to get the chan channel lock. Also,
holding the target lock when calling ast_channel_masquerade() is
not a good idea because that routine does deadlock avoidance. *
Removed the need to hold the target lock after marking the target
with a datastore and getting the connected line data off of the
target channel. * Moved can_pickup() to ast_can_pickup() in
features.c. Now all the call pickup methods use the same basic
call pickup availability check. Review:
https://reviewboard.asterisk.org/r/1234/
2011-06-09 14:06 +0000 [r322585] Jonathan Rose <jrose@digium.com>
* main/utils.c, include/asterisk/utils.h, channels/chan_sip.c,
tests/test_utils.c: Adds ast_escape_encoded utility to properly
handle escaping of quoted field before uri. This commit backports
a feature in trunk affecting initreqprep so that display name
won't be encoded improperly. Also includes unit tests for the
ast_escape_quoted function. This patch gives 1.8 a much improved
outlook in countries which don't use standard ASCII characters.
(closes issue ASTERISK-16949) Reported by: Örn Arnarson Review:
https://reviewboard.asterisk.org/r/1235/
2011-06-08 20:46 +0000 [r322425-322484] Richard Mudgett <rmudgett@digium.com>
* apps/app_queue.c: Ring all queue with more than 255 agents will
cause crash. 1. Create a ring-all queue with 500 permanent
agents. 2. Call it. 3. Asterisk will crash. The watchers array in
app_queue.c has a hard limit of 255. Bounds checking is not done
on this array. No sane person should put 255 people in a ring-all
queue, but we should not crash anyway. * Added bounds checking to
the watchers array. JIRA AST-464 JIRA SWP-2903
* main/dnsmgr.c: SRV lookup attempted for SIP peers listed as an IP
address. Asterisk attempts to SRV lookup a host name even if the
host name is an IP address. Regression introduced when IPv6
support was added. * Restored the check in ast_dnsmgr_lookup() to
see if the given host name is an IP address. The IP address could
be in either IPv4 or IPv6 formats. (closes issue ASTERISK-17815)
Reported by: Byron Clark Tested by: Byron Clark, Richard Mudgett
Patches: issue19248_v1.8.patch - uploaded by Richard Mudgett
(License #5621) Review: https://reviewboard.asterisk.org/r/1240/
2011-06-08 06:18 +0000 [r322322] Gregory Nietsky <gregory@distrotech.co.za>
* channels/chan_sip.c: Make handle_request_publish do dialog
expiration and destruction. This patch fixes
handle_request_publish so that it does dialog expiration and
destruction. Without this patch the incoming PUBLISH requests
will get stuck in the dialog list. Restarting asterisk is the
only way to remove them. Personal observation on one system the
server hung up while looping through the channels rendering
asterisk unusable and all sip phones unregisterd when they try
reregister more requests are added. (closes issue #18898)
Reported by: gareth Tested by: loloski, Chainsaw, wimpy, se, kuj,
irroot Jira:
https://issues.asterisk.org/jira/browse/ASTERISK-17915 Review:
https://reviewboard.asterisk.org/r/1253
2011-06-07 17:59 +0000 [r322189] Paul Belanger <pabelanger@digium.com>
* configs/sip_notify.conf.sample: Use correct syntax for 'sip
notify snom-reboot' (closes issue ASTERISK-17915)
2011-06-06 19:07 +0000 [r322069] Jonathan Rose <jrose@digium.com>
* main/asterisk.c, include/asterisk/logger.h: Fixes level toggling
for logger set levels since it was reversed (closes issue
ASTERISK-17850) Reported by: Luke H Tested by: jrose, Luke H
Review: https://reviewboard.asterisk.org/r/1244/
2011-06-03 22:09 +0000 [r321812-321926] Richard Mudgett <rmudgett@digium.com>
* cdr/cdr_radius.c, cel/cel_radius.c: Asterisk crash when unloading
cdr_radius/cel_radius. The rc_openlog() API call is passed a
string that is used by openlog() to format log messages. The
openlog() does not copy the string it just keeps a pointer to it.
When the module is unloaded, the string is gone from memory.
Depending upon module load order and if the other module then has
an error, a crash happens. * Pass rc_openlog() a strdup'd string
with the understanding that there will be a small memory leak if
the cdr_radius/cel_radius modules are unloaded. * Call
rc_destroy() to free the rc handle memory when the module is
unloaded. JIRA AST-483 JIRA SWP-3062
* main/ccss.c: Be more explicit for CCSS generic device state event
subscription. Make CCSS generic device state event subscription
specify the AST_EVENT_IE_STATE ie exists to be safe.
* main/event.c, tests/test_event.c: Event subscription fixes. Must
commit the subscription fixes together with the integration
subscription tests. The subscription fixes cause an erroneously
passing test to fail. The new subscription tests detect errors
without the subscription fixes. * Added missing event_names[]
table entry. * Reworked
ast_event_check_subscriber()/match_sub_ie_val_to_event() to
correctly detect if a subscriber exists for the proposed event. *
Made match_ie_val() and match_sub_ie_val_to_event() check the
buffer length for RAW payload types. * Fixed error handling
memory leak in ast_event_sub_activate(), ast_event_unsubscribe(),
and ast_event_queue(). * Made ast_event_new() and
ast_event_check_subscriber() better protect themselves from an
invalid payload type. * Added container lock protection between
removing old cache events and adding the new cached event in
ast_event_queue_and_cache()/event_update_cache(). * Added new
event subscription tests.
* main/event.c, include/asterisk/event.h: Constify subscription
description parameter string.
* channels/chan_iax2.c, channels/chan_sip.c: Correct IAX2 and SIP
event subscription description string.
2011-06-03 18:32 +0000 [r321753] Russell Bryant <russell@digium.com>
* tests/test_astobj2.c: Backport an astobj2 unit test so that it
runs on 1.8 as well.
2011-06-03 13:17 +0000 [r321685] Leif Madsen <lmadsen@digium.com>
* configs/queues.conf.sample: Also document the 'queue-minute'
option. (closes issue #19386) Reported by: juanmol
2011-06-01 23:11 +0000 [r321547] Richard Mudgett <rmudgett@digium.com>
* main/cdr.c: CDR comment tweaks.
2011-06-01 20:10 +0000 [r321537] Brett Bryant <bbryant@digium.com>
* apps/app_voicemail.c: This patch fixes an issue with using the
wrong voicemail folders with greetings. (closes issue #17871)
Reported by: edhorton Patches: digium_bug_17871_2 uploaded by
fhackenberger (license 592) Tested by: edhorton, fhackenberger
2011-06-01 10:40 +0000 [r321528] Alexandr Anikin <may@telecom-service.ru>
* addons/ooh323c/src/oochannels.c, addons/chan_ooh323.c,
addons/ooh323c/src/ooh245.c: Fix double alerting, add forced
alerting before answer Fix double alerting (it wasn't fixed here
by issue #18542) Add forced alerting before connect (if it wasn't
before) Try to send all packets from outgoing queue rather than
one only Call goes into clearing state when disconnect command is
received (closes issue #19361) Reported by: vmikhelson Patches:
issue19361-3.patch uploaded by may213 (license 454) Tested by:
vmikhelson
2011-05-31 20:54 +0000 [r321517] Richard Mudgett <rmudgett@digium.com>
* include/asterisk/dnsmgr.h, include/asterisk/acl.h: Update some
comments.
2011-05-31 18:52 +0000 [r321515] David Vossel <dvossel@digium.com>
* channels/chan_local.c: Chan_local locking cleanup. This patch
removes all of the unnecessary deadlock avoidance loops that
occur in chan_local. It also resolves an issue with a deadlock
triggered by local channel optimizations. (issue #18028) Review:
https://reviewboard.asterisk.org/r/1231/
2011-05-31 16:04 +0000 [r321511] Leif Madsen <lmadsen@digium.com>
* channels/chan_sip.c: Enhance NOTICE message to know who couldn't
access the dialplan. (closes issue #19390) Reported by: lmadsen
Patches: __20110531-sip-notice-tweak.txt uploaded by lmadsen
(license 10) Tested by: russell
2011-05-28 00:27 +0000 [r321337-321436] Richard Mudgett <rmudgett@digium.com>
* res/res_agi.c: Some hagi launch cleanup. Inspired by issue 19256.
This patch would also fix the crash.
* main/srv.c: Crash when using hagi and no servers are available.
When none of the servers returned by the SRV querey respond,
asterisk crashes. The problem is that if the loop over all the
SRV entries finishes then the srv_context has already been
cleaned up. * Make ast_srv_cleanup() check to see if the context
is already cleaned up. (closes issue #19256) Reported by:
byronclark
* apps/app_privacy.c: The app_privacy args have undocumented
"options" position, interferes with "context" position. * Add
documention for unused "options" position to match existing code.
(closes issue #19273) Reported by: mdavenport
2011-05-27 21:54 +0000 [r321333-321335] Leif Madsen <lmadsen@digium.com>
* include/asterisk/frame.h, main/file.c: Fix issue with playback of
H.261 video. (closes issue #19379) Reported by: neutrino88
Patches: videoprompt.patch uploaded by neutrino88 (license 297)
(changes by russell)
* main/features.c: Allow parking lot hints and musicclass to be
set. (closes issue #19378) Reported by: sboily_proformatique
Patches: pf_parkinghint_music_fix uploaded by sboily
proformatique (license 206) Tested by: russell
2011-05-27 21:31 +0000 [r321330] Richard Mudgett <rmudgett@digium.com>
* apps/app_privacy.c: The app_privacy args have undocumented
"options" position, interferes with "context" position. * Add
documention for unused "options" position to match existing code.
The trunk(v1.10) version will remove the unused options position.
(closes issue #19273) Reported by: mdavenport
2011-05-27 14:59 +0000 [r321273] Jonathan Rose <jrose@digium.com>
* channels/sip/reqresp_parser.c: markm committed a patch I was
working on yesterday, this fixes it to mesh up with suggestions
by mnicholson.
2011-05-27 08:31 +0000 [r321211] Alec L Davis <sivad.a@paradise.net.nz>
* main/features.c: Fix *8 directed pickup locks system during
pickupsound play out move playout from sip_pickup_thread to
bridge using BRIDGE_PLAY_SOUND method, This stop the clash of 2
threads trying to write audio to same channel. In addition fixes
choppy audio beep in issue 19177. (issue #18654) (issue #19177)
Reported by: Docent Patches: review1232-1.88888888 alecdavis
(license 585) Tested by: alecdavis Review:
https://reviewboard.asterisk.org/r/1232/
2011-05-26 21:48 +0000 [r321100-321155] Mark Murawki <markm@intellasoft.net>
* channels/chan_sip.c, channels/sip/reqresp_parser.c: Fixed build
problem with dev mode enabled, which was caused by commit 321100.
Reformulated patch to be more generic. Moved the sip uri parse
variable initalization to parse_uri_full in reqresp_parser.c.
This will ensure that any use of parse uri will have null output
variables if the parse fails. (closes issue #19346) Reported by:
kobaz Tested by: kobaz,JonathanRose Review: [full review board
URL with trailing slash]
* main/netsock2.c, channels/chan_sip.c: ast_sockaddr_resolve() in
netsock2.c may deref a null pointer Added a null check in
netsock2 ast_sockaddr_resolve() as well as added default
initalizers in chan_sip parse_uri_legacy_check() to make sure
that invalid uris will make null (and not undefined)
user,pass,domain,transport variables (closes issue #19346)
Reported by: kobaz Patches: netsock2.patch uploaded by kobaz
(license 834) Tested by: kobaz, Marquis
2011-05-26 18:10 +0000 [r321044] Richard Mudgett <rmudgett@digium.com>
* include/asterisk/netsock2.h: Update ast_sockaddr comment with an
important note.
2011-05-26 17:29 +0000 [r321042] Terry Wilson <twilson@digium.com>
* main/rtp_engine.c: Initialize stack-allocated ast_sockaddrs
before use It is important to always initialize ast_sockaddrs
before use--even if they are passed to ast_sockaddr_copy as the
underlying storage could be bigger than what ends up being
copied--leaving part of the data unitialized.
2011-05-26 15:57 +0000 [r320947] Russell Bryant <russell@digium.com>
* channels/chan_alsa.c, channels/chan_mgcp.c: Remove some variables
that were set but unused.
2011-05-25 22:25 +0000 [r320796-320883] Richard Mudgett <rmudgett@digium.com>
* channels/chan_sip.c: Native SIP CCSS sends bad CC cancel
SUBSCRIBE message. The SUBSCRIBE message used to cancel a CC
request has incorrect To/From SIP headers. They are reversed and
the dialog tags are the same when they should not be. If pedantic
mode was disabled, then the cancel would have succeeded despite
the incorrect message. * The SIP_OUTGOING flag was not set
correctly for the dialog and I had to move some CC subscribe
handling code as a result. * Initialized the dialog subscribed
type to CALL_COMPLETION earlier. If a CC request SUBSCRIBE
message comes in and the CC instance is not found, the 404
response was duplicated. JIRA AST-568 JIRA SWP-3493
* UPGRADE.txt, CHANGES, apps/app_queue.c, apps/app_dial.c,
main/channel.c, main/manager.c, apps/app_meetme.c,
apps/app_fax.c, main/features.c: The AMI Newstate event contains
different information between v1.4 and v1.8. The addition of
connected line support in v1.8 changes the behavior of the
channel caller ID somewhat. The channel caller ID value no longer
time shares with the connected line ID on outgoing call legs. The
timing of some AMI events/responses output the connected line ID
as caller ID. These party ID's are now separate. * The
ConnectedLineNum and ConnectedLineName headers were added to many
AMI events/responses if the CallerIDNum/CallerIDName headers were
also present. (closes issue #18252) Reported by: gje Tested by:
rmudgett Review: https://reviewboard.asterisk.org/r/1227/
* include/asterisk/channel.h, main/channel.c, main/features.c: Give
zombies a safe channel driver to use. Recent crashes from zombie
channels suggests that they need a safe home to goto. When a
masquerade happens, the physical part of the zombie channel is
hungup. The hangup normally sets the channel private pointer to
NULL. If someone then blindly does a callback to the channel
driver, a crash is likely because the private pointer is NULL.
The masquerade now sets the channel technology of zombie channels
to the kill channel driver. Related to the following issues:
(issue #19116) (issue #19310) Review:
https://reviewboard.asterisk.org/r/1224/
2011-05-25 00:49 +0000 [r320716] Terry Wilson <twilson@digium.com>
* addons/chan_mobile.c: Cast data as char * before using S_OR This
is required for compiling successfully under dev mode
2011-05-23 17:53 +0000 [r320650] Richard Mudgett <rmudgett@digium.com>
* CHANGES, main/manager.c: Add ConnectedLineNum/Name headers to
output of AMI action Status. * Add ConnectedLineNum and
ConnectedLineName headers to the output of the AMI action Status.
This makes it easier to find out who the channel is connected to
without having to lookup BridgedChannel or when they are
connected to an application (e.g.: VoiceMail) which has no
bridged channel. * Bridged channels with no CallerID had ""
instead of "<unknown>" output, that might be a bug as "<unknown>"
was what older versions used. (closes issue #18158) Reported by:
gareth Patches: svn-292308.diff uploaded by gareth (license 208)
2011-05-23 16:19 +0000 [r320573] Tilghman Lesher <tilghman@meg.abyt.es>
* configure, configure.ac: GNU libiconv uses symbol "libiconv_open"
instead of "iconv_open". (closes issue #19344) Reported by:
rohanl Patches: iconv-check.patch uploaded by rohanl (license
1284)
2011-05-23 16:18 +0000 [r320568] David Vossel <dvossel@digium.com>
* main/tcptls.c, /: Merged revisions 320562 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r320562 | dvossel | 2011-05-23 11:15:18 -0500 (Mon, 23 May 2011)
| 9 lines Adds missing part to the ast_tcptls_server_start fails
second attempt to bind patch. (closes issue #19289) Reported by:
wdoekes Patches:
issue19289_delay_old_address_setting_tcptls_2.patch uploaded by
wdoekes (license 717) ........
2011-05-23 15:47 +0000 [r320560] Kevin P. Fleming <kpfleming@digium.com>
* configure, configure.ac: Don't generate spurious "No: command not
found" messages when running the configure script on a system
that has neither gmime-config nor pkg-config.
2011-05-23 14:33 +0000 [r320504] Jonathan Rose <jrose@digium.com>
* channels/chan_sip.c: Fixes segfault occuring in chan_sip.c at
__set_address_from_contact Checks to see if domain contains
anything before sending it off to ast_sockaddr_resolve which is
where the segfault was occuring due to null str. (closes issue
#18857) Reported by: sybasesql Review:
https://reviewboard.asterisk.org/r/1225/
2011-05-22 23:34 +0000 [r320445] Tilghman Lesher <tilghman@meg.abyt.es>
* res/res_odbc.c, /: Merged revisions 320444 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r320444 | tilghman | 2011-05-22 18:25:51 -0500 (Sun, 22 May 2011)
| 8 lines Don't crash when the connection fails. (closes issue
#19250) Reported by: seadweller Patches:
20110514__issue19250.diff.txt uploaded by tilghman (license 14)
Tested by: seadweller, sum ........
2011-05-20 21:39 +0000 [r320338] David Vossel <dvossel@digium.com>
* main/tcptls.c, /: Merged revisions 320271 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r320271 | dvossel | 2011-05-20 16:24:48 -0500 (Fri, 20 May 2011)
| 8 lines Fixes issue with ast_tcptls_server_start failing on
second attempt to bind. (closes issue #19289) Reported by:
wdoekes Patches:
issue19289_delay_old_address_setting_tcptls.patch uploaded by
wdoekes (license 717) ........
2011-05-20 20:49 +0000 [r320237] Richard Mudgett <rmudgett@digium.com>
* /, apps/app_meetme.c: Merged revisions 320236 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r320236 | rmudgett | 2011-05-20 15:44:54 -0500
(Fri, 20 May 2011) | 20 lines Merged revisions 320235 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r320235 | rmudgett | 2011-05-20 15:38:22 -0500 (Fri, 20 May 2011)
| 13 lines The meetme CLI command completion leaves conferences
mutex locked. When issuing a meetme kick CLI command and an
invalid (non-existent) conference number is specified, pressing
Tab leaves the conferences mutex locked and, therefore, all
conferences deadlock. Add missing unlock. (closes issue #19336)
Reported by: zvision Patches: app_meetme.diff uploaded by zvision
(license 798) ........ ................
2011-05-20 18:48 +0000 [r320180] Matthew Nicholson <mnicholson@digium.com>
* channels/chan_sip.c: This commit modifies the way polling is done
on TLS sockets. Because of the buffering the TLS layer does,
polling is unreliable. If poll is called while there is data
waiting to be read in the TLS layer but not at the network layer,
the messaging processing engine will not proceed until something
else writes data to the socket, which may not occur. This change
modifies the logic around TLS sockets to only poll after a failed
read on a non-blocking socket. This way we know that there is no
data waiting to be read from the buffering layer. (closes issue
#19182) Reported by: st Patches: ssl-poll-fix3.diff uploaded by
mnicholson (license 96) Tested by: mnicholson
2011-05-20 18:12 +0000 [r320162] Jonathan Rose <jrose@digium.com>
* apps/app_voicemail.c: Fixes an imapfolder related crash
imapfolders being set in the general section of voicemail would
cause the inbox folder name to change. Since sound file names are
made based on the names of the folders, this would cause the
audio related to that folder name to change and if Asterisk
attempted to play it, the channel would instantly hang up when
the audio file couldn't be found. This patch searches for the
name of the folder first to leave existing behavior in tact and
if that fails, it uses the normal inbox name to get the sound
file instead. (closes issue #16104) Reported by: blkline Review:
https://reviewboard.asterisk.org/r/1215/
2011-05-20 17:03 +0000 [r319997-320059] Richard Mudgett <rmudgett@digium.com>
* main/features.c: Misc comment cleanup in features.c.
* main/channel.c, main/features.c: Crash while transferring a call
during DTMF feature timeout. When a call is being attended
transferred during the time between AST_FRAME_DTMF_BEGIN and
AST_FRAME_DTMF_END, the transferred channel becomes a zombie (so
tech data is not available), making ast_dtmf_stream() segfault
when it tries to send the DTMF digit (at least with SIP
channels). Patch based on feature-end-zombie.patch uploaded by
Irontec (license 1256) * Check for zombies when
ast_channel_bridge() returns. * Guarantee that the fo parameter
value is initialized in ast_channel_bridge() before any returns.
(closes issue #19116) Reported by: Irontec Tested by: rmudgett
* apps/app_directed_pickup.c, main/features.c: Change some variable
names to make pickup code easier to understand.
* apps/app_directed_pickup.c, main/features.c: Crash when using
directed pickup applications. The directed pickup applications
can cause a crash if the pickup was successful because the
dialplan keeps executing. This patch does the following: *
Completes the channel masquerade on a successful pickup before
the application returns. The channel is now guaranteed a zombie
and must not continue executing the dialplan. * Changes the
return value of the directed pickup applications to return zero
if the pickup failed and nonzero(-1) if the pickup succeeded. *
Made some code optimizations that no longer require re-checking
the pickup channel to see if it is still available to pickup.
(closes issue #19310) Reported by: remiq Patches:
issue19310_v1.8_v2.patch uploaded by rmudgett (license 664)
Tested by: alecdavis, remiq, rmudgett Review:
https://reviewboard.asterisk.org/r/1221/
2011-05-20 13:28 +0000 [r319938] Jonathan Rose <jrose@digium.com>
* configs/sip.conf.sample, channels/sip/include/sip.h,
channels/chan_sip.c: Adds legacy_useroption_parsing to address
interoperability concerns. With the new option engaged, Asterisk
should interpret user fields with useroptions contained within
the userfield of the uri by stripping them out of the original
message whenever a semicolon is encountered in the userfield
string. (closes issue #18344) Reported by: danimal Tested by:
jrose Review: https://reviewboard.asterisk.org/r/1223/
2011-05-19 23:28 +0000 [r319920] Terry Wilson <twilson@digium.com>
* main/bridging.c, include/asterisk/bridging_technology.h,
include/asterisk/bridging.h: Revert part of a change to the
bridging API code The capabilities used in the bridging API are
very different than the ones used for formats. When the
conversion was made expanding the bit width of codecs, the
bridging code was accidentally accosted in ways that it didn't
deserve.
2011-05-19 18:32 +0000 [r319866] Jonathan Rose <jrose@digium.com>
* main/features.c: Fix Randomize option on Park() The randomize
option was generally not working like it should have at all on
Park(). This patch restores intended functionality. (closes issue
#18862) Reported by: davidw Tested by: jrose Review:
https://reviewboard.asterisk.org/r/1222/
2011-05-19 17:59 +0000 [r319812] Mark Murawki <markm@intellasoft.net>
* cel/cel_odbc.c: In cel_odbc, an uninitialized RWLIST is attempted
to be locked. Added INIT and DESTROY for the RWLIST odbc_tables
(closes issue #19331) Reported by: kobaz Patches: odbc_cel.patch
uploaded by kobaz (license 834)
2011-05-19 16:50 +0000 [r319758] Richard Mudgett <rmudgett@digium.com>
* main/ccss.c: CCSS generic agent with POTS and ISDN phones fail
caller busy call-back test. If the following is true after a CCSS
activation: * The generic agent is for an analog phone or ISDN
phone. (Caller party) * The called party becomes available. * The
caller party is not available. When the caller party becomes
available, the caller is not alerted to the called party being
available. The generic agent still thinks the caller is busy. *
Fixed the generic agent device state event subscription to look
for all device states that are considered available. *
Encapsulated the device state test for CCSS generic device
available in cc_generic_is_device_available(). Made the generic
agent and monitor use the new function instead of the manually
coded inline equivalent. JIRA AST-559 JIRA SWP-3462
2011-05-18 23:15 +0000 [r319529-319654] Terry Wilson <twilson@digium.com>
* /, channels/chan_sip.c: Merged revisions 319653 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r319653 | twilson | 2011-05-18 16:11:57 -0700
(Wed, 18 May 2011) | 15 lines Merged revisions 319652 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r319652 | twilson | 2011-05-18 16:04:35 -0700 (Wed, 18 May 2011)
| 8 lines Make sure everyone gets an unhold when a transfer
succeeds Some phones, like the Snom phones, send a hold to the
transfer target after before sending the REFER. We need to make
sure that we unhold the parties that are being connected after
the masquerade. If Local channels with the /nm option are used
when dialing the parties, hold music would still be playing on
the transfer target, even after being connected with the
transferee. ........ ................
* channels/chan_sip.c: Unbreak the storing of registrations for
restart The fix for issue 18882 broke retrieving non-realtime
peers from the ast_db on restart/reload. This patch tries to
unbreak things while leaving the intent of the original fix
intact. (closes issue #19318) Reported by: remiq Patches:
diff.txt uploaded by twilson (license 396) Tested by: lmadsen,
remiq
* apps/app_dial.c, /: Merged revisions 319528 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r319528 | twilson | 2011-05-18 13:02:06 -0700
(Wed, 18 May 2011) | 17 lines Merged revisions 319527 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r319527 | twilson | 2011-05-18 12:56:08 -0700 (Wed, 18 May 2011)
| 10 lines Fix app_dial ring groups Revert part of r315643. We
need to remove the datastore here as well. The code in bridging
code will catch anything that app_dial might miss. (closes issue
#19311) Reported by: mspuhler Patches: issue_19311_no_answer.diff
uploaded by elguero (license 37) ........ ................
2011-05-17 21:57 +0000 [r319469] Richard Mudgett <rmudgett@digium.com>
* channels/misdn/isdn_lib.c: Merged revision 319468 from
https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
.......... r319468 | rmudgett | 2011-05-17 16:49:31 -0500 (Tue,
17 May 2011) | 15 lines The mISDN HDLC mode is prevented on
dialed channels. The use of mISDN HDLC mode is prevented if the
mISDN dial technology option 'h1' is used when config option
astdtmf=yes. There is a bug in channels/misdn/isdn_lib.c which
prevents the use of HDLC mode. Instead of setting the channel to
HDLC mode it is set to transparent(no dsp, no hdlc), although
hdlc is not "no hdlc". I.e the logging message is correct, but
the if condition is not. Make check the nodsp and hdlc flags.
JIRA ABE-2787 JIRA SWP-3437 ..........
2011-05-17 12:53 +0000 [r319365-319367] Leif Madsen <lmadsen@digium.com>
* apps/app_voicemail.c: Don't create [general] voicemail context
when using users.conf Prior to this patch, app_voicemail would
create a [general] context when parsing users.conf. (closes issue
#18891) Reported by: pdugas Patches:
app_voicemail-ignore-general.patch uploaded by pdugas (license
1222) app_voicemail-ignore-general-style-guidelines.patch
uploaded by seanbright (license 71) Tested by: pdugas
* contrib/init.d/rc.debian.asterisk: Make Debian init script lsb
compliant (closes issue #18896) Reported by: manwe Patches:
debian_init_lsb.patch uploaded by manwe (license 1223)
2011-05-16 21:00 +0000 [r319261] Jonathan Rose <jrose@digium.com>
* main/dsp.c: Makes busy detection in dsp.c always allow for at
least one frame (20ms) of error so that 200ms tone lengths don't
get ignored by single frame error lengths.
2011-05-16 20:33 +0000 [r319259] Richard Mudgett <rmudgett@digium.com>
* main/ccss.c: Deadlock between generic CCSS agent and native ISDN
CCSS. Deadlock can occur when the generic CCSS agent is deleting
duplicate CC offers and the native ISDN CC driver is processing
an incoming CC message. The cc_core_instances container lock
cannot be held when an agent or monitor callback is invoked
without the possibility of a deadlock. * Make
kill_duplicate_offers() remove the reference in cc_core_instances
outside of the container lock. JIRA AST-566 JIRA SWP-3469
2011-05-16 18:17 +0000 [r319204] Terry Wilson <twilson@digium.com>
* /, channels/chan_sip.c: Merged revisions 319202 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r319202 | twilson | 2011-05-16 11:00:21 -0700 (Mon, 16 May 2011)
| 4 lines Unlink a peer from peers_by_ip when expiring a
registration Review: https://reviewboard.asterisk.org/r/1218/
........
2011-05-16 15:57 +0000 [r319145] David Vossel <dvossel@digium.com>
* /, channels/chan_sip.c: Merged revisions 319144 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r319144 | dvossel | 2011-05-16 10:56:16 -0500 (Mon, 16 May 2011)
| 2 lines Fixes issue with peer ref-counting during
handle_request_subscribe. (closes issue #19293) Reported by:
irroot ........
2011-05-16 15:53 +0000 [r319142] Matthew Nicholson <mnicholson@digium.com>
* channels/chan_sip.c: Make sure tcptls_session exists before
dereferencing it. (closes issue #19192) Reported by: stknob
Patches: 10-tcptls-unreachable-peer-segfault.patch uploaded by
Chainsaw (license 723) Tested by: vois, Chainsaw
2011-05-16 14:35 +0000 [r319085] Paul Belanger <pabelanger@digium.com>
* res/res_http_post.c, configure, include/asterisk/autoconfig.h.in,
configure.ac: Support gmime-2.4 (closes issue #18863) Reported
by: tzafrir Patches: gmime-2.4-18.diff uploaded by tzafrir
(license 46) Tested by: tzafrir Review:
https://reviewboard.asterisk.org/r/1213/
2011-05-16 14:26 +0000 [r319083] David Vossel <dvossel@digium.com>
* formats/format_wav.c: Fixes Big Endian build issue. (closes issue
#19298) Reported by: tzafrir
2011-05-13 18:09 +0000 [r318917-318921] Brett Bryant <bbryant@digium.com>
* main/channel.c: Fixes a segmentation fault in dynamic hints when
a channel technology isn't loaded for a hint. (closes issue
#18495) Reported by: bertrand Tested by: bertrand
* res/res_srtp.c: This patch fixes an issue with SRTP which makes
HOLD/UNHOLD impossible when too much time has passed between
sending audio. (closes issue #18206) Reported by: bernhardsi
Patches: res_srtp_unhold.patch uploaded by bernhards (license
1138) Tested by: bernhards, notthematrix
* channels/chan_sip.c: This patch allows TCP peers into the ast_db
where they were previously restricted. (closes issue #18882)
Reported by: cmaj Patches:
patch-chan_sip-1.8.3-rc2-allow-tcp-peer-store-db-and-readonly-rt-backend.diff.txt
uploaded by cmaj (license 830) Tested by: cmaj
2011-05-13 16:28 +0000 [r318783-318868] Richard Mudgett <rmudgett@digium.com>
* main/features.c: CDR's are being written immediately on caller
hangup. CDR's are being written immediately on caller hangup. The
dialplan is not able to modify it in the h exten. The h exten in
the initial context is not run before closing CDR's when the
bridge is unlinked if a macro is active and does not have an h
exten. * Make ast_bridge_call() check for an h exten in the
current context and if a macro is active then the initial
context. The first h exten found is then run before closing the
CDR. (closes issue #18212) Reported by: leearcher Patches:
issue18212_v1.8.patch uploaded by rmudgett (license 664) Tested
by: rmudgett Review: https://reviewboard.asterisk.org/r/1206/
* channels/sig_pri.c: PRI early media won't ring. And another way
to pass early media. Don't indicate that there is inband
information present, just assume that the B channel is connected.
* Restore clearing the dialing flag Rx squelch unconditionally
when a PROCEEDING message comes in. (closes issue #19268)
Reported by: tbsky Patches: issue19268_v1.8.patch uploaded by
rmudgett (license 664) Tested by: tbsky
2011-05-12 23:35 +0000 [r318720] Matthew Nicholson <mnicholson@digium.com>
* channels/sip/reqresp_parser.c: Handle ipv6 addresses in the
sent-by Via: field. This change fixes a regression in via header
parsing and ipv6 handling. (closes issue #18951)
2011-05-12 22:52 +0000 [r318671] Alec L Davis <sivad.a@paradise.net.nz>
* include/asterisk/features.h, channels/chan_sip.c,
apps/app_directed_pickup.c, main/features.c: Fix directed group
pickup feature code *8 with pickupsounds enabled Since 1.6.2, the
new pickupsound and pickupfailsound in features.conf cause many
issues. 1). chan_sip:handle_request_invite() shouldn't be playing
out the fail/success audio, as it has 'netlock' locked. 2).
dialplan applications for directed_pickups shouldn't beep. 3).
feature code for directed pickup should beep on success/failure
if configured. Created a sip_pickup() thread to handle the pickup
and playout the audio, spawned from handle_request_invite. Moved
app_directed:pickup_do() to features:ast_do_pickup(). Functions
below, all now use the new ast_do_pickup() app_directed_pickup.c:
pickup_by_channel() pickup_by_exten() pickup_by_mark()
pickup_by_part() features.c: ast_pickup_call() (closes issue
#18654) Reported by: Docent Patches:
ast_do_pickup_1.8_trunk.diff.txt uploaded by alecdavis (license
585) Tested by: lmadsen, francesco_r, amilcar, isis242,
alecdavis, irroot, rymkus, loloski, rmudgett Review:
https://reviewboard.asterisk.org/r/1185/
2011-05-11 18:47 +0000 [r318549-318550] Terry Wilson <twilson@digium.com>
* channels/chan_sip.c: Comment out the REF_DEBUG that slipped in
during debugging
* /, channels/chan_sip.c: Merged revisions 318548 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r318548 | twilson | 2011-05-11 12:15:39 -0500 (Wed, 11 May 2011)
| 19 lines Clean up several chan_sip reference leaks Several
situations in the code could lead to peers or sip_pvt references
being leaked. This would cause RTP ports to never be destroyed
(leading to exhaustion of all available RTP ports) and memory
leaks. The original patch for this issue from rgagnon was the
result of an obscene amount of testing and hard work, for which I
am very grateful. I did some cleanup and added a few additional
refcount fixes that I found. (closes issue #17255) Reported by:
kvveltho Patches: tag-1.6.2.17-r309252-sip-dos-mem-leak-fix.diff
uploaded by rgagnon (license 1202) Tested by: rgagnon, twilson,
wdoekes, loloski Review: https://reviewboard.asterisk.org/r/1101/
Review: https://reviewboard.asterisk.org/r/1207/ Review:
https://reviewboard.asterisk.org/r/1210/ ........
2011-05-10 23:41 +0000 [r318499] Richard Mudgett <rmudgett@digium.com>
* channels/sig_pri.c, channels/sig_ss7.c: Unable to pickup
DAHDI/PRI call because call state is reported as DIALING. The
channel state is not updated to RINGING when an ALERTING message
is received. Regression caused when sig_pri.c (also sig_ss7.c)
extracted from chan_dahdi.c. * Added missing channel state update
to RINGING when the AST_CONTROL_RINGING frame is queued for ISDN
and SS7. (closes issue #19257) Reported by: alecdavis Patches:
issue19257_v1.8_v2.patch uploaded by rmudgett (license 664)
Tested by: alecdavis, rmudgett
2011-05-10 18:46 +0000 [r318485] Leif Madsen <lmadsen@digium.com>
* main/manager.c: Filter out blacklisted manager events when using
eventfilter. Merging change from trunk in revision 306432.
(closes issue #19260) Reported by: dhubbard Tested by: dhubbard
2011-05-10 15:13 +0000 [r318436] Russell Bryant <russell@digium.com>
* channels/chan_iax2.c: chan_iax2: change LOG_NOTICE to LOG_DEBUG
in iax2_read().
2011-05-09 23:15 +0000 [r318351] Richard Mudgett <rmudgett@digium.com>
* res/Makefile, res/res_features.exports.in (removed): Remove
references to res_features and its export file. The contents of
res/res_features.c was moved to into main/features.c awhile ago.
There is no longer any need for the res/Makefile to reference
res_features or the res_features linker exports file to exist.
2011-05-09 20:23 +0000 [r318337] Terry Wilson <twilson@digium.com>
* /, channels/chan_sip.c: Merged revisions 318331 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r318331 | twilson | 2011-05-09 15:04:41 -0500 (Mon, 09 May 2011)
| 12 lines Don't offer video to directmedia callee unless caller
offered it as well Make sure that when directmedia is enabled,
that video is not offered to the callee even if it supports it.
p->vrtp will not exist since the caller didn't offer video.
(closes issue #19195) Reported by: one47 Patches:
sip_cant_add_video_rtp uploaded by one47 (license 23) ........
2011-05-09 19:07 +0000 [r318282] Richard Mudgett <rmudgett@digium.com>
* main/features.c: Hangup extension executed twice. When a user
hangs up a call, in certain circumstances, the hangup extension
can end up being executed twice: 1) If a call is bridged and the
'h' extension executes the Hangup application, then the 'h'
extension will be executed twice. 2) If a call is bridged within
a macro (Dial or Queue), it has its own 'h' extension, the main
context also has an 'h' extension, and the macro 'h' extension
executes the Hangup application, then both 'h' extensions will be
executed. * Revert originally commited fix for #16106 and just
set AST_FLAG_BRIDGE_HANGUP_RUN unconditionally in
ast_bridge_call(). The bridge code just executed an 'h' extension
so the main PBX loop does not need to execute one as well. (issue
#16106) Reported by: ajohnson (issue #16548) Reported by: hajekd
2011-05-09 17:09 +0000 [r318233] David Vossel <dvossel@digium.com>
* /, channels/chan_sip.c: Merged revisions 318230 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r318230 | dvossel | 2011-05-09 11:51:45 -0500 (Mon, 09 May 2011)
| 7 lines Fixes cases where sip_set_rtp_peer can return too early
during media path reset. (closes issue #19225) Reported by: one47
Patches: sip_set_rtp_peer.patch uploaded by one47 (license 23)
........
2011-05-09 16:57 +0000 [r318231] Richard Mudgett <rmudgett@digium.com>
* channels/sig_pri.c: Don't get early media for ISDN on outgoing
calls. It looks to be a long-standing misinterpretation of the
progress indicator ie values: 1 - Call is not end-to-end ISDN;
further call progress information may be available in-band. 8 -
In-band information or an appropriate pattern is now available.
Only value 8 is handled by chan_dahdi/sig_pri. The 1 value is not
handled as early media probably because the meaning of the second
half of it's description was overlooked. * Test to see if either
PRI_PROG_CALL_NOT_E2E_ISDN(1) or PRI_PROG_INBAND_AVAILABLE(8)
bits are set to open the media path. (closes issue #18868)
Reported by: isrl Patches: issue18868_19246_v1.8.patch uploaded
by rmudgett (license 664) Tested by: satish_lx .......... No
inband progress on PRI_EVENT_RINGING even if inband flag set. My
ISDN-PRI provider sends an ALERTING with "Inband information or
appropriate pattern now available", but Asterisk only generates
and passes the RING to the SIP extension, not the inband message.
Unfortunately, the inband message is not a ringback tone but a
prompt that says the number is not in service. The SIP extension
then hears two rings and the call is hungup which confuses the
caller. * Post an AST_CONTROL_PROGRESS as well as opening the
media path if inband audio is indicated with an ALERTING message.
(closes issue #19246) Reported by: cristiandimache Patches:
issue19246_v1.8.patch uploaded by rmudgett (license 664) Tested
by: cristiandimache
2011-05-09 14:18 +0000 [r318148] Jonathan Rose <jrose@digium.com>
* configs/features.conf.sample: Documenting an observed behavior of
features in features.conf. Since parkinglots use an integer for
the parkinglot extensions, leading zeros specified in the
configuration file are ignored.
2011-05-09 14:09 +0000 [r318142] Matthew Nicholson <mnicholson@digium.com>
* main/channel.c: Make indicate/control frames WRITE events on
framehooks. Also, if a framehook returns a non-control frame,
don't forward it to the channel. (closes issue #19251) Reported
by: irroot Patches: (modified) framehook_indicate.patch2 uploaded
by irroot (license 52) Tested by: irroot
2011-05-07 23:35 +0000 [r318055-318057] Russell Bryant <russell@digium.com>
* res/res_config_curl.c: res_config_curl: fix a crash with static
realtime. (closes issue #18413) Reported by: jmls Patches:
20101202__issue18413.diff.txt uploaded by tilghman (license 14)
Tested by: jmls
* channels/chan_iax2.c: chan_iax2: Don't overwrite port found with
an SRV lookup. (closes issue #17291) Reported by: jcovert
Patches: chan_iax2.c.1.8.3-srvlookup-corrected.patch uploaded by
jcovert (license 551)
2011-05-06 21:49 +0000 [r317967-317969] Russell Bryant <russell@digium.com>
* apps/app_meetme.c: Use the right variable to print the time in a
debug message. The original patch also increased some buffer
sizes, but that was already done in this version. (closes issue
#17034) Reported by: sysreq Patches: asterisk-issue-17034.patch
uploaded by sysreq (license 1009)
* apps/app_meetme.c: Fix some more "set but unused" compiler
warnings.
2011-05-06 21:06 +0000 [r317918] David Vossel <dvossel@digium.com>
* res/res_rtp_asterisk.c: Fixes missing colon from To/From headers
in RTCP manager events. (closes issue #18221) Reported by:
clegall_proformatique Patches: 18221_1.patch uploaded by ebroad
(license 878)
2011-05-06 21:06 +0000 [r317861-317917] Russell Bryant <russell@digium.com>
* main/pbx.c: Fix calculation of free RAM to make minmemfree option
work. (closes issue #17124) Reported by: loic Patches: pbx_c.diff
uploaded by loic (license 1020)
* channels/chan_sip.c: chan_sip: Destroy variables on a sip_pvt
before copying vars from the sip_peer. Don't duplicate variables
on the sip_pvt. Just reset the variable list each time. (closes
issue #19202) Reported by: wdoekes Patches:
issue19202_destroy_challenged_invite_chanvars.patch uploaded by
wdoekes (license 717)
* channels/chan_sip.c: chan_sip: fix a deadlock in
check_rtp_timeout. Don't block doing silly deadlock avoidance.
Just return and try again later. The funciton gets called often
enough that it's fine. Also, this change was already made in
trunk. (closes issue #18791) Reported by: irroot Patches:
chan_sip.rtptimeout.patch uploaded by irroot (license 52)
* channels/chan_sip.c: URI encode less characters in the RPID and
Contact headers. If this change causes any problems, we will need
to backport the more extensive uri encoding and decoding handling
changes that are in trunk/1.10. (closes issue #18686) Reported
by: wolfgang Patches: quick-and-dirty.patch uploaded by wdoekes
(license 717) Tested by: wdoekes, devellow, wolfgang, mav3rick
2011-05-06 19:31 +0000 [r317858] Matthew Nicholson <mnicholson@digium.com>
* pbx/pbx_lua.c: pbx_lua autoservice fixes Don't start an
autoservice in pbx_lua if pbx_lua already started one and don't
stop one if we didn't start one. Also start and stop the
autoservice when transferring control from and to the pbx.
2011-05-06 19:24 +0000 [r317805-317837] Russell Bryant <russell@digium.com>
* addons/app_mysql.c: Fix a crash in the MySQL() application. This
code was not handling channel datastores safely. The channel must
be locked. (closes issue #17964) Reported by: wuwu Patches:
issue17964_addon_1.6.2_svn.patch uploaded by seanbright (license
71) Tested by: wuwu
* contrib/realtime/mysql/sipfriends.sql: Add a new sipfriends.sql
for MySQL that has more fields in it. (closes issue #16399)
Reported by: pabelanger Patches: sipfriends.sql.v3 uploaded by
pabelanger (license 224)
2011-05-06 16:19 +0000 [r317670] Richard Mudgett <rmudgett@digium.com>
* channels/chan_sip.c: Fix SIP connected line updates. This patch
fixes a couple SIP connected line update problems: 1) The
connected line needs to be updated when the initial INVITE is
sent if there is a peer callerid configured. Previously, the
connected line information did not get reported until the call
was connected so SIP could not report connected line information
in ringing or progress messages. 2) The connected line should not
be updated on initial connect if there is no connected line
information. Previously, all it did was wipe out any default
preset CONNECTEDLINE information set by the dialplan with empty
strings. (closes issue #18367) Reported by: GeorgeKonopacki
Patches: issue18367_v1.8.patch uploaded by rmudgett (license 664)
Tested by: rmudgett Review:
https://reviewboard.asterisk.org/r/1199/
2011-05-06 08:18 +0000 [r317584] Terry Wilson <twilson@digium.com>
* apps/app_queue.c, /: Merged revisions 317575 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r317575 | twilson | 2011-05-06 01:04:17 -0700
(Fri, 06 May 2011) | 13 lines Merged revisions 317574 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r317574 | twilson | 2011-05-06 00:55:21 -0700 (Fri, 06 May 2011)
| 6 lines Re-fix queue round-robin This part of the change for
r315596 was incorrect. No bridge occurs when doing a roundrobin
dial and no one answers, so this code shouldn't have been
removed. ........ ................
2011-05-05 23:46 +0000 [r317425-317530] Russell Bryant <russell@digium.com>
* Makefile: If the configure script runs, force a rebuild of
menuselect-tree. Some contents in the menuselect tree are
dependent on configure script parameters, namely
--enable-dev-mode. (closes issue #17219) Reported by: Nick_Lewis
Patches: issue_17219.rev1.txt uploaded by russell (license 2)
* contrib/realtime/mysql/queue_log.sql,
contrib/realtime/mysql/sipfriends.sql: Fix some more realtime
MySQL schema issues. (closes issue #18537) Reported by: denzs
Patches: sipfriends.sql.svndiff uploaded by denzs (license 1182)
queue_log.sql.svndiff uploaded by denzs (license 1182)
meetme.sql.svndiff uploaded by denzs (license 1182)
* contrib/realtime/mysql/sipfriends.sql,
contrib/realtime/mysql/meetme.sql: Fix some errors in sample
MySQL realtime schema files. (closes issue #18915) Reported by:
Dovid Patches: sipfriends.patch uploaded by Dovid (license 652)
meetme.patch uploaded by Dovid (license 652)
* cdr/cdr_syslog.c: Don't lose cdr_syslog config on a reload.
(closes issue #18679) Reported by: enegaard Patches:
issue18679_seanbright.patch uploaded by seanbright (license 71)
Tested by: enegaard
* channels/chan_alsa.c, channels/chan_console.c,
channels/chan_oss.c, channels/chan_mgcp.c,
channels/misdn_config.c, channels/chan_unistim.c,
channels/chan_usbradio.c, channels/chan_dahdi.c,
channels/chan_sip.c, channels/chan_skinny.c,
channels/chan_h323.c: Fix some consistency issues with
jitterbuffer config. Store the defaults noted in the sample
config files in the jitterbuffer config data structure. This
makes the CLI commands that output these settings show the right
thing. Also only show the settings that are relevant in the
settings CLI commands, based on which jitterbuffer is selected
and whether it's enabled. (closes issue #19083) Reported by:
rgagnon Patches: issue-19083-trunk-r313139.diff uploaded by
rgagnon (license 1202)
* pbx/pbx_lua.c: Add a datastore fixup to fix a pbx_lua crash.
(closes issue #19055) Reported by: jamhed Patches:
lua_datastore_fixup1.diff uploaded by mnicholson (license 96)
Tested by: mnicholson, jamhed
* channels/iax2-provision.c, pbx/pbx_dundi.c,
channels/chan_console.c, cdr/cdr_radius.c, channels/chan_iax2.c,
res/res_jabber.c, res/res_config_sqlite.c, cel/cel_pgsql.c,
channels/chan_jingle.c, channels/sip/sdp_crypto.c,
res/res_config_odbc.c, channels/chan_sip.c, res/res_crypto.c,
pbx/pbx_lua.c: Fix more "set but unused" warnings.
* main/dsp.c: Only display inband DTMF warning if inband DTMF
detection is enabled. (closes issue #18901) Reported by: irroot
* apps/app_rpt.c: Fix potential memory leak, and use of
uninitialized memory. (closes issue #16476) Reported by: junky
Patches: M16476.diff uploaded by junky (license 177)
* main/manager.c: Add missing ActioID handling to Events action.
(closes issue #18949) Reported by: edersohe Patches:
0018949.patch uploaded by edersohe (license 1228)
2011-05-05 20:25 +0000 [r317370] Sean Bright <sean@malleable.com>
* addons/res_config_mysql.c: Don't duplicate our data on the stack
and just use the MYSQL_ROW directly. With large result sets we
were blowing out the stack. (closes issue #19090) Reported by:
mickecarlsson Patches: issue19090_trunk_svn.patch uploaded by
seanbright (license 71) Tested by: mickecarlsson
2011-05-05 19:55 +0000 [r317336] Russell Bryant <russell@digium.com>
* apps/app_queue.c: Increase buffer size to be PATH_MAX for a path.
(closes issue #19239) Reported by: byronclark Patches:
queue_announce_length.patch uploaded by byronclark (license 1200)
2011-05-05 19:09 +0000 [r317283] Jonathan Rose <jrose@digium.com>
* channels/chan_sip.c: Resolves a deadlock that occurs during
sip_new This is based on an uncommitted patch by jpeeler for the
issue. Instead of relocking and then unlocking the channel
though, we keep the lock on the channel until we are finished
doing what we need to the channel. (closes issue #18441) Reported
by: Alric
2011-05-05 18:39 +0000 [r317280-317281] Russell Bryant <russell@digium.com>
* /, channels/chan_sip.c: Merged revisions 317255 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r317255 | russell | 2011-05-05 13:29:53 -0500
(Thu, 05 May 2011) | 22 lines Merged revisions 317211 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r317211 | russell | 2011-05-05 13:20:29 -0500 (Thu, 05 May 2011)
| 15 lines chan_sip: fix broken realtime peer count, fix memory
leak This patch addresses two bugs in chan_sip: 1) The count of
realtime peers and users was off. The increment checked the value
of the caching option, while the decrement did not. 2) Add a
missing regfree() for a regex. (closes issue #19108) Reported by:
vrban Patches: missing_regfree.patch uploaded by vrban (license
756) sip_object_counter.patch uploaded by vrban (license 756)
........ ................
* /: Restore branch-1.6.2-merged and branch-1.6.2-blocked
properties.
2011-05-05 18:02 +0000 [r317196] Matthew Nicholson <mnicholson@digium.com>
* channels/chan_sip.c: Set SO_KEEPALIVE on SIP TCP sockets so that
they eventually go away when a peer abruptly disappears. This
mostly occurs after a successful registration. (closes issue
#17544) Reported by: marcelloceschia Patches: (modified)
tcptls.patch uploaded by st (license 907)
2011-05-05 15:04 +0000 [r317058-317104] Leif Madsen <lmadsen@digium.com>
* contrib/scripts/safe_asterisk, /: Merged revisions 317102 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r317102 | lmadsen | 2011-05-05 10:54:46 -0400 (Thu, 05 May 2011)
| 8 lines Disable console colourization inside safe_asterisk
checks. (closes issue #19213) Reported by: lefoyer Patches:
issue19213_strip_color_in_safe_asterisk-svn.patch uploaded by
wdoekes (license 717) Tested by: wdoekes, lefoyer ........
* Makefile, configs/cel.conf.sample: Remove unused directory and
clear up some documentation. (closes issue #19193) Reported by:
bchia Patches: cel-csv.diff uploaded by lathama (license 1028)
Tested by: lathama, Marquis42
2011-05-05 02:30 +0000 [r316917-316919] Sean Bright <sean@malleable.com>
* main/http.c: Use the correct HTTP method when generating our
digest, otherwise we always fail. When calculating the 'A2'
portion of our digest for verification, we need the HTTP method
that is currently in use. Unfortunately our mapping function was
incorrect, resulting in invalid hashes being generated and, in
turn, failures in authentication. (closes issue #18598) Reported
by: ksn
* main/utils.c: Look at the correct buffer for our digest info
instead of an empty one. (issue #18598) Reported by: ksn
* main/manager.c: Make sure that tcptls_session is properly
initialized. (issue #18598) Reported by: ksn
2011-05-04 20:50 +0000 [r316874] Alexandr Anikin <may@telecom-service.ru>
* addons/ooh323c/src/ooSocket.c: Fix trivial bug in ooSocket.c
codes Revert condition for result code of ast_gethostbyname
(closes issue #19185) Reported by: dswartz Patches:
issue19185-patch uploaded by may213 (license 454)
2011-05-04 18:51 +0000 [r316831] Richard Mudgett <rmudgett@digium.com>
* apps/app_meetme.c: Wait for leader with Music On Hold allows
crosstalk between participants. Parenthesis in the wrong
position. Regression from issue #14365 when expanding conference
flags to use 64 bits. (closes issue #18418) Reported by: MrHanMan
Tested by: rmudgett
2011-05-04 16:15 +0000 [r316663-316709] Sean Bright <sean@malleable.com>
* apps/app_voicemail.c, /: Merged revisions 316708 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r316708 | seanbright | 2011-05-04 12:10:59 -0400
(Wed, 04 May 2011) | 15 lines Merged revisions 316707 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r316707 | seanbright | 2011-05-04 12:08:50 -0400 (Wed, 04 May
2011) | 8 lines If sox fails when processing a voicemail, don't
delete the original file. (closes issue #18111) Reported by:
sysreq Patches: issue18111_trunk.patch uploaded by seanbright
(license 71) Tested by: seanbright ........ ................
* main/manager.c: Only return a single error via AMI when
requesting a forbidden action. (closes issue #19216) Reported by:
oej Patches: issue19216-1.8-r316204.patch uploaded by seanbright
(license 71) Tested by: seanbright
2011-05-04 14:25 +0000 [r316617-316650] David Vossel <dvossel@digium.com>
* apps/app_chanspy.c, /: Merged revisions 316644 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r316644 | dvossel | 2011-05-04 09:23:39 -0500 (Wed, 04 May 2011)
| 9 lines Fixes one-way-audio when chanspy activated with the 'o'
option (closes issue #18382) Reported by: jkister Patches:
0001-Bugfix-18382-one-way-audio-when-chanspy-activated.patch.txt
uploaded by malin (license ) Tested by: firstsip, Greenlightcrm,
malin, wdoekes, boroda, dvossel ........
* /, channels/chan_sip.c: Merged revisions 316616 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r316616 | dvossel | 2011-05-04 08:40:41 -0500 (Wed, 04 May 2011)
| 12 lines Fixes session-timers=refuse not being enforced for
*caller* During handle_request_invite, the session timer mode was
retrieved from a cached variable. This patch forces a peer lookup
of the session timer mode in the case of an incoming invite.
(closes issue #18804) Reported by: wdoekes Patches:
issue18804_session_timer_refuse_caller.patch uploaded by wdoekes
(license 717) issue_18804_v2.diff uploaded by dvossel (license
671) ........
2011-05-04 02:34 +0000 [r316476] Sean Bright <sean@malleable.com>
* /, apps/app_meetme.c: Merged revisions 316475 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r316475 | seanbright | 2011-05-03 22:23:01 -0400 (Tue, 03 May
2011) | 10 lines Honor the C option to MeetMe when L is passed.
This fixes a case that r304773 and friends missed. (closes issue
#17317) Reported by: var Patches: meetme-continue-on-l_16218.diff
uploaded by var (license 1227) Tested by: seanbright ........
2011-05-04 00:12 +0000 [r316429] Tilghman Lesher <tilghman@meg.abyt.es>
* addons/cdr_mysql.c, addons/res_config_mysql.c: Escape column
names in case they contain illegal characters ('-') or reserved
words. (closes issue #19063) Reported by: festr Patches: patch
uploaded by festr (license 443)
2011-05-03 22:13 +0000 [r316336] Russell Bryant <russell@digium.com>
* pbx/pbx_dundi.c, channels/chan_mgcp.c, channels/chan_skinny.c:
Use htons() instead of ntohs() in some places. (closes issue
#19200) Reported by: wdoekes Patches: issue19200-trunk.patch
uploaded by wdoekes (license 717) issue19200-1.8.x.patch uploaded
by wdoekes (license 717)
2011-05-03 22:05 +0000 [r316334] David Vossel <dvossel@digium.com>
* main/channel.c: Fixes framehook segfault on indicate (closes
issue #19215) Reported by: irroot Patches:
framehook_indicate.patch uploaded by irroot (license 52)
2011-05-03 21:41 +0000 [r316331] Russell Bryant <russell@digium.com>
* apps/app_minivm.c: Resolve another warning.
2011-05-03 21:37 +0000 [r316330] David Vossel <dvossel@digium.com>
* channels/chan_local.c, /: Merged revisions 316329 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r316329 | dvossel | 2011-05-03 16:29:55 -0500
(Tue, 03 May 2011) | 17 lines Merged revisions 316328 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r316328 | dvossel | 2011-05-03 16:27:59 -0500 (Tue, 03 May 2011)
| 10 lines Fixes chan_local crashs in local_fixup() Thanks OEJ
for tracking down the issue and submitting the patch. (closes
issue #19053) Reported by: oej Tested by: oej Review:
https://reviewboard.asterisk.org/r/1158/ ........
................
2011-05-03 19:55 +0000 [r316265] Russell Bryant <russell@digium.com>
* res/res_musiconhold.c, apps/app_ices.c, apps/app_followme.c,
main/config.c, main/cdr.c, main/channel.c, channels/chan_phone.c,
funcs/func_enum.c, main/manager.c, channels/chan_skinny.c,
res/res_agi.c, main/plc.c, main/features.c, apps/app_minivm.c,
apps/app_amd.c, main/pbx.c, res/res_fax.c, formats/format_wav.c,
apps/app_festival.c, channels/chan_agent.c, apps/app_originate.c,
apps/app_queue.c, codecs/lpc10/dyptrk.c,
include/asterisk/linkedlists.h, main/audiohook.c,
pbx/pbx_config.c, main/asterisk.c, main/dsp.c,
res/res_calendar.c, apps/app_voicemail.c, main/udptl.c,
channels/chan_unistim.c, main/fskmodem_float.c,
main/rtp_engine.c: Fix a bunch of compiler warnings generated by
gcc 4.6.0. Most of these are -Wunused-but-set-variable, but there
were a few others mixed in here, as well.
2011-05-03 19:18 +0000 [r316224] Richard Mudgett <rmudgett@digium.com>
* channels/sig_pri.c, channels/chan_dahdi.c, channels/sig_analog.c:
The dahdi_hangup() call does not clean up the channel fully.
After dahdi_hangup() has supposedly hungup an ISDN channel there
is still traffic on the S0-bus because the channel was not
cleaned up fully. Shuffled the hangup code to include some
missing cleanup. Also fixed some code formatting in the area. I
think the primary missing clean up code was the call to
tone_zone_play_tone() to turn off any active tones on the
channel. (closes issue #19188) Reported by: jg1234 Patches:
issue19188_v1.8.patch uploaded by rmudgett (license 664) Tested
by: jg1234
2011-05-03 18:59 +0000 [r316215-316217] David Vossel <dvossel@digium.com>
* channels/chan_sip.c: Never put the Require: timer header in an
Invite. This has already been discussed and should have been
resolved earlier. View revsion 285565's log for more information
about why it is important to not put timer in the Require header.
(closes issue #18704) Reported by: mfrager
* res/res_odbc.c: Fixes a random crash (NULL reference) in
res_odbc.c. (closes issue #19180) Reported by: pruiz Patches:
tmp.diff uploaded by pruiz (license 1152) Tested by: pruiz,
seanbright
2011-05-03 18:17 +0000 [r316206] Sean Bright <sean@malleable.com>
* main/manager.c: If we aren't interested in events, don't generate
the FullyBooted event on AMI login. (closes issue #19089)
Reported by: bklang Patches: issue19089-1.8-r316204.patch
uploaded by seanbright (license 71) Tested by: seanbright
2011-05-03 10:57 +0000 [r316193] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
* autoconf/ast_check_pwlib.m4, configure: Re-fix bashism in
./configure: s/let/$(( ))/ A forward-port in r278985 accidentally
re-introduced issue 17485. Fixing it. Thanks to Jilles Tjoelker
for the good report. (closes issue #17485)
2011-05-02 19:09 +0000 [r316094] Tilghman Lesher <tilghman@meg.abyt.es>
* funcs/func_curl.c, /: Merged revisions 316093 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r316093 | tilghman | 2011-05-02 14:04:36 -0500 (Mon, 02 May 2011)
| 8 lines More possible crashes based upon invalid inputs.
(closes issue #18161) Reported by: wdoekes Patches:
20110301__issue18161.diff.txt uploaded by tilghman (license 14)
Tested by: wdoekes ........
2011-04-27 19:14 +0000 [r315894] Matthew Nicholson <mnicholson@digium.com>
* /, channels/chan_sip.c, channels/sip/reqresp_parser.c: Merged
revisions 315893 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r315893 | mnicholson | 2011-04-27 14:03:05 -0500
(Wed, 27 Apr 2011) | 21 lines Merged revisions 315891 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r315891 | mnicholson | 2011-04-27 13:57:56 -0500 (Wed, 27 Apr
2011) | 14 lines Fix our compliance with RFC 3261 section 18.2.2.
This change optimizes the free_via() function and removes some
redundant null checking. It also fixes compliance with RFC 3261
section 18.2.2 by always using the port specified in the Via
header for routing responses (even when maddr is not set). Also
the htons() function is now used when setting the port.
Additional documentation comments have been added in various
places to make the logic in the code clearer. (closes issue
#18951) Reported by: jmls Patches:
issue18951_set_proper_port_from_via.patch uploaded by wdoekes
(license 717) (modified) ........ ................
2011-04-27 15:55 +0000 [r315810] Russell Bryant <russell@digium.com>
* main/asterisk.c: Set the copyright year to 2011 in the startup
message.
2011-04-27 12:36 +0000 [r315765] Leif Madsen <lmadsen@digium.com>
* sounds/sounds.xml, sounds/Makefile: Enable Russian core sound
selection in menuselect. (closes issue #18724) Reported by:
pbxware
2011-04-26 22:56 +0000 [r315673] Terry Wilson <twilson@digium.com>
* /, channels/chan_sip.c: Merged revisions 315672 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r315672 | twilson | 2011-04-26 15:52:25 -0700
(Tue, 26 Apr 2011) | 18 lines Merged revisions 315671 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r315671 | twilson | 2011-04-26 15:47:56 -0700 (Tue, 26 Apr 2011)
| 11 lines Make sure unregistering a peer unlinks it from the
peer container Instead of mostly copying the code from
expire_register, just use the function that "does the right
thing". (closes issue #16033) Reported by: kkm Patches:
016033-tilgman-fixed-refcount.diff uploaded by kkm (license 888)
Tested by: kkm, tilghman, twilson ........ ................
2011-04-26 22:14 +0000 [r315645] Richard Mudgett <rmudgett@digium.com>
* main/pbx.c: The 'e' special extension fails to trigger in at
least two cases. The 'e' extension is a fall back for the 'i',
't', or 'T' extensions if any of them do not exist. Many of the
places the 'e' extension was supposed to be invoked fail because
the priority was set wrong. There were two places where the 'e'
extension was not even checked for fall back. * Made invoke the
'e' extension similarly to the previous 'i', 't', or 'T'
extension check and added the 'e' extension as a fall back to the
two missing locations. * Prioritized and optimized some hangup
tests associated with the 'e' extension. (closes issue #19136)
Reported by: kshumard Tested by: rmudgett Review:
https://reviewboard.asterisk.org/r/1196/
2011-04-26 21:39 +0000 [r315644] Terry Wilson <twilson@digium.com>
* apps/app_queue.c, apps/app_dial.c, /, main/features.c: Merged
revisions 315643 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r315643 | twilson | 2011-04-26 14:27:44 -0700
(Tue, 26 Apr 2011) | 25 lines Merged revisions 315596 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r315596 | twilson | 2011-04-26 14:16:10 -0700 (Tue, 26 Apr 2011)
| 18 lines Allow transfer loops without allowing forwarding loops
We try to avoid the situation where two phones may be forwarded
to each other causing an infinite loop by storing each dialed
interface in a channel datastore and checking the list before
dialing out. This works, but currently breaks situations like A
calls B, A transfers B to C, B transfers C to A, and A transfers
C to B. Since human interaction is happening here and not an
automated forwarding loop, it should be allowed. This patch
removes the dialed_interfaces datastore when a call is bridged (a
suggestion from the brilliant mmichelson). If a call is being
bridged, it should be safe to assume that we aren't stuck in a
loop. Since we are now handling this is the bridge code, the
previous attempts at handling it in app_dial and app_queue are
removed. Review: https://reviewboard.asterisk.org/r/1195/
........ ................
2011-04-26 19:32 +0000 [r315503] Tilghman Lesher <tilghman@meg.abyt.es>
* include/asterisk/select.h, /: Merged revisions 315502 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r315502 | tilghman | 2011-04-26 14:22:52 -0500
(Tue, 26 Apr 2011) | 21 lines Merged revisions 315501 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r315501 | tilghman | 2011-04-26 14:18:46 -0500 (Tue, 26 Apr 2011)
| 14 lines Fix the bounds-checking code. The code that set the
bit within the select bitfield was correct, but the
bounds-checking code was not. The change to that line uses the
new _bitsize macro for clarity. Also, FD_ZERO macro did not
zero-out anything but the first word of the bitfield, so this
could have caused problems with modules using that macro with the
expanded bitfield. (closes issue #18773) Reported by: jamicque
Patches: 20110423__issue18773.diff.txt uploaded by tilghman
(license 14) Tested by: chris-mac ........ ................
2011-04-26 18:00 +0000 [r315452] Richard Mudgett <rmudgett@digium.com>
* apps/app_dial.c: Add missing set of name valid flag when dialing.
2011-04-26 17:40 +0000 [r315446] Russell Bryant <russell@digium.com>
* channels/chan_local.c: chan_local: resolve a deadlock. This patch
resolves a fairly complex deadlock that can occur with the
combination of chan_local and a dialplan switch, such as dynamic
realtime extensions, which pulls autoservice into the picture
when doing a dialplan lookup. (closes issue #18818) Reported by:
nic Patches: issue18818.patch uploaded by jthurman (license 614)
18818.v1.txt uploaded by russell (license 2) Tested by: nic,
jthurman, kterzi, steve-howes, sysreq, IshMalik
2011-04-26 02:18 +0000 [r315394] Paul Belanger <pabelanger@digium.com>
* pbx/pbx_config.c, /: Merged revisions 315393 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r315393 | pabelanger | 2011-04-25 22:17:43 -0400 (Mon, 25 Apr
2011) | 7 lines Add back CLI command 'dialplan save' (closes
issue #19140) Reported by: lmadsen Patches:
__20110419_dialplan_save.patch.txt uploaded by lmadsen (license
10) ........
2011-04-25 21:49 +0000 [r315349] Richard Mudgett <rmudgett@digium.com>
* channels/chan_mgcp.c: When using MGCP realtime gateway
definitions, random crashes occur. Fixed incorrect linked list
node removal for realtime gateways. (closes issue #18291)
Reported by: nahuelgreco Patches:
dangling-pointers-when-pruning.patch uploaded by nahuelgreco
(license 162)
2011-04-25 19:37 +0000 [r315213-315259] Russell Bryant <russell@digium.com>
* /, formats/format_wav.c: Merged revisions 315258 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r315258 | russell | 2011-04-25 14:31:44 -0500
(Mon, 25 Apr 2011) | 17 lines Merged revisions 315257 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r315257 | russell | 2011-04-25 14:28:41 -0500 (Mon, 25 Apr 2011)
| 10 lines Be more flexible with unknown chunks in wav files.
This patch makes format_wav ignore unknown chunks instead of
erroring out on them. (closes issue #18306) Reported by: jhirsch
Patches: wav_skip_unknown_blocks.diff uploaded by jhirsch
(license 1156) ........ ................
* /, channels/chan_sip.c: Merged revisions 315212 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r315212 | russell | 2011-04-25 14:00:24 -0500 (Mon, 25 Apr 2011)
| 7 lines Don't link non-cached realtime peers into the
peers_by_ip container. (closes issue #18924) Reported by: wdoekes
Patches: issue18924_uncached_realtime_peers_leak-1.6.2.17.patch
uploaded by wdoekes (license 717) ........
2011-04-25 07:14 +0000 [r315053] Alec L Davis <sivad.a@paradise.net.nz>
* channels/chan_local.c, /: Merged revisions 315052 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r315052 | alecdavis | 2011-04-25 19:11:12 +1200
(Mon, 25 Apr 2011) | 16 lines Merged revisions 315051 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r315051 | alecdavis | 2011-04-25 19:06:29 +1200 (Mon, 25 Apr
2011) | 11 lines chan_local:check_bridge() misplaced misplaced
ast_mutex_unlock if !p->chan->_bridge->_softhangup path isn't
followed, brigde remains locked. (closes issue #19176) Reported
by: alecdavis Patches: bug19176.diff.txt uploaded by alecdavis
(license 585) ........ ................
2011-04-22 22:59 +0000 [r315001] Alec L Davis <sivad.a@paradise.net.nz>
* channels/chan_dahdi.c: chan_dahdi: Can't return to normal ring
after distinctive ring on FXS clear a previous distinctivering
pattern before each new call (closes issue #18985) Reported by:
bromont Patches: bug18985.diff.txt uploaded by alecdavis (license
585) Tested by: alecdavis, bromont
2011-04-22 21:20 +0000 [r314959] Matthew Nicholson <mnicholson@digium.com>
* /, channels/chan_agent.c: Merged revisions 314958 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r314958 | mnicholson | 2011-04-22 15:49:45 -0500
(Fri, 22 Apr 2011) | 17 lines Merged revisions 311203,314908 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r311203 | mnicholson | 2011-03-17 14:14:37 -0500 (Thu, 17 Mar
2011) | 4 lines Don't hold the pvt lock while streaming a file.
ABE-2756 ........ r314908 | mnicholson | 2011-04-22 15:01:48
-0500 (Fri, 22 Apr 2011) | 4 lines Prevent the login thread and
the app threads from using the asterisk channel at the same time.
ABE-2756 ........ ................
2011-04-22 14:02 +0000 [r314780] Russell Bryant <russell@digium.com>
* /, res/res_agi.c: Merged revisions 314778 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r314778 | russell | 2011-04-22 08:58:03 -0500 (Fri, 22 Apr 2011)
| 11 lines Initialize buffers in getvar and getvarfull.
Initialize the buffers used to hold the result from GET VARIABLE
or GET VARIABLE FULL. The bug report shows func_read returning
garbage in the result. It assumed that the buffer passed in was
initialized, like many other functions do. In the more common
code path (through the dialplan), it is initialized, so just
initialize it here too. (closes issue #19050) Reported by: johnz
........
2011-04-22 13:59 +0000 [r314779] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
* res/res_fax_spandsp.c, channels/chan_unistim.c: Fix a few typos
(shown by Lintian)
2011-04-22 13:35 +0000 [r314777] Russell Bryant <russell@digium.com>
* /: Recorded merge of revisions 314776 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r314776 | russell | 2011-04-22 08:35:22 -0500 (Fri, 22 Apr 2011)
| 10 lines Fix handling of some call parking config options. This
patch adjusts the handling of some call parking config options to
fix some issues that have already been addressed in 1.8 and
trunk. (closes issue #19167) Reported by: bluecrow76 Patches:
asterisk-1.6.2.17.2-fix-build-parkinglot-parked-AST_FEATURE_FLAGS.diff
uploaded by bluecrow76 (license 270) ........
2011-04-21 22:38 +0000 [r314732] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c: Correct DAHDIShowChannels XML
documentation.
2011-04-21 18:24 +0000 [r314628] Matthew Nicholson <mnicholson@digium.com>
* configs/sip.conf.sample, configs/skinny.conf.sample,
channels/sip/include/sip.h, configs/http.conf.sample,
main/manager.c, /, channels/chan_sip.c, channels/chan_skinny.c,
main/http.c: Merged revisions 314620 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r314620 | mnicholson | 2011-04-21 13:22:19 -0500
(Thu, 21 Apr 2011) | 20 lines Merged revisions 314607 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r314607 | mnicholson | 2011-04-21 13:19:21 -0500 (Thu, 21 Apr
2011) | 14 lines Added limits to the number of unauthenticated
sessions TCP based protocols are allowed to have open
simultaneously. Also added timeouts for unauthenticated sessions
where it made sense to do so. Unrelated, the manager interface
now properly checks if the user has the "system" privilege before
executing shell commands via the Originate action. AST-2011-005
AST-2011-006 (closes issue #18787) Reported by: kobaz (related to
issue #18996) Reported by: tzafrir ........ ................
2011-04-21 00:23 +0000 [r314550] Terry Wilson <twilson@digium.com>
* /, channels/chan_sip.c: Merged revisions 314549 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r314549 | twilson | 2011-04-20 17:17:34 -0700 (Wed, 20 Apr 2011)
| 6 lines Don't allocate more space than necessary for a sip_pkt
This extra allocation is a hold-over from when pkt->data was a
character array. Now that it is an allocated string, just
allocate enough for the sip_pkt. ........
2011-04-20 16:54 +0000 [r314417] Richard Mudgett <rmudgett@digium.com>
* include/asterisk/frame.h: AST_CONTROL_XXX comment changes.
2011-04-20 05:25 +0000 [r314358] Terry Wilson <twilson@digium.com>
* main/lock.c: Initialize track pointer ast_reentrancy_init checks
to see if it is NULL before initializing with calloc
2011-04-19 15:42 +0000 [r314203-314251] Leif Madsen <lmadsen@digium.com>
* main/tcptls.c: Use SSLv23_client_method instead of old SSLv2
only. (closes issue #19095) (closes issue #19138) Reported by:
tzafrir Patches: no_ssl2.diff uploaded by tzafrir (license 46)
Tested by: russell, chazzam
* /, funcs/func_channel.c: Merged revisions 314205 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
........ r314205 | lmadsen | 2011-04-19 09:27:50 -0500 (Tue, 19
Apr 2011) | 6 lines Remove duplicate documentation from
func_channel.c (closes issue #18970) Reported by: IgorG Patches:
func_channel.c.doc.diff uploaded by IgorG (license 20) ........
* apps/app_dial.c, /: Merged revisions 314202 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r314202 | lmadsen | 2011-04-19 09:23:39 -0500 (Tue, 19 Apr 2011)
| 7 lines Update seconds to milliseconds in ast_verb output.
(closes issue #19084) Reported by: smurfix Patches:
app_dial.patch uploaded by smurfix (license 547) Tested by:
lmadsen, smurfix ........
2011-04-18 16:10 +0000 [r314068-314069] Richard Mudgett <rmudgett@digium.com>
* res/res_agi.c: The AsyncAGI command loop is lax in the value it
returns for the return status. * Return correct status:
SUCCESS/FAILED/HANGUP. Previously, abnormal exits from the
command loop such as hangup would return SUCCESS. * The "asyncagi
break" command now returns SUCCESS and is now the only way to
break the command loop with that status. Previously, it returned
FAILED. * The AMI event AsyncAGI End is no longer sent if the
AsyncAGI Start event is not sent. Previously, this happened
because of an error setting up the AGI pipes. * All executed AGI
commands now get an AsyncAGI Exec result event. Previously, if
the command returned failure (because of hangup), the command
loop just exited with FAILURE and did not send the AsyncAGI Exec
result event. * Makes sure that the channel frame queue is empty
on hangup. Review: https://reviewboard.asterisk.org/r/1183/
* apps/app_dial.c: Unclear code in app_dial.c. Make code formatting
clear. (closes issue #19134) Reported by: oej
2011-04-18 15:23 +0000 [r314017-314067] David Vossel <dvossel@digium.com>
* channels/chan_sip.c: Remove the need for deadlock avoidance in
chan_sip do_monitor. Deadlock avoidance between the sip pvt and
the pvt->owner is very difficult. Now that channel's are ao2
objects, this complication is no longer necessary. It turns out
the pvt's msg queue only exists because of deadlock avoidance
(when deadlock avoidance fails msgs were added to a queue to be
processed later), so this goes away as well. The technique used
in the new sip_lock_pvt_full() function should be used as a
template for replacing all locations where deadlock avoidance
occurs between a channel tech_pvt and the pvt's owner. My hope is
that this will begin a reversal of the invalid channel driver
locking architecture we have been using for so long. This patch
also resolves an issue where the pvt->owner gets unlocked during
processing the msg queue. (closes issue #18690) Reported by:
dvossel Review: https://reviewboard.asterisk.org/r/1182/
* include/asterisk/rtp_engine.h, main/rtp_engine.c,
channels/chan_sip.c: sip codec negotiation of dynamic rtp
payloads error fix This patch fixes how chan_sip handles dynamic
rtp payload types it does not understand. At the moment if a
dynamic payload's mime type does not match one we understand, the
payload does not get removed from our payload table. As a result
of this, the payload is set to whatever dynamic codec we use
internally for that payload number on outgoing INVITES. This is
incorrect. This patch fixes this by properly checking the rtpmap
set function's return code to make sure it was found. The
function can return both -1 and -2 depending on the source of the
mismatch. We were just checking -1 explicitly. Review:
https://reviewboard.asterisk.org/r/1169/
2011-04-15 15:08 +0000 [r313860] Jonathan Rose <jrose@digium.com>
* main/cli.c, /: Merged revisions 313859 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r313859 | jrose | 2011-04-15 09:58:37 -0500 (Fri, 15 Apr 2011) |
10 lines Fix a Tab Completion bug that occurs due to multiple
matches on a substring. Makes word_match function in cli.c repeat
a search for a command string until a proper match is found or
the string is searched to the last point. (closes issue #17494)
Reported by: ffossard Review:
https://reviewboard.asterisk.org/r/1180/ ........
2011-04-14 20:59 +0000 [r313517-313780] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c: Leftover debug messages unconditionally
sent to the console. Executing Dial(DAHDI/1/18475551212,300,)
with the echotraining config option enabled outputs the following
debug messages unconditionally: Dialing T1847555121 on 1 Dialing
www2w on 1 * Made debug messages in my_dial_digits() normal debug
messages that do not get output unless enabled. * Reworded some
debug messages in my_dial_digits() to be clearer. * Replace
strncpy() with ast_copy_string() in my_dial_digits() which does
the same job better. (closes issue #18847) Reported by:
vmikhelson Tested by: rmudgett
* res/res_agi.c: Revert flushing stale AsyncAGI commands from
-r313615. It looks like it was intentional to leave any commands
or in-flight commands in the queue in case Async AGI is run again
on the call.
* res/res_agi.c: Miscellaneous AGI diagnostic message cleanup and
code optimization.
* res/res_agi.c: * Add missing channel lock to
handle_cli_agi_add_cmd(). * Flush any Async AGI commands left
over from earlier Async AGI control of the call.
* main/channel.c, /, res/res_agi.c: Merged revisions 313579 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r313579 | rmudgett | 2011-04-13 11:29:49 -0500
(Wed, 13 Apr 2011) | 48 lines Merged revisions 313545 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r313545 | rmudgett | 2011-04-13 11:21:24 -0500 (Wed, 13 Apr 2011)
| 41 lines Asterisk does not hangup a channel after endpoint
hangs up. If the call that the dialplan started an AGI script for
is hungup while the AGI script is in the middle of a command then
the AGI script is not notified of the hangup. There are many AGI
Exec commands that this can happen with. The reported
applications have been: Background, Wait, Read, and Dial. Also
the AGI Get Data command. * Don't wait on the Asterisk channel
after it has hung up. The channel is likely to never need
servicing again. * Restored the AGI script's ability to return
the AGI_RESULT_HANGUP value in run_agi(). It previously only
could return AGI_RESULT_SUCCESS or AGI_RESULT_FAILURE after the
DeadAGI and AGI applications were merged. (closes issue #17954)
Reported by: mn3250 Patches: issue17954_v1.8.patch uploaded by
rmudgett (license 664) issue17954_v1.6.2.patch uploaded by
rmudgett (license 664) issue17954_v1.4.patch uploaded by rmudgett
(license 664) Tested by: rmudgett JIRA SWP-2171 (closes issue
#18492) Reported by: devmod Tested by: rmudgett JIRA SWP-2761
(closes issue #18935) Reported by: nvitaly Tested by: astmiv,
rmudgett JIRA SWP-3216 (closes issue #17393) Reported by: siby
Tested by: rmudgett JIRA SWP-2727 Review:
https://reviewboard.asterisk.org/r/1165/ ........
................
* apps/app_dumpchan.c: Bring the dumpchan application inline with
"core show channel". * Added fields that are in "core show
channel" to dumpchan output. * Fixed reuse of formatbuf before
the previous string stored there was used by snprintf. All output
strings now have their own buffer. * Adjusted the buffer sizes to
not be so abusive of the stack now that there are more buffers.
Change requested by oej.
2011-04-12 18:47 +0000 [r313434-313436] Jonathan Rose <jrose@digium.com>
* channels/chan_dahdi.c, /: fixing stupid mistake with putting code
before variable declaration ........ Merged revisions 313435 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r313432 | jrose | 2011-04-12 13:12:29 -0500 (Tue, 12 Apr 2011) |
14 lines reload Chan_dahdi memory leak caused by variables
chan_dahdi reloading with variables set via setvar in
chan_dahdi.conf would stay in the dahdi_pvt structs for
individual channels (causing them to just continue adding the new
ones to the list) and also there was a memory leak causes by the
conf objects. This patch resolves both of these by using
ast_variables_destroy during the loading process. (closes issue
#17450) Reported by: nahuelgreco Patches: patch.diff uploaded by
jrose (license 1225) Tested by: tilghman, jrose Review:
https://reviewboard.asterisk.org/r/1170/ ........ ........
* channels/chan_dahdi.c, /: Merged revisions 313432 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
........ r313432 | jrose | 2011-04-12 13:12:29 -0500 (Tue, 12 Apr
2011) | 14 lines reload Chan_dahdi memory leak caused by
variables chan_dahdi reloading with variables set via setvar in
chan_dahdi.conf would stay in the dahdi_pvt structs for
individual channels (causing them to just continue adding the new
ones to the list) and also there was a memory leak causes by the
conf objects. This patch resolves both of these by using
ast_variables_destroy during the loading process. (closes issue
#17450) Reported by: nahuelgreco Patches: patch.diff uploaded by
jrose (license 1225) Tested by: tilghman, jrose Review:
https://reviewboard.asterisk.org/r/1170/ ........
2011-04-11 23:08 +0000 [r313366-313369] Richard Mudgett <rmudgett@digium.com>
* apps/app_dial.c: Frames from the inbound channel should go to all
outbound channels in app_dial.c. In app_dial.c:wait_for_answer()
frames from the inbound channel should be sent to all outbound
channels instead of only if there is just one outbound channel.
Control frames like AST_CONTROL_CONNECTED_LINE need to be passed
to all of the the outbound channels. This can happen if a blond
transfer is done by a remote switch on the inbound channel. JIRA
AST-443 JIRA SWP-2730
* apps/app_dial.c: Backport a restructuring change from trunk to
make the next change stand out.
* main/cli.c: Added "Connected Line ID" and "Connected Line ID
Name" to "core show channel" output.
2011-04-11 19:36 +0000 [r313279] Leif Madsen <lmadsen@digium.com>
* /, configure, include/asterisk/autoconfig.h.in, configure.ac:
Merged revisions 313278 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r313278 | lmadsen | 2011-04-11 14:33:03 -0500
(Mon, 11 Apr 2011) | 14 lines Merged revisions 313277 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r313277 | lmadsen | 2011-04-11 14:30:20 -0500 (Mon, 11 Apr 2011)
| 6 lines Fix detection of OpenSSL 1.0 (closes issue #19093)
Reported by: tzafrir Patches: detect_openssl_10.diff uploaded by
tzafrir (license 46) ........ ................
2011-04-11 15:40 +0000 [r313190] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c, channels/sig_analog.c, /: Merged revisions
313189 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r313189 | rmudgett | 2011-04-11 10:32:53 -0500
(Mon, 11 Apr 2011) | 32 lines Merged revisions 313188 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r313188 | rmudgett | 2011-04-11 10:27:52 -0500 (Mon, 11 Apr 2011)
| 25 lines Stuck channel using FEATD_MF if caller hangs up at the
right time. The cause was actually a caller hanging up just at
the end of the Feature Group D DTMF tones that setup the call.
The reason for this is a "guard timer" that's implemented using
ast_safe_sleep(100). If the caller happens to hang up AFTER the
final tone of the DTMF string but BEFORE the end of that
ast_safe_sleep(), then ast_safe_sleep() will return non-zero.
This causes the code to bounce to the end of ss_thread(), but it
does NOT tear down the call properly. This should be a rare
occurrence because the caller has to hang up at EXACTLY the right
time. Nonetheless, it was happening quite regularly on the
reporter's system. It's not easily reproducible, unless you
purposely increase the guard-time to 2000 or more. Once you do
that, you can reproduce it every time by watching the DTMF debug
and hanging up just as it ends. Simply add an ast_hangup() before
goto quit. (closes issue #15671) Reported by: jcromes Patches:
issue15671.patch uploaded by pabelanger (license 224) Tested by:
jcromes ........ ................
2011-04-09 20:56 +0000 [r313142] Alexandr Anikin <may@telecom-service.ru>
* addons/chan_ooh323.c: fix trivial bug in ooh323_indicate on
AST_CONTROL_SRC... check p->rtp is not null
2011-04-07 13:35 +0000 [r313048] Jonathan Rose <jrose@digium.com>
* /, main/features.c: Merged revisions 313047 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r313047 | jrose | 2011-04-07 08:23:01 -0500 (Thu, 07 Apr 2011) |
9 lines Makes parking lots clear and rebuild properly when
features reload is invoked from CLI Before, default parkinglot in
context parkedcalls with ext 700 would always be present and when
reload was invoked, the previous parkinglots would not be
cleared. (closes issue #18801) Reported by: mickecarlsson Review:
https://reviewboard.asterisk.org/r/1161/ ........
2011-04-07 10:24 +0000 [r313001-313002] Alec L Davis <sivad.a@paradise.net.nz>
* apps/app_voicemail.c: app_voicemail: close_mailbox change
LOG_WARNING to LOG_NOTICE
* channels/sig_pri.c: Fix ISDN calling subaddr User Specified
Odd/Even Flag Calculation of the Odd/Even flag was wrong.
Implement correct algo, and set odd/even=0 if data would be
truncated. Only allow automatic calculation of the O/E flag,
don't let dialplan influence. (closes issue #19062) Reported by:
festr Patches: bug19062.diff2.txt uploaded by alecdavis (license
585) Tested by: festr, alecdavis, rmudgett
2011-04-05 18:45 +0000 [r312866-312949] Richard Mudgett <rmudgett@digium.com>
* channels/sig_pri.c, channels/sig_pri.h, channels/chan_dahdi.c:
Crash if ISDN span layer 1 is down on initial load. Regression
from -r312575 B channel shifting during negotiation. * Also
combine updating the alarm flag with clearing the resetting flag.
* channels/chan_sip.c: Add 416 response to OPTIONS packet. RFC3261
Section 11.2 says the response code to an OPTIONS packet needs to
be the same as if it were an INVITE.
* channels/chan_sip.c: Responding to OPTIONS packet with 404
because Asterisk not looking for "s" extension. The
get_destination() function was not using the "s" extension when
the request URI did not specify an extension. This is a
regression caused when the URI parsing code was extracted into
parse_uri(). Made get_destination() substitute the "s" extension
when the parsed URI results in an empty string. (closes issue
#18348) Reported by: shmaize Patches: issue18348_v1.8.patch
uploaded by rmudgett (license 664) Tested by: shmaize
2011-04-05 14:14 +0000 [r312766] Matthew Nicholson <mnicholson@digium.com>
* configs/manager.conf.sample, main/manager.c, /: Merged revisions
312764 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r312764 | mnicholson | 2011-04-05 09:13:07 -0500
(Tue, 05 Apr 2011) | 15 lines Merged revisions 312761 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r312761 | mnicholson | 2011-04-05 09:10:34 -0500 (Tue, 05 Apr
2011) | 8 lines Limit the number of unauthenticated manager
sessions and also limit the time they have to authenticate.
AST-2011-005 (closes issue #18996) Reported by: tzafrir Tested
by: mnicholson ........ ................
2011-04-05 14:13 +0000 [r312765] Jonathan Rose <jrose@digium.com>
* /, apps/app_meetme.c: Merged revisions 312762 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r312762 | jrose | 2011-04-05 09:11:36 -0500 (Tue, 05 Apr 2011) |
1 line Backporting trunk change to add verbosity to 'L' option in
meetme ........
2011-04-04 16:10 +0000 [r312575] Richard Mudgett <rmudgett@digium.com>
* channels/sig_pri.c, channels/sig_pri.h, channels/chan_dahdi.c, /:
Merged revisions 312574 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r312574 | rmudgett | 2011-04-04 11:00:02 -0500
(Mon, 04 Apr 2011) | 45 lines Merged revisions 312573 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r312573 | rmudgett | 2011-04-04 10:49:30 -0500 (Mon, 04 Apr 2011)
| 38 lines Issues with ISDN calls changing B channels during call
negotiations. The handling of the PROCEEDING message was not
using the correct call structure if the B channel was changed.
(The same for PROGRESS.) The call was also not hungup if the new
B channel is not provisioned or is busy. * Made all call
connection messages (SETUP_ACKNOWLEDGE, PROCEEDING, PROGRESS,
ALERTING, CONNECT, CONNECT_ACKNOWLEDGE) ensure that they are
using the correct structure and B channel. If there is any
problem with the operations then the call is now hungup with an
appropriate cause code. * Made miscellaneous messages
(INFORMATION, FACILITY, NOTIFY) find the correct structure by
looking for the call and not using the channel ID. NOTIFY is an
exception with versions of libpri before v1.4.11 because a call
pointer is not available for Asterisk to use. * Made all hangup
messages (DISCONNECT, RELEASE, RELEASE_COMPLETE) find the correct
structure by looking for the call and not using the channel ID.
(closes issue #18313) Reported by: destiny6628 Tested by:
rmudgett JIRA SWP-2620 (closes issue #18231) Reported by:
destiny6628 Tested by: rmudgett JIRA SWP-2924 (closes issue
#18488) Reported by: jpokorny JIRA SWP-2929 JIRA AST-437 (The
issues fixed here are most likely causing this JIRA issue.) JIRA
DAHDI-406 JIRA LIBPRI-33 (Stuck resetting flag likely fixed)
........ ................
2011-04-01 23:15 +0000 [r312461-312509] Richard Mudgett <rmudgett@digium.com>
* channels/chan_misdn.c: When a call going out an NT-PTMP port gets
rejected, Asterisk crashes. If a call is sent to an ISDN phone
that rejects the call with RELEASE_COMPLETE(cause: call
reject(21), or busy(17)) Asterisk crashes. I could not get my
setup to crash. However, I could see the possibility from a race
condition between queuing an AST_CONTROL_BUSY to the core and
then queueing an AST_CONTROL_HANGUP. If the AST_CONTROL_BUSY is
processed before the AST_CONTROL_HANGUP is queued, the
ast_channel could be destroyed out from under chan_misdn. Avoid
this particular crash scenario by not queueing the
AST_CONTROL_HANGUP if the AST_CONTROL_BUSY was queued. (closes
issue #18408) Reported by: wimpy Patches: issue18408_v1.8.patch
uploaded by rmudgett (license 664) Tested by: rmudgett, wimpy
JIRA SWP-2679
* main/ccss.c: CallCompletionRequest()/CallCompletionCancel() exit
non-zero if fail. The
CallCompletionRequest()/CallCompletionCancel() dialplan
applications exit nonzero on normal failure conditions. The
nonzero exit causes the dialplan to hangup immediately. The
dialplan author has no opportunity to report success/failure to
the user. * Made always return zero so the dialplan can continue.
* Made set CC_REQUEST_RESULT/CC_REQUEST_REASON and
CC_CANCEL_RESULT/CC_CANCEL_REASON channel variables respectively.
Also documented the values set. * Reduced the warning about no
core instance in CallCompletionCancel() to a debug message. It is
a normal event and should not be output at the WARNING level.
(closes issue #18763) Reported by: p_lindheimer Patches:
ccss.patch uploaded by p lindheimer (license 558) Modified Tested
by: p_lindheimer, rmudgett JIRA SWP-3042
2011-04-01 10:58 +0000 [r312286-312288] Tilghman Lesher <tilghman@meg.abyt.es>
* main/asterisk.c, include/asterisk/select.h, /: Merged revisions
312287 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r312287 | tilghman | 2011-04-01 05:51:24 -0500
(Fri, 01 Apr 2011) | 14 lines Merged revisions 312285 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r312285 | tilghman | 2011-04-01 05:36:42 -0500 (Fri, 01 Apr 2011)
| 7 lines Found some leaking file descriptors while looking at
ast_FD_SETSIZE dead code. (issue #18969) Reported by: oej
Patches: 20110315__issue18969__14.diff.txt uploaded by tilghman
(license 14) ........ ................
* addons/cdr_mysql.c: Reload must react correctly against a
possibly changed table, so dropping the conditional reload flag.
2011-04-01 09:03 +0000 [r312117-312211] Alec L Davis <sivad.a@paradise.net.nz>
* apps/app_voicemail.c, /: Merged revisions 312210 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r312210 | alecdavis | 2011-04-01 21:47:29 +1300
(Fri, 01 Apr 2011) | 29 lines Merged revisions 312174 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r312174 | alecdavis | 2011-04-01 21:29:49 +1300 (Fri, 01 Apr
2011) | 23 lines voicemail: get real last_message_index and
count_messages, ODBC resequence change last_message_index to read
the max msgnum stored in the database change count_messages to
actually count the number of messages. last_message_index change:
This fixed overwriting of the last message if msgnum=0 was
missing. Previously every incoming message would overwrite
msgnum=1. count_messages change: allows us to detect when
requencing is required in opneA_mailbox. resequence enabled for
ODBC storage: Assists with fixing up corrupt databases with gaps,
but only when a user actively opens there mailboxes. (closes
issue #18692,#18582,#19032) Reported by: elguero Patches: based
on odbc_resequence_mailbox2.1.diff uploaded by elguero (license
37) Tested by: elguero, nivek, alecdavis Review:
https://reviewboard.asterisk.org/r/1153/ ........
................
* apps/app_voicemail.c, /: Merged revisions 312103 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r312103 | alecdavis | 2011-04-01 20:25:54 +1300
(Fri, 01 Apr 2011) | 22 lines Merged revisions 312070 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r312070 | alecdavis | 2011-04-01 19:46:56 +1300 (Fri, 01 Apr
2011) | 16 lines app_voicemail: close_mailbox needs to respect
additional messages while mailbox is open. close_mailbox leave
gaps in message sequence if messages are deleted and new messages
arrive during this time, this is because the shuffle down to slot
0, only shuffles the number of pre-existing messages when mailbox
is opened, ignoring new arrivals. Fix: in close_mailbox
re-evaluate number of messages before the shuffle, this then
includes new arrivals. Happens on filebased or ODBC storage.
(issues #19032,#18582,#18692,#18998) Reported by:
alecdavis,tootai,afosorio Review:
https://reviewboard.asterisk.org/r/1153/ ........
................
2011-03-31 20:11 +0000 [r312022] Richard Mudgett <rmudgett@digium.com>
* channels/chan_misdn.c: chan_misdn segfaults when DEBUG_THREADS is
enabled. The segfault happens because jb->mutexjb is
uninitialized from the ast_malloc(). The internals of
ast_mutex_init() were assuming a nonzero value meant mutex
tracking initialization had already happened. Recent changes to
mutex tracking code to reduce excessive memory consumption
exposed this uninitialized value. Converted misdn_jb_init() to
use ast_calloc() instead of ast_malloc(). Also eliminated
redundant zero initialization code in the routine. (closes issue
#18975) Reported by: irroot
2011-03-31 06:43 +0000 [r311930] Tilghman Lesher <tilghman@meg.abyt.es>
* configs/cdr_mysql.conf.sample: Incorrect default example; the
field is actually internally named "clid", not "callerid".
(closes issue #19040) Reported by: wcselby Tested by: tilghman
2011-03-30 01:56 +0000 [r311874] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c: Update some setup_dahdi_int() comments.
2011-03-29 07:08 +0000 [r311799] Tilghman Lesher <tilghman@meg.abyt.es>
* cel/cel_odbc.c: Remove extraneous check from integer-type fields.
(closes issue #19027) Reported by: mlehner Review:
https://reviewboard.asterisk.org/r/1149/
2011-03-28 22:00 +0000 [r311751] Russell Bryant <russell@digium.com>
* apps/app_voicemail.c: Cross-reference VoiceMail() and
VoiceMailMain() in the xml docs.
2011-03-27 21:47 +0000 [r311687] Alexandr Anikin <may@telecom-service.ru>
* addons/chan_ooh323.c: correct return values in ooh323_indicate
for AST_CONTROL_T38_PARAMETERS
2011-03-23 21:54 +0000 [r311612-311615] Brett Bryant <bbryant@digium.com>
* apps/app_meetme.c: This patch fixes a bug with MeetMe behavior
where the 'P' option for always prompting for a pin is ignored
for the first caller. (closes issue #18070) Reported by: mav3rick
Review: https://reviewboard.asterisk.org/r/1132/
* channels/sip/reqresp_parser.c: Fix a possible crash in
sip/reqresp_parser.c that is caused by a possible null value.
(closes issue #18821) Reported by: cmaj Patches:
patch-reqresp_parser_sip_uri_domain_cmp_c_locale-crash-1.8.3-rc2.diff.tx
uploaded by cmaj (license 830)
2011-03-23 02:24 +0000 [r311558] Terry Wilson <twilson@digium.com>
* channels/sip/reqresp_parser.c: Don't use static declared buf in
parse_name_andor_addr This function isn't used anywhere yet, but
we definitely don't want to keep the same value for buf between
calls to the function.
2011-03-22 15:25 +0000 [r311497] David Vossel <dvossel@digium.com>
* /, apps/app_meetme.c: Merged revisions 311496 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r311496 | dvossel | 2011-03-22 10:24:45 -0500 (Tue, 22 Mar 2011)
| 2 lines Fixes memory leak in MeetMe AMI action ........
2011-03-18 16:19 +0000 [r311352] Jonathan Rose <jrose@digium.com>
* res/res_jabber.c, channels/chan_sip.c, res/res_fax.c: Changes
some print statements/events to use a blank string in place of
NULL if the string in question is NULL. This is supposed to
improve Solaris compatibility since Solaris goes berserk when
trying to output NULL strings. (closes issue #18759) Reported by:
bklang Patches: null-strings.patch uploaded by bklang (license
919)
2011-03-18 16:02 +0000 [r311342] Matthew Nicholson <mnicholson@digium.com>
* res/res_fax.c: Properly populate the LOCALSTATIONID channel
variable.
2011-03-18 02:59 +0000 [r311295-311297] Richard Mudgett <rmudgett@digium.com>
* channels/sig_pri.c: Race condition when ISDN
CallRerouting/CallDeflection invoked. The queued AST_CONTROL_BUSY
could sometimes be processed before the call_forward dial string
is recognized. * Moved setting the call_forwarding dial string
after sending a response to the initiator and just queue an empty
frame to wake up the media thread instead of an AST_CONTROL_BUSY.
* Added check for empty rerouting/deflection number and respond
with an error.
* apps/app_dial.c: Merged revision 310986 from
https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
.......... r310986 | rmudgett | 2011-03-16 13:56:28 -0500 (Wed,
16 Mar 2011) | 28 lines Dial() o option broke when connected line
feature added. The patch restores the o option behavior and adds
the ability to specify the CallerID. The Dial o and f options are
complementary to each other. The o option stores the CallerID on
the outgoing channel as the channel's CallerID. The f option
forces the CallerID sent by the outgoing channel. o(x) - The
argument 'x' is optional. If not present, then specify that the
CallerID that was present on the *calling* channel be stored as
the CallerID on the *called* channel. This was the behavior of
Asterisk 1.0 and earlier. If present, then specify the CallerID
stored on the *called* channel. Note that o(${CALLERID(all)}) is
similar to option o without parameters. f(x) - The argument 'x'
is optional and its presence changes the behavior of this option.
If not present, then force the outgoing CallerID on a
call-forward or deflection to the dialplan extension for this
Dial() using a dialplan 'hint'. For example, some PSTNs do not
allow CallerID to be set to anything other than the numbers
assigned to you. If present, then force the outgoing CallerID to
'x'. Patches: jira_abe_2752_dial_fo_options.patch uploaded by
rmudgett (license 664) Tested by: rmudgett JIRA ABE-2752 JIRA
SWP-3096 ..........
2011-03-17 19:03 +0000 [r311197] Jonathan Rose <jrose@digium.com>
* apps/app_chanspy.c: This fixes a nasty chanspy bug which was
causing a channel leak every time a spied on channel made a call.
In addition to the above, it makes certain channel destruction
occurs so that applications don't get stuck waiting for datastore
destruction while monitored by chanspy. (closes issue #18742)
Reported by: jkister Tested by: jkister, jcovert, jrose Review:
http://reviewboard.digium.internal/r/106/
2011-03-17 15:00 +0000 [r311141] Matthew Nicholson <mnicholson@digium.com>
* main/manager.c, /: Merged revisions 311140 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r311140 | mnicholson | 2011-03-17 09:58:52 -0500 (Thu, 17 Mar
2011) | 4 lines Don't write items to the manager socket twice.
AST-2011-003 (closes issue 0018987) Reported by: ks-steven
........
2011-03-17 10:49 +0000 [r311050] Alec L Davis <sivad.a@paradise.net.nz>
* /, configs/indications.conf.sample: Merged revisions 311049 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r311049 | alecdavis | 2011-03-17 23:45:47 +1300
(Thu, 17 Mar 2011) | 17 lines Merged revisions 311048 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r311048 | alecdavis | 2011-03-17 23:43:35 +1300 (Thu, 17 Mar
2011) | 12 lines Remove extra quote in indications.conf Picking
low hanging fruit. (closes issue #18971) Reported by: IgorG
Patches: based on indications.conf.sample.diff uploaded by IgorG
(license 20) Tested by: IgorG ........ ................
2011-03-16 19:47 +0000 [r310902-310999] Terry Wilson <twilson@digium.com>
* main/tcptls.c, /: Merged revisions 310998 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r310998 | twilson | 2011-03-16 14:46:36 -0500 (Wed, 16 Mar 2011)
| 11 lines Fix crash on fdopen failure See security advisory
AST-2011-004 (closes issue #18845) Reported by: cmaj Patches:
patch-main-tcptls-1.8.3-rc2-open-session-crash-take2.diff.txt
uploaded by cmaj (license 830)
patch-main-tcptls-1.8.3-rc2-open-session-crash-take3.diff.txt
uploaded by cmaj (license 830) Tested by: cmaj, twilson ........
* main/manager.c, /: Merged revisions 310992 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r310992 | twilson | 2011-03-16 14:23:03 -0500 (Wed, 16 Mar 2011)
| 4 lines Don't keep trying to write to a closed connection See
security advisory AST-2011-003. ........
* /, main/features.c: Merged revisions 310889 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r310889 | twilson | 2011-03-16 12:03:27 -0500
(Wed, 16 Mar 2011) | 36 lines Merged revisions 310888 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r310888 | twilson | 2011-03-16 11:58:42 -0500 (Wed, 16 Mar 2011)
| 29 lines Don't delay DTMF in core bridge while listening for
DTMF features This patch is mostly the work of Olle Johansson. I
did some cleanup and added the silence generating code if
transmit_silence is set. When a channel listens for DTMF in the
core bridge, the outbound DTMF is not sent until we have received
DTMF_END. For a long DTMF, this is a disaster. We send 4 seconds
of DTMF to Asterisk, which sends no audio for those 4 seconds.
Some products see this delay and the time skew on RTP packets
that results and start ignoring the audio that is sent afterward.
With this change, the DTMF_BEGIN frame is inspected and checked.
If it matches a feature code, we wait for DTMF_END and activate
the feature as before. If transmit_silence=yes in asterisk.conf,
silence is sent if we paritally match a multi-digit feature. If
it doesn't match a feature, the frame is forwarded along with the
DTMF_END without delay. By doing it this way, DTMF is not
delayed. (closes issue #15642) Reported by: jasonshugart Patches:
issue_15652_dtmf_ast-1.4.patch.txt uploaded by twilson (license
396) Tested by: globalnetinc, jde (closes issue #16625) Reported
by: sharvanek Review: https://reviewboard.asterisk.org/r/1092/
Review: https://reviewboard.asterisk.org/r/1125/ ........
................
2011-03-15 01:48 +0000 [r310834] Tilghman Lesher <tilghman@meg.abyt.es>
* addons/chan_ooh323.c: Fix branch compile.
2011-03-15 01:00 +0000 [r310781] Alec L Davis <sivad.a@paradise.net.nz>
* main/utils.c: core show locks: display ThreadID in hexadecimal
Allow easier cross referencing of thread ID's with GDB backtraces
(closes issue #18968) Reported by: alecdavis Patches:
bug18968.diff.txt uploaded by alecdavis (license 585)
2011-03-14 21:45 +0000 [r310734] Alexandr Anikin <may@telecom-service.ru>
* addons/chan_ooh323.c, addons/ooh323c/src/ooCapability.c,
addons/ooh323c/src/ooCalls.h: Introduce t.38 parameters control
functionality not full but enough for Send/RcvFax support
Introduce t.38 controls between asterisk core and channel/proto
layers. Not all parameters are transferred from proto layers but
*Fax apps tested and work ok. (issue #18693) Reported by:
benngard2 Patches: issue-18693.patch uploaded by may213 (license
454)
2011-03-14 21:30 +0000 [r310726-310733] Jonathan Rose <jrose@digium.com>
* main/channel.c: Undoes 310726 for further analysis
* main/channel.c: Moves data store destruction from channel
destruction to hangup in channel.c This moves the data store
destruction and app signaling events for a call to ast_hangup so
that threads which wait for data store destruction don't become
stuck forever when attached to an application/function/etc that
keeps the channel open. (closes issue #18742) Reported by:
jkister Patches: patch.diff uploaded by jrose (license 1225)
Tested by: jkister, jcovert, jrose Review:
https://reviewboard.asterisk.org/r/1136/
2011-03-14 16:50 +0000 [r310636] Richard Mudgett <rmudgett@digium.com>
* /, main/callerid.c: Merged revisions 310635 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r310635 | rmudgett | 2011-03-14 11:47:54 -0500
(Mon, 14 Mar 2011) | 32 lines Merged revisions 310633 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r310633 | rmudgett | 2011-03-14 11:38:24 -0500 (Mon, 14 Mar 2011)
| 25 lines "Caller*ID failed checksum" on Wildcard TDM2400P and
TDM410 The last character in the caller id message is getting a
framing error. The checksum is the last character in the message.
A framing error in the checksum could be because: 1) The sender
did not send a full stop bit. 2) The sender cut off the FSK
carrier too soon. 3) The sender opted to send zero of the
specified zero to 10 trailing mark bits and round-off errors in
the code resulted in the code not being where it thought it was
in the demodulated bit stream. Bit 8 of 'b' is set when parity
error. Bit 9 of 'b' is set when framing error. Made ignore the
framing and parity error bits if the errored character is the
checksum. We can tolerate a framing/parity error there. The
checksum character validates the message. (closes issue #18474)
Reported by: nivek Patches: callerid.c.1.patch uploaded by nivek
(license 636) (with modifications) Tested by: nivek ........
................
2011-03-14 15:27 +0000 [r310587] Jonathan Rose <jrose@digium.com>
* /, funcs/func_volume.c: Merged revisions 310585 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r310585 | jrose | 2011-03-14 08:56:22 -0500 (Mon, 14 Mar 2011) |
8 lines Adds 'p' as an option to func_volume. When it is on, the
old behavior with DTMF controlling volume adjustment will be
enforced. When it is off, DTMF will not be processed by the
function. Programmed by Jonathan Rose Reviewed by David Vossel,
Leif Madsen, and Russell Bryant
http://reviewboard.digium.internal/r/93/ ........
2011-03-12 20:27 +0000 [r310415-310462] Tilghman Lesher <tilghman@meg.abyt.es>
* /, pbx/pbx_ael.c: Merged revisions 310448 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r310448 | tilghman | 2011-03-12 14:24:54 -0600
(Sat, 12 Mar 2011) | 38 lines Recorded merge of revisions 310435
via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r310435 | tilghman | 2011-03-12 14:22:07 -0600 (Sat, 12 Mar 2011)
| 31 lines Add AELSub, which provides a stable entry point into
AEL subroutines. This commit needs some explanation, given that
we're adding a new application into an existing release branch.
This is generally a violation of our release policy, except in
very limited circumstances, and I believe this is one of those
circumstances. The problem that this solves is one of the sanity
of using multiple dialplan languages to define a dialplan. In the
case of the reporter, he or she is using AEL is define
subroutines, while using Realtime extensions to invoke those
subroutines. While you can do this, it's based upon the reality
of AEL using actual dialplan extensions; however, there is no
guarantee that the details of _how_ AEL is compiled into
extensions will remain stable. In fact, at the time of this
commit, it has already changed twice, once in a fundamental way.
Now normally, a new application would only be added to trunk.
However, this application is explicitly to create a stable
user-level API between versions, and adding it to trunk only will
not solve the user's problem of switching between 1.6.2 and 1.8,
nor will it help anybody switching from 1.8 to 1.10. Therefore,
it needs to go into existing release branches. For the sake of
consistency, and also because one of the changes was between 1.4
and 1.6.x, I am also electing to commit this to 1.4. (closes
issue #18910) Reported by: alexandrekeller Patches:
20110304__issue18919__1.6.2.diff.txt uploaded by tilghman
(license 14) 20110304__issue18919__1.4.diff.txt uploaded by
tilghman (license 14) Tested by: alexandrekeller ........
................
* /, funcs/func_odbc.c: Merged revisions 310414 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r310414 | tilghman | 2011-03-12 13:51:23 -0600 (Sat, 12 Mar 2011)
| 7 lines Transactional handles should be used for the insertbuf,
if available. Also, fix a possible resource leak. (closes issue
#18943) Reported by: irroot ........
2011-03-11 06:47 +0000 [r310287] Alec L Davis <sivad.a@paradise.net.nz>
* main/rtp_engine.c: remote_bridge_loop: prevent segfault when
after transfer of IAX2 of DAHDI call If the channel condition is
one of the following after breaking out of the loop, don't try to
update_peer (where x = 0/1) 1). ZOMBIE 2). cx->tech_pvt != pvtx
3). gluex != ast_rtp_instance_get_glue(cx->tech->type)) (closes
issue #18781) Reported by: alecdavis Patches: bug18781.diff3.txt
uploaded by alecdavis (license 585) Tested by: alecdavis, ZX81
Review: https://reviewboard.asterisk.org/r/1128/
2011-03-10 16:05 +0000 [r310240] Terry Wilson <twilson@digium.com>
* main/manager.c, res/res_phoneprov.c: Add \r\n to remaining http
headers passed to ast_http_send r309204 changed the behavior of
ast_http_send. It now requires headers to be passed with trailing
\r\n. This change updates the remaining instances in the code
that did not pass the \r\n. (closes issue #18186) Reported by:
nivaldomjunior Patches: res_phoneprov.c.diff uploaded by lathama
(license 1028) manager.diff.txt uploaded by twilson (license 396)
Tested by: lathama
2011-03-10 15:17 +0000 [r310231] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Be more tolerant of what URI we accept for
call completion PUBLISH requests. (closes issue #18946) Reported
by: GeorgeKonopacki Patches: 18946.patch uploaded by mmichelson
(license 60) Tested by: GeorgeKonopacki
2011-03-10 05:53 +0000 [r310142] Tilghman Lesher <tilghman@meg.abyt.es>
* apps/app_voicemail.c, res/res_config_odbc.c, /,
funcs/func_odbc.c: Merged revisions 310141 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r310141 | tilghman | 2011-03-09 23:51:37 -0600
(Wed, 09 Mar 2011) | 12 lines Merged revisions 310140 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r310140 | tilghman | 2011-03-09 23:38:44 -0600 (Wed, 09 Mar 2011)
| 5 lines Initialize column size to 0 to deal with a potential
UnixODBC bug on 64-bit systems. (closes issue #18295) Reported
by: pruiz ........ ................
2011-03-08 20:19 +0000 [r310088] Jonathan Rose <jrose@digium.com>
* channels/sip/dialplan_functions.c: Returns with an error notice
if CHANNEL function of SIP channel is read without arguments.
(Closes issue #18653) Reported by: wuwu Patches: diff.patch
uploaded by jrose (license 1225) Tested by: jrose
2011-03-08 18:10 +0000 [r310039] Terry Wilson <twilson@digium.com>
* res/res_calendar.c: Spelling fix in "calendar show calendar"
s/Cartegories/Catagories/ (closes issue #18931) Reported by:
pdugas Patches: res_calendar.c.patch uploaded by pdugas (license
1222)
2011-03-08 16:37 +0000 [r309994] Richard Mudgett <rmudgett@digium.com>
* channels/sig_pri.c: Make pri parameter description consistent.
2011-03-07 22:07 +0000 [r309858] Jonathan Rose <jrose@digium.com>
* apps/app_mixmonitor.c, /: Merged revisions 309857 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r309857 | jrose | 2011-03-07 16:04:44 -0600
(Mon, 07 Mar 2011) | 15 lines Merged revisions 309856 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r309856 | jrose | 2011-03-07 16:02:12 -0600 (Mon, 07 Mar 2011) |
8 lines Bug fix for MixMonitor involving filenames with '.' not
in the extension Closes issue #18391) Reported by: pabelanger
Patches: bugfix.patch uploaded by jrose (license 1225) Tested by:
jrose ........ ................
2011-03-07 00:54 +0000 [r309808] Tilghman Lesher <tilghman@meg.abyt.es>
* main/ast_expr2.fl, channels/chan_dahdi.c, /, configure,
include/asterisk/autoconfig.h.in, main/ast_expr2f.c,
configure.ac: Merged revisions 309251 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r309251 | tilghman | 2011-03-01 19:06:02 -0600 (Tue, 01 Mar 2011)
| 7 lines Revert previous 2 commits, and instead conditionally
redefine the same macro used in flex 2.5.35 that clashed with our
workaround. Not surprisingly, the workaround was exactly the same
code as was provided by the Flex maintainers, albeit in two
different places, in different macros. This should fix the
FreeBSD builds, which have an older version of Flex. ........
2011-03-07 00:13 +0000 [r309765] Mark Michelson <mmichelson@digium.com>
* configs/sip.conf.sample: Indicate that Asterisk uses the Allow
header to determine if MESSAGE requests should be sent.
2011-03-05 17:44 +0000 [r309720] Moises Silva <moises.silva@gmail.com>
* channels/chan_dahdi.c: Fix caller id passed to
openr2_chan_make_call (closes issue #18894) Reported by: malufrj
Tested by: moy
2011-03-05 10:29 +0000 [r309678] Tilghman Lesher <tilghman@meg.abyt.es>
* main/asterisk.c, /: Merged revisions 309677 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r309677 | tilghman | 2011-03-05 04:28:24 -0600 (Sat, 05 Mar 2011)
| 7 lines Missed part of the conversion when we started passing
ppid to astcanary. (closes issue #18850) Reported by: viraptor
Patches: canary_ppid.patch uploaded by viraptor (license 543)
........
2011-03-04 19:38 +0000 [r309448-309585] Matthew Nicholson <mnicholson@digium.com>
* /, pbx/pbx_lua.c: Merged revisions 309584 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r309584 | mnicholson | 2011-03-04 13:37:13 -0600 (Fri, 04 Mar
2011) | 2 lines Restore mysterious lua_pushvalue() call removed
in r309494. The mystery has been solved. ........
* /, pbx/pbx_lua.c: Merged revisions 309541 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r309541 | mnicholson | 2011-03-04 12:59:20 -0600 (Fri, 04 Mar
2011) | 4 lines Check for errors from fseek() when loading config
file, properly abort on errors from fread(), and supply a
traceback for errors generated when loading the config file.
Also, prepend a newline to traceback output so that the main
error message is on it's own line. ........
* /, pbx/pbx_lua.c: Merged revisions 309494 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r309494 | mnicholson | 2011-03-04 11:55:57 -0600 (Fri, 04 Mar
2011) | 2 lines remove mysterious lua_pushvalue() that is never
used ........
* pbx/pbx_lua.c: Export global symbols from pbx_lua to allow
modules to be loaded. Fixes a regression introduced in r278132.
(closes issue #18671) Reported by: Igels Patches:
pbx_lua_global_symbols1.diff uploaded by mnicholson (license 96)
Tested by: Igels
2011-03-04 15:22 +0000 [r309445] Richard Mudgett <rmudgett@digium.com>
* UPGRADE.txt, channels/sig_pri.c, channels/sig_pri.h,
channels/chan_dahdi.c, funcs/func_channel.c: Get real channel of
a DAHDI call. Starting with Asterisk v1.8, the DAHDI channel name
format was changed for ISDN calls to:
DAHDI/i<span>/<number>[:<subaddress>]-<sequence-number> There
were several reasons that the channel name had to change. 1) Call
completion requires a device state for ISDN phones. The generic
device state uses the channel name. 2) Calls do not necessarily
have B channels. Calls placed on hold by an ISDN phone do not
have B channels. 3) The B channel a call initially requests may
not be the B channel the call ultimately uses. Changes to the
internal implementation of the Asterisk master channel list
caused deadlock problems for chan_dahdi if it needed to change
the channel name. Chan_dahdi no longer changes the channel name.
4) DTMF attended transfers now work with ISDN phones because the
channel name is "dialable" like the chan_sip channel names. For
various reasons, some people need to know which B channel a DAHDI
call is using. * Added CHANNEL(dahdi_span),
CHANNEL(dahdi_channel), and CHANNEL(dahdi_type) so the dialplan
can determine the B channel currently in use by the channel. Use
CHANNEL(no_media_path) to determine if the channel even has a B
channel. * Added AMI event DAHDIChannel to associate a DAHDI
channel with an Asterisk channel so AMI applications can
passively determine the B channel currently in use. Calls with
"no-media" as the DAHDIChannel do not have an associated B
channel. No-media calls are either on hold or call-waiting.
(closes issue #17683) Reported by: mrwho Tested by: rmudgett
(closes issue #18603) Reported by: arjankroon Patches:
issue17683_18603_v1.8_v2.patch uploaded by rmudgett (license 664)
Tested by: stever28, rmudgett
2011-03-04 01:50 +0000 [r309403] David Ruggles <thedavidfactor@gmail.com>
* apps/app_externalivr.c, /: Merged revisions 309356 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r309356 | diruggles | 2011-03-03 19:42:28 -0500
(Thu, 03 Mar 2011) | 16 lines Merged revisions 309355 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r309355 | diruggles | 2011-03-03 19:34:13 -0500 (Thu, 03 Mar
2011) | 9 lines fix small memory leak fix small memory leak
caused by a string allocation that wasn't freed (closes issue
#18907) Reported by: andy11 Patches:
asterisk_trunk-app_externalivr-leak.patch uploaded by andy11
(license 1224) ........ ................
2011-03-02 19:54 +0000 [r309204-309256] Jason Parker <jparker@digium.com>
* /, channels/chan_sip.c: Merged revisions 309255 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r309255 | qwell | 2011-03-02 13:53:47 -0600 (Wed, 02 Mar 2011) |
8 lines Fix usage of "hasvoicemail=yes" and "mailbox=" in
users.conf for SIP. Since it's a duplicate, nothing is going to
be done, so delme doesn't need to be set at all. Strangely, when
this was added, this was being set to 1 in 1.6, and 0 in trunk.
(issue AST-439) ........
* main/http.c: Fix consistency of CRLFs on HTTP headers that get
sent out. (closes issue #18186) Reported by: nivaldomjunior
Patches: 18186-httpheadernewline.diff uploaded by qwell (license
4)
2011-03-01 21:57 +0000 [r309126-309170] Richard Mudgett <rmudgett@digium.com>
* funcs/func_channel.c: Document CHANNEL(keypad_digits) and
CHANNEL(no_media_path). * Added XML documentation for
CHANNEL(keypad_digits) and CHANNEL(no_media_path). * Tweaked XML
documentation for CHANNEL(reversecharge).
* channels/sig_analog.c: Chan_dahdi does not retain CID when
detecting DTMF CID without polarity reversal. Looks like an
unintended change when sig_analog.c was extracted from
chan_dahdi.c. Removed useless conditional around needed code and
fixed resulting compiler warning. (closes issue #18667) Reported
by: enegaard Patches: issue18667.patch uploaded by enegaard
(license 1197) Tested by: enegaard JIRA SWP-2965
2011-03-01 16:09 +0000 [r309084] David Vossel <dvossel@digium.com>
* /, channels/chan_sip.c: Merged revisions 309083 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r309083 | dvossel | 2011-03-01 10:05:25 -0600 (Tue, 01 Mar 2011)
| 9 lines Fixes thread blocking issue in the sip TCP/TLS
implementation. (closes issue #18497) Reported by: vois Patches:
issues_18497.diff uploaded by dvossel (license 671) Tested by:
vois, rossbeer, kowalma, Freddi_Fonet ........
2011-02-28 11:10 +0000 [r308991-309035] Tilghman Lesher <tilghman@meg.abyt.es>
* main/ast_expr2.fl, /, configure,
include/asterisk/autoconfig.h.in, main/ast_expr2f.c,
configure.ac: Merged revisions 309033-309034 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r309033 | tilghman | 2011-02-28 04:43:12 -0600 (Mon, 28 Feb 2011)
| 4 lines A later version of flex already includes the fwrite
workaround code, which if used twice causes a compilation error.
Detect whether Flex will compile without the workaround; if so,
suppress our workaround code. ........ r309034 | tilghman |
2011-02-28 05:07:52 -0600 (Mon, 28 Feb 2011) | 2 lines Clarify
meaning, removing double negative (stupid!) ........
* /, funcs/func_odbc.c: Merged revisions 308990 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r308990 | tilghman | 2011-02-28 03:32:22 -0600 (Mon, 28 Feb 2011)
| 7 lines Statements updating zero rows may return SQL_NO_DATA.
This is fine; it's handled. (closes issue #18815) Reported by:
irroot Patches: func_odbc.insert_nodata.patch uploaded by irroot
(license 52) ........
2011-02-25 18:52 +0000 [r308945] Alec L Davis <sivad.a@paradise.net.nz>
* channels/chan_sip.c: Fix Deadlock with attended transfer of SIP
call Call path sip_set_rtp_peer (locks chan then pvt)
transmit_reinvite_with_sdp try_suggested_sip_codec
pbx_builtin_getvar_helper (locks p->owner) But by the time
p->owner lock was attempted, seems as though chan and p->owner
were different. So in sip_set_rtp_peer, lock pvt first then lock
p->owner using deadlocking methods. (closes issue #18837)
Reported by: alecdavis Patches: bug18837-trunk.diff3.txt uploaded
by alecdavis (license 585) Tested by: alecdavis, Irontec, ZX81,
cmaj Review: [https://reviewboard.asterisk.org/r/1126/]
2011-02-24 21:38 +0000 [r308903] Richard Mudgett <rmudgett@digium.com>
* main/channel.c: Invalid read in ast_channel_set_caller_event().
Valgrind reported that ast_channel_set_caller_event() was reading
data from a freed buffer when using the pre_set structure.
Rearange things to pre-calculate the name and number pointer
before updating the caller party structure to see if the name or
number was changed.
2011-02-24 17:57 +0000 [r308815] Terry Wilson <twilson@digium.com>
* main/manager.c, /: Merged revisions 308814 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r308814 | twilson | 2011-02-24 11:54:49 -0600
(Thu, 24 Feb 2011) | 19 lines Merged revisions 308813 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r308813 | twilson | 2011-02-24 11:42:16 -0600 (Thu, 24 Feb 2011)
| 12 lines Don't broadcast FullyBooted to every AMI connection
The FullyBooted event should not be sent to every AMI connection
every time someone connects via AMI. It should only be sent to
the user who just connected. (closes issue #18168) Reported by:
FeyFre Patches: bug0018168.patch uploaded by FeyFre (license
1142) Tested by: FeyFre, twilson ........ ................
2011-02-24 15:06 +0000 [r308723] Matthew Nicholson <mnicholson@digium.com>
* main/udptl.c, /: Merged revisions 308722 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r308722 | mnicholson | 2011-02-24 08:59:41 -0600
(Thu, 24 Feb 2011) | 9 lines Merged revisions 308721 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r308721 | mnicholson | 2011-02-24 08:54:56 -0600 (Thu,
24 Feb 2011) | 2 lines silence gcc 4.2 compiler warning ........
................
2011-02-24 03:41 +0000 [r308679] Terry Wilson <twilson@digium.com>
* configs/sip.conf.sample, /, channels/chan_sip.c: Merged revisions
308678 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r308678 | twilson | 2011-02-23 21:38:22 -0600 (Wed, 23 Feb 2011)
| 8 lines Use remotesecret to authenticate with a remote party
The remotesecret option was only being used for outbound
registration and not for placing calls. This patch uses
remotesecret on outbound calls if it is set, otherwise secret is
still used. Review: https://reviewboard.asterisk.org/r/1107/
........
2011-05-09 Leif Madsen <lmadsen@digium.com>
* Asteris 1.8.4 Released.
2011-04-25 Leif Madsen <lmadsen@digium.com>
* Asterisk 1.8.4-rc3 Released.
* Use SSLv23_client_method instead of old SSLv2 only.
(closes issue 0019095)
(closes issue 0019138)
Reported by: tzafrir
Patches:
no_ssl2.diff uploaded by tzafrir (license 46)
Tested by: russell, chazzam
* Resolve crash in ast_mutex_init()
2011-02-25 Leif Madsen <lmadsen@digium.com>
* Asterisk 1.8.4-rc2 Released.
* Fix Deadlock with attended transfer of SIP call
(Closes issue #18837. Reported, patched by alecdavis. Tested by
alecdavid, Irontec, ZX81, cmaj)
2011-02-23 Leif Madsen <lmadsen@digium.com>
* Asterisk 1.8.4-rc1 Released.
2011-02-23 23:38 +0000 [r308622] Richard Mudgett <rmudgett@digium.com>
* channels/sig_pri.c: sig_pri_new_ast_channel() should return NULL
when new_ast_channel() fails. (closes issue #18874) Reported by:
cmaj Patches:
patch-sig_pri-crash-possible-null-channel-pointer.diff.txt
uploaded by cmaj (license 830) JIRA SWP-3172
2011-02-22 15:31 +0000 [r308526] Andrew Latham <lathama@gmail.com>
* main/http.c: Use ast_debug for console logging Guessed the log
levels based on info that level 3 is the soft roof. Can we create
a page / document to define the levels?
2011-02-21 15:02 +0000 [r308416] Matthew Nicholson <mnicholson@digium.com>
* main/udptl.c, /: Merged revisions 308414 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r308414 | mnicholson | 2011-02-21 09:00:22 -0600
(Mon, 21 Feb 2011) | 12 lines Merged revisions 308413 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r308413 | mnicholson | 2011-02-21 08:57:15 -0600 (Mon, 21 Feb
2011) | 5 lines Properly check the bounds of arrays when decoding
UDPTL packets. Also, remove broken support for receiving UDPTL
packets larger than 16k. That shouldn't ever happen anyway.
AST-2011-002 FAX-281 ........ ................
2011-02-21 14:24 +0000 [r308393] Andrew Latham <lathama@gmail.com>
* main/http.c: Add HTTP URI Debug logging and update notice enable
reporting of the request URI / URL in debugging change funny
debug note to a serious note.
2011-02-19 14:06 +0000 [r308330] Andrew Latham <lathama@gmail.com>
* main/http.c: Add CSS MIME Type Modern browsers are checking for
the MIME Type of pages and in some cases will not load a file if
the type is wrong.
2011-02-19 11:02 +0000 [r308288] Tilghman Lesher <tilghman@meg.abyt.es>
* utils: A few more (copies of) files to ignore in this directory.
2011-02-18 00:07 +0000 [r308242] Alexandr Anikin <may@telecom-service.ru>
* addons/ooh323cDriver.c, addons/ooh323cDriver.h,
addons/chan_ooh323.c: added g729onlyA option for announce only
AnnexA g.729 codec in h.323 capabilities. Option can be global or
per user/peer.
2011-02-16 20:21 +0000 [r308150] Paul Belanger <pabelanger@digium.com>
* addons/ooh323c/src/ooSocket.c: Fix FreeBSD builds.
2011-02-16 07:57 +0000 [r308098] Alexandr Anikin <may@telecom-service.ru>
* addons/ooh323c/src/ooSocket.c: ifdef __linux__ keepalive
variables also
2011-02-15 23:34 +0000 [r308010] Jason Parker <jparker@digium.com>
* apps/app_queue.c, /: Merged revisions 308007 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r308007 | qwell | 2011-02-15 17:33:24 -0600
(Tue, 15 Feb 2011) | 17 lines Merged revisions 308002 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r308002 | qwell | 2011-02-15 17:32:20 -0600 (Tue, 15 Feb 2011) |
10 lines Fix regression that changed behavior of queues when
ringing a queue member. This reverts r298596, which was to fix a
highly bizarre and contrived issue with a queue member that
called into his own queue being transferred back into his own
queue. I couldn't reproduce that issue in any way. I think one of
the other recent transfer fixes actually fixed this. (closes
issue #18747) Reported by: vrban ........ ................
2011-02-15 23:08 +0000 [r307970] Alexandr Anikin <may@telecom-service.ru>
* addons/ooh323c/src/ooSocket.c: include tcp keepalive socket calls
only on linux, freebsd and others don't have these options on
sockets.
2011-02-15 19:52 +0000 [r307879-307962] Richard Mudgett <rmudgett@digium.com>
* apps/app_dial.c: Don't crash when forcing caller id.
* channels/sig_pri.c, include/asterisk/ccss.h, channels/sig_pri.h,
channels/chan_dahdi.c, channels/chan_sip.c, main/ccss.c: No
response sent for SIP CC subscribe/resubscribe request. Asterisk
does not send a response if we try to subscribe for call
completion after we have received a 180 Ringing. You can only
subscribe for call completion when the call has been cleared.
When we receive the 180 Ringing, for this call, its
call-completion state is 'CC_AVAILABLE'. If we then send a
subscribe message to Asterisk, it trys to change the
call-completion state to 'CC_CALLER_REQUESTED'. Because this is
an invalid state change, it just ignores the message. The only
state Asterisk will accept our subscribe message is in the
'CC_CALLER_OFFERED' state. Asterisk will go into the
'CC_CALLER_OFFERED' when the SIP client clears the call by
sending a CANCEL. Asterisk should always send a response. Even if
its a negative one. The fix is to allow for the CCSS core to
notify a CC agent that a failure has occurred when CC is
requested. The "ack" callback is replaced with a "respond"
callback. The "respond" callback has a parameter indicating
either a successful response or a specific type of failure that
may need to be communicated to the requester. (closes issue
#18336) Reported by: GeorgeKonopacki Tested by: mmichelson,
rmudgett JIRA SWP-2633 (closes issue #18337) Reported by:
GeorgeKonopacki Tested by: mmichelson JIRA SWP-2634
2011-02-15 07:02 +0000 [r307750-307837] Tilghman Lesher <tilghman@meg.abyt.es>
* /, funcs/func_odbc.c: Merged revisions 307836 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r307836 | tilghman | 2011-02-15 01:01:37 -0600 (Tue, 15 Feb 2011)
| 8 lines Need to retrieve the rows affected before using the
associated variable. (closes issue #18795) Reported by: irroot
Patches: 20110211__issue18795.diff.txt uploaded by tilghman
(license 14) Tested by: tilghman ........
* res/res_odbc.c, /: Merged revisions 307792 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r307792 | tilghman | 2011-02-14 14:10:28 -0600 (Mon, 14 Feb 2011)
| 8 lines Increment usage count at first reference, to avoid a
race condition with many threads creating connections all at
once. (issue #18156) Reported by: asgaroth Patches:
20110214__issue18156.diff.txt uploaded by tilghman (license 14)
Tested by: tilghman ........
* apps/app_queue.c, apps/app_dial.c: Calling a gosub routine
defined in AEL from Dial/Queue ceased to work. A bug in AEL did
not distinguish between the "s" extension generated by AEL and an
"s" extension that was required to exist by the chan_dahdi (or
another channel) that was not supplied with a starting extension.
Therefore, AEL made incorrect assumptions about what commands
were permissable in the context. This was fixed by making AEL
generate a different extension name. However, Dial and Queue make
additional assumptions about the name of the default gosub
extension. Therefore, they needed to be brought into line with a
"macro" rendered by AEL (as a gosub), without breaking
traditional dialplans written without the aid of AEL. Related to
(issue #18480) Reported by: nivek (closes issue #18729) Reported
by: kkm Patches: 20110209__issue18729.diff.txt uploaded by
tilghman (license 14) 018729-dial-queue-gosub-try3.patch uploaded
by kkm (license 888) Tested by: kkm
2011-02-10 22:39 +0000 [r307536] Jason Parker <jparker@digium.com>
* main/asterisk.c, contrib/init.d/rc.debian.asterisk, /: Merged
revisions 307535 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r307535 | qwell | 2011-02-10 16:35:49 -0600
(Thu, 10 Feb 2011) | 15 lines Merged revisions 307534 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r307534 | qwell | 2011-02-10 16:33:09 -0600 (Thu, 10 Feb 2011) |
8 lines Remove color when executing commands via a remote
console. Essentially this makes '-x' imply '-n' on rasterisk.
This was done in a different and incomplete way previously, which
I'm reverting here. (issue #18776) Reported by: alecdavis
........ ................
2011-02-10 18:50 +0000 [r307509] Alexandr Anikin <may@telecom-service.ru>
* addons/ooh323c/src/ooSocket.c, addons/ooh323c/src/oochannels.c,
addons/ooh323c/src/ooStackCmds.c, addons/ooh323c/src/ooq931.c,
addons/ooh323c/src/ooh323.c, addons/chan_ooh323.c,
addons/ooh323c/src/ooh245.c, addons/ooh323c/src/ooCalls.h:
Corrections for properly work with H.323v2 (older) endpoints and
other small fixes. Interpret remote side H.225 version.
Corrections for H.323v2 endpoints: don't start TCS and MSD before
connect, don't start TCS and MSD by accepting H.245 connection,
start TCS and MSD by StartH245 facility message. Other fixes: fix
non zeroended remoteDisplayName issue, small fixes in call
clearing by closing H.245 connection, tcp keepalive introduced on
TCP connections (now is hardcoded, will be configurable in the
future), don't force H.245tunneling if FastStart is active, don't
send Alerting singal more than once per call. (issue 0018542)
Reported by: vmikhelson Patches: issue18542-final-3.patch
uploaded by may213 (license 454) Tested by: vmikhelson
2011-02-10 17:44 +0000 [r307467] Mark Michelson <mmichelson@digium.com>
* configs/ccss.conf.sample: Fix a gaffe in the CCSS sample
configuration. Discovered by Philippe Lindheimer and pointed out
on #asterisk-dev
2011-02-09 21:44 +0000 [r307314] Andrew Latham <lathama@gmail.com>
* contrib/init.d/rc.debian.asterisk: Disable color during running
test (closes issue #18776) Reported by: alecdavis Patches:
ast_deb_init.diff uploaded by lathama (license 1028) Tested by:
andrel, lathama
2011-02-09 21:06 +0000 [r307228-307273] Jeff Peeler <jpeeler@digium.com>
* main/astobj2.c: Add missing debug info for ao2_link for use with
REF_DEBUG in ao2 callback. (closes issue #18758) Reported by:
rgagnon Patches: branch-1.8-r306540-astobj-fix.diff uploaded by
rgagnon (license 1202) trunk-r306540-astobj-fix.diff uploaded by
rgagnon (license 1202)
* /, main/features.c: Merged revisions 307227 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r307227 | jpeeler | 2011-02-09 13:52:12 -0600 (Wed, 09 Feb 2011)
| 11 lines Make sure to set parking dial context for non-default
parking lots. Since parking_con_dial isn't settable, set all
parking lots to "park-dial". (closes issue #17946) Reported by:
bluecrow76 Patches:
asterisk-1.8.0-beta4-multipark-fixes-2010SEP02.diff uploaded by
bluecrow76 (license 270) modified by me ........
2011-02-09 05:39 +0000 [r307142] Tilghman Lesher <tilghman@meg.abyt.es>
* main/lock.c: Initialize tracking variable in structure properly.
Fixes a memory leak. (Reported by The_Boy_Wonder on IRC, fixed by
me.)
2011-02-08 21:24 +0000 [r307092] Jason Parker <jparker@digium.com>
* main/logger.c: Fix issue with verbose messages not showing on
remote console. This code was reworked recently, and since the
logchannel list hadn't been created yet at this point, and it was
a verbose message, it was being dropped on the floor. Now it'll
continue on to where it should be handled. (closes issue #18580)
Reported by: pabelanger
2011-02-08 21:13 +0000 [r307065] Mark Michelson <mmichelson@digium.com>
* main/ccss.c: Add a couple of useful channel variables for the CC
recall macro. CC_EXTEN and CC_CONTEXT will allow you to determine
the channel and context that will be called when the recall
occurs.
2011-02-08 20:22 +0000 [r306999] Andrew Latham <lathama@gmail.com>
* doc/asterisk.sgml, doc/asterisk.8, configs/asterisk.conf.sample,
configs/voicemail.conf.sample: Documentation Updates Note default
polling setting in voicemail.conf Add missing config to
asterisk.conf Update manpage (issue #16505) Reported by: tzafrir
Patches: asterisk_sgml_fixes_demo.diff uploaded by tzafrir
(license 46) Tested by: lathama, tzafrir
2011-02-08 20:18 +0000 [r306979] Terry Wilson <twilson@digium.com>
* /, channels/chan_sip.c: Merged revisions 306973 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r306973 | twilson | 2011-02-08 12:14:09 -0800
(Tue, 08 Feb 2011) | 9 lines Merged revisions 306972 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r306972 | twilson | 2011-02-08 12:05:13 -0800 (Tue, 08
Feb 2011) | 2 lines Fix comparison for REFER Replaces tags with
pedantic=yes ........ ................
2011-02-08 19:41 +0000 [r306866-306967] Jeff Peeler <jpeeler@digium.com>
* apps/app_voicemail.c, /: Merged revisions 306966 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r306966 | jpeeler | 2011-02-08 13:41:21 -0600
(Tue, 08 Feb 2011) | 9 lines Merged revisions 306965 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r306965 | jpeeler | 2011-02-08 13:40:58 -0600 (Tue, 08
Feb 2011) | 1 line fix this line again ........ ................
* apps/app_voicemail.c, /: Merged revisions 306961 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r306961 | jpeeler | 2011-02-08 13:25:10 -0600
(Tue, 08 Feb 2011) | 15 lines Merged revisions 306960 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r306960 | jpeeler | 2011-02-08 13:18:50 -0600 (Tue, 08 Feb 2011)
| 9 lines Backup file storing message duration is not used with
IMAP_STORAGE, remove code. The message duration is stored in the
body of the email when using IMAP_STORAGE, so nothing needs to
happen with the backup file. (closes issue #18718) Reported by:
kerframil ........ ................
* apps/app_voicemail.c, /: Merged revisions 306865 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r306865 | jpeeler | 2011-02-08 10:21:25 -0600
(Tue, 08 Feb 2011) | 9 lines Merged revisions 306864 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r306864 | jpeeler | 2011-02-08 10:19:17 -0600 (Tue, 08
Feb 2011) | 1 line make this safer and fully correct, pointed out
by Steve Davis ........ ................
2011-02-08 01:45 +0000 [r306826] Andrew Latham <lathama@gmail.com>
* UPGRADE.txt, include/asterisk/manager.h, doc/asterisk.sgml,
include/asterisk/doxygen/mantisworkflow.h: Documentation Updates.
More updates to the removed doc folder and start updates to the
man page. (issue #16505) Reported by: tzafrir Tested by: lathama
2011-02-07 22:43 +0000 [r306619-306674] Terry Wilson <twilson@digium.com>
* /, main/features.c: Merged revisions 306673 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r306673 | twilson | 2011-02-07 14:40:20 -0800
(Mon, 07 Feb 2011) | 17 lines Merged revisions 306672 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r306672 | twilson | 2011-02-07 14:35:20 -0800 (Mon, 07 Feb 2011)
| 10 lines Don't try to pickup a call in the middle of a
masquerade If A calls B which doesn't answer and C & D both try
to do a call pickup, it is possible for ast_pickup_call to answer
the call, then fail to masquerade one of the calls because the
other one is already in the process of masquerading. This patch
checks to see if the channel is in the process of masquerading
before call before selecting it for a pickup. Review:
https://reviewboard.asterisk.org/r/1094/ ........
................
* /, channels/chan_sip.c: Merged revisions 306618 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r306618 | twilson | 2011-02-07 13:59:54 -0800
(Mon, 07 Feb 2011) | 17 lines Merged revisions 306617 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r306617 | twilson | 2011-02-07 13:51:43 -0800 (Mon, 07 Feb 2011)
| 10 lines Don't allow a REFER w/replaces to replace its own
dialog Asterisk currently accepts a REFER with a Refer-To with an
embedded Replaces header that matches the dialog of the REFER.
This would be a situation like A calls B, A calls C, A transfers
B to A, which is just silly. This patch makes the transfer fail
instead of making Asterisk freak out and forget to hang other
channels up. Review: https://reviewboard.asterisk.org/r/1093/
........ ................
2011-02-07 17:36 +0000 [r306575] Mark Michelson <mmichelson@digium.com>
* main/ccss.c: Rearrange a bit of code in the generic CC recall
operation. By waiting to call the callback macro after the
CC_INTERFACES, extension, priority, and context have been set,
this information can be accessed more easily within the callback
macro. Reported by Philippe Lindheimer.
2011-02-04 19:24 +0000 [r306356] Jason Parker <jparker@digium.com>
* apps/app_queue.c, /: Merged revisions 306346 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r306346 | qwell | 2011-02-04 13:21:43 -0600 (Fri, 04 Feb 2011) |
9 lines Don't fallthrough to 'unknown' in the 'ringing' case.
This could cause improper exits from the queue. (closes issue
#18499) Reported by: zaltar Patches: app_queue.patch uploaded by
zaltar (license 1148) ........
2011-02-04 18:53 +0000 [r306324] Richard Mudgett <rmudgett@digium.com>
* apps/app_queue.c, apps/app_dial.c: Don't send redirecting updates
to the caller if the dialplan forked the call. Each fork in the
dial could be redirected and confuse the caller. For ISDN the
DivLeg1 and DivLeg3 messages would get confused because ISDN
redirects calls in sequence not in parallel. * Also fixed a
formatting inconsistency in app_dial.c and make a warning message
more useful about what frame type could not be written.
2011-02-03 23:49 +0000 [r306215] Jeff Peeler <jpeeler@digium.com>
* channels/chan_sip.c: Fix SIP deadlock involving state changes.
Once again a call to pbx_builtin_getvar_helper (and
pbx_builtin_setvar_helper) has caused locking problems. Both of
these functions lock the channel when the channel argument is
passed in! In this case, the suspected problem (the backtrace
makes it impossible to tell) was the private being locked in
sip_set_rtp_peer and then: transmit_reinvite_with_sdp
try_suggested_sip_codec pbx_builtin_getvar_helper (Traced to
verify that the fix was only required in 1.8 and later.) (closes
issue #18491) Reported by: cmaj Patches:
chan_sip_fix_deadlocks_bug_18491.txt uploaded by cmaj (license
830) Tested by: cmaj
2011-02-03 21:03 +0000 [r306127] Terry Wilson <twilson@digium.com>
* channels/chan_local.c, /: Merged revisions 306126 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r306126 | twilson | 2011-02-03 12:56:00 -0800
(Thu, 03 Feb 2011) | 16 lines Merged revisions 306119 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r306119 | twilson | 2011-02-03 12:36:34 -0800 (Thu, 03 Feb 2011)
| 9 lines Set hangup cause in local_hangup When a call involves a
local channel (like SIP -> Local -> SIP), the hangup cause was
not being set. This resulted in SIP channels sometimes getting a
503 error instead of a 486 when the far side sent a busy. In
Asterisk 1.8+ this also can cause issues with CCSS that involve a
local channel. This patch sets the hangupcause for one side of
the local channel to the other in local_hangup for outbound
calls. ........ ................
2011-02-03 20:50 +0000 [r306124] Jeff Peeler <jpeeler@digium.com>
* /, main/features.c: Merged revisions 306123 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r306123 | jpeeler | 2011-02-03 14:49:48 -0600 (Thu, 03 Feb 2011)
| 10 lines Set exception on channel in parking thread when
POLLPRI event detected. This is done just to make the code be
equivalent to the old select code. As noted in 303106 the same
issue was already fixed in this branch, but the exception was not
set on the channel in the case of POLLPRI. The reason that this
did not cause a problem here is because in 122923 the check in
__ast_read to check the exception flag was removed. (related to
#18637) ........
2011-02-03 15:50 +0000 [r305987] Andrew Latham <lathama@gmail.com>
* phoneprov/snom-mac.xml (added), configs/phoneprov.conf.sample, /:
res_phoneprov add snom 300, 320, 360, 370, 820, 821, 870 support
(issue #18713) Reported by: lathama Patches: snom_dir.diff
uploaded by lathama (license 1028) Tested by: lathama
2011-02-03 00:24 +0000 [r305923] Richard Mudgett <rmudgett@digium.com>
* main/channel.c, main/manager.c, /, channels/chan_sip.c,
apps/app_sendtext.c: Merged revisions 305889 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r305889 | rmudgett | 2011-02-02 18:15:07 -0600
(Wed, 02 Feb 2011) | 17 lines Merged revisions 305888 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r305888 | rmudgett | 2011-02-02 18:02:43 -0600 (Wed, 02 Feb 2011)
| 8 lines Minor AST_FRAME_TEXT related issues. * Include the null
terminator in the buffer length. When the frame is queued it is
copied. If the null terminator is not part of the frame buffer
length, the receiver could see garbage appended onto it. * Add
channel lock protection with ast_sendtext(). * Fixed AMI SendText
action ast_sendtext() return value check. ........
................
2011-02-02 20:05 +0000 [r305844] Tilghman Lesher <tilghman@meg.abyt.es>
* funcs/func_env.c: Eliminate a file descriptor leak when using the
FILE() dialplan function. (closes issue #18731) Reported by:
marioabajo
2011-02-02 19:27 +0000 [r305753-305838] Andrew Latham <lathama@gmail.com>
* apps/app_externalivr.c, configs/sip.conf.sample,
configs/skinny.conf.sample, configs/h323.conf.sample,
configs/sla.conf.sample, apps/app_voicemail.c,
configs/iax.conf.sample, funcs/func_enum.c,
configs/dundi.conf.sample, funcs/func_callcompletion.c,
configs/mgcp.conf.sample, configs/iaxprov.conf.sample,
configs/unistim.conf.sample: Replacing doc/* and asterisk.pdf
with wiki links Adding links to http(s)://wiki.asterisk.org
* configs/ccss.conf.sample, configs/sip.conf.sample,
configs/skinny.conf.sample, main/config.c,
configs/h323.conf.sample, configs/sla.conf.sample,
main/ast_expr2.fl, res/res_srtp.c,
configs/chan_dahdi.conf.sample, configs/extconfig.conf.sample,
configs/res_snmp.conf.sample, main/ast_expr2f.c,
res/res_timing_dahdi.c: Replacing doc/* with wiki links Adding
links to http(s)://wiki.asterisk.org
* channels/chan_sip.c: Replace link to old doc with new wiki page.
Link to
https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
2011-02-01 22:48 +0000 [r305692] Jason Parker <jparker@digium.com>
* channels/chan_iax2.c: Reverse sense of an error test when reading
from astdb. (closes issue #18545) Reported by: jcovert Patches:
chan_iax2.c.patch uploaded by jcovert (license 551)
2011-02-01 21:14 +0000 [r305649] Andrew Latham <lathama@gmail.com>
* configs/sip.conf.sample: SIP Configuration Documentation sip show
settings reports qualifyfreq in milliseconds. sip.conf configures
qualifyfreg in seconds.
2011-02-01 19:23 +0000 [r305603] Brett Bryant <bbryant@digium.com>
* cel/cel_pgsql.c: Add a possible solution to a customer problem
with reloading cel_pgsql.so quickly.
2011-02-01 18:02 +0000 [r305560] Andrew Latham <lathama@gmail.com>
* CHANGES, Makefile, README, /: doc/tex dir removed, but
corresponding entries still exists Update README, CHANGES, and
Makefile. Direct users to http://wiki.asterisk.org for
documentation or to the AST.txt and AST.pdf included in the
tarball. (closes issue #18443) Reported by: bas Patches:
changes.diff uploaded by lathama (license 1028) readme.diff
uploaded by lathama (license 1028) Tested by: lathama bas
2011-02-01 17:04 +0000 [r305473] Jason Parker <jparker@digium.com>
* res/res_musiconhold.c, /: Merged revisions 305472 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r305472 | qwell | 2011-02-01 11:02:09 -0600
(Tue, 01 Feb 2011) | 16 lines Merged revisions 305471 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r305471 | qwell | 2011-02-01 11:00:55 -0600 (Tue, 01 Feb 2011) |
9 lines Close file descriptor for timing source when a MOH class
gets destroyed. (closes issue #18457) Reported by: mcallist
Patches: 18457-closetimer.diff uploaded by qwell (license 4)
18457-closetimer_trunk.diff uploaded by qwell (license 4) Tested
by: qwell, loloski ........ ................
2011-02-01 00:01 +0000 [r305343] Richard Mudgett <rmudgett@digium.com>
* channels/sig_pri.c, /: Merged revisions 305342 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r305342 | rmudgett | 2011-01-31 17:50:10 -0600
(Mon, 31 Jan 2011) | 14 lines Merged revisions 305341 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r305341 | rmudgett | 2011-01-31 17:45:58 -0600 (Mon, 31 Jan 2011)
| 7 lines Obtain the pri lock for PRI queue counters. Need to
obtain the pri lock when calling pri_dump_info_str() to avoid a
reentrancy problem when calculating the Q.921 Q count statistic.
JIRA AST-484 ........ ................
2011-01-31 23:07 +0000 [r305131-305254] Jason Parker <jparker@digium.com>
* apps/app_dial.c, /, channels/chan_sip.c: Merged revisions 305253
via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r305253 | qwell | 2011-01-31 16:59:34 -0600
(Mon, 31 Jan 2011) | 17 lines Merged revisions 305252 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r305252 | qwell | 2011-01-31 16:56:54 -0600 (Mon, 31 Jan 2011) |
10 lines Prevent a crash when dialing a technology with no
destination (ex: Dial(SIP/)) chan_iax2 and other channel drivers
already had code to prevent this. The attempt that app_dial was
making to prevent it was not correct, so I fixed that. (closes
issue #18371) Reported by: gbour Patches: 18371.patch uploaded by
gbour (license 1162) ........ ................
* configs/sip.conf.sample, main/tcptls.c: Add alternative name for
config option. The SIP sample configuration had "tlscadir" as the
option name, but chan_sip used the more correct "tlscapath". Now
both are accepted. Discovered (sort of) by a user on IRC in
#asterisk
* res/res_musiconhold.c: Fix compile error. pseudofd no longer
exists.
* res/res_musiconhold.c, /: Merged revisions 305130 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r305130 | qwell | 2011-01-31 14:59:37 -0600
(Mon, 31 Jan 2011) | 9 lines Merged revisions 305129 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r305129 | qwell | 2011-01-31 14:56:25 -0600 (Mon, 31 Jan
2011) | 2 lines Set file descriptors to -1 on creation, so that
we don't see weirdness later. ........ ................
2011-01-31 13:56 +0000 [r305083] Andrew Latham <lathama@gmail.com>
* main/http.c: Asterisk HTTP response Content-type Address content
type for BSD and other platforms (closes issue #18456) Reported
by: alexo Patches: asterisk18_http.patch uploaded by alexo
(license 1175) Tested by: alexo
2011-01-31 07:51 +0000 [r304950-305040] Tilghman Lesher <tilghman@meg.abyt.es>
* include/asterisk/lock.h: Use the non-specific API aliases, to
avoid a problem with building the utils directory.
* apps/app_voicemail.c, /: Merged revisions 304978 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r304978 | tilghman | 2011-01-31 01:25:14 -0600
(Mon, 31 Jan 2011) | 9 lines Merged revisions 304952 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r304952 | tilghman | 2011-01-31 00:54:45 -0600 (Mon, 31
Jan 2011) | 2 lines Fix compilation when ODBC_STORAGE is defined.
........ ................
* main/utils.c, include/asterisk/lock.h, .cleancount, main/lock.c,
main/heap.c: Change mutex tracking so that it only consumes
memory in the core mutex object when it's actually being used.
This reduces the overall size of a mutex which was 3016 bytes
before this back down to 216 bytes (this is on 64-bit Linux with
a glibc-implemented mutex). The exactness of the numbers here may
vary slightly based upon how mutexes are implemented on a
platform, but the long and short of it is that prior to this
commit, chan_iax2 held down 98MB of memory on a 64-bit system for
nothing more than a table of 32767 locks. After this commit, the
same table occupies a mere 7MB of memory. (closes issue #18194)
Reported by: job Patches: 20110124__issue18194.diff.txt uploaded
by tilghman (license 14) Tested by: tilghman Review:
https://reviewboard.asterisk.org/r/1066
2011-01-30 00:11 +0000 [r304908] Andrew Latham <lathama@gmail.com>
* apps/app_externalivr.c, apps/app_queue.c, apps/app_voicemail.c,
funcs/func_realtime.c, res/res_calendar.c,
funcs/func_callcompletion.c: Add Function and Application
Relationships to documentation Add and extend the see-also
sections to the documentation for applications and functions in
an effort to expand the online documentation of the wiki. Also
check for and update any links to moved documentation in the doc
folder.
2011-01-29 23:07 +0000 [r304638-304866] Sean Bright <sean@malleable.com>
* res/res_config_ldap.c, /: Merged revisions 304865 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
........ r304865 | seanbright | 2011-01-29 18:05:25 -0500 (Sat,
29 Jan 2011) | 7 lines Plug some memory leaks in the LDAP
realtime driver. (closes issue #18435) Reported by: zaltar
Patches: res_config_ldap.patch uploaded by zaltar (license 1148)
........
* /, apps/app_meetme.c: Merged revisions 304776 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r304776 | seanbright | 2011-01-29 13:08:14 -0500 (Sat, 29 Jan
2011) | 15 lines If we fail to allocate our announcement objects,
make sure we don't leak objects. The majority of this patch was
committed already in r304726 and r304729. (issue #18225) Reported
by: kenji (issue #18444) Reported by: junky (closes issue #18343)
Reported by: kobaz Patches: meetme-refs.diff uploaded by kobaz
(license 834) ........
* /, apps/app_meetme.c: Merged revisions 304773 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r304773 | seanbright | 2011-01-29 12:51:28 -0500 (Sat, 29 Jan
2011) | 9 lines When we pass the S() or L() options to MeetMe,
make sure that we honor C as well. Without this patch, if the
user was kicked from the conference via the S() or L() mechanism,
we would just hang up on them even if we also passed C (continue
in dialplan when kicked). With this patch we honor the C flag in
those cases. (closes issue #17317) Reported by: var ........
* /, apps/app_meetme.c: Merged revisions 304729 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r304729 | seanbright | 2011-01-29 12:01:51 -0500 (Sat, 29 Jan
2011) | 15 lines Make sure that we unref the correct object when
ejecting the most recent caller. Currently, when we kick the last
user to enter, we decrement our own reference count which results
in a crash when we kick another user or when we exit the
conference ourselves. This will fix #18225 in 1.8 and trunk, but
that particular bug does not exist in 1.6.2. (closes issue
#18225) Reported by: kenji Patches: issue18225.patch uploaded by
seanbright (license 71) Tested by: seanbright ........
* /, apps/app_meetme.c: Merged revisions 304726 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r304726 | seanbright | 2011-01-29 11:26:57 -0500 (Sat, 29 Jan
2011) | 9 lines Fix user reference leak in MeetMe. We were
unlinking the user from the conferences user container, but not
decrementing the reference count of the user as well, resulting
in a leak. (closes issue #18444) Reported by: junky Tested by:
seanbright ........
* /, apps/app_meetme.c: Merged revisions 304659,304682 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
........ r304659 | seanbright | 2011-01-28 16:22:09 -0500 (Fri,
28 Jan 2011) | 5 lines Don't leak references if we can't create a
pseudo channel for mixing in MeetMe. If there was a problem
allocating a pseudo channel when building our meetme, we weren't
destroying our user container or destroying the mutexes that we
created. ........ r304682 | seanbright | 2011-01-28 17:38:05
-0500 (Fri, 28 Jan 2011) | 2 lines Revert part of the previous
commit that snuck in. ........
* main/acl.c: Restore some conditionals that we lost in r277814.
There are some cases where ast_append_ha() is called with a NULL
instead of a valid int pointer. So if we get a NULL, don't try to
dereference it. (closes issue #18162) Reported by: imcdona
Patches: issue0018162.patch uploaded by pabelanger (license 224)
Tested by: enegaard
2011-01-27 19:08 +0000 [r304554] Richard Mudgett <rmudgett@digium.com>
* main/ccss.c: Warning message if CALLCOMPLETION(cc_callback_macro
or cc_agent_dialstring) are empty. Test if the value pointer is
not NULL instead of not ast_strlen_zero().
2011-01-27 17:03 +0000 [r304462-304466] Jason Parker <jparker@digium.com>
* /, configure, configure.ac: Merged revisions 304465 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r304465 | qwell | 2011-01-27 11:01:24 -0600
(Thu, 27 Jan 2011) | 16 lines Merged revisions 304464 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r304464 | qwell | 2011-01-27 10:57:46 -0600 (Thu, 27 Jan 2011) |
9 lines Fix default prefix=/usr regression on non-Linux systems.
This partially reverts a change made in branches/1.4/ r267759,
which will cause issue #17013 to be reopened. This issue was
pointed out by a user on #asterisk, who helpfully discovered that
paths were being set incorrectly. To truly understand what was
wrong, one should run: svn diff --force -c<this revision>
configure ........ ................
* /, configure: Merged revisions 304461 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r304461 | qwell | 2011-01-27 10:48:00 -0600
(Thu, 27 Jan 2011) | 9 lines Merged revisions 304460 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r304460 | qwell | 2011-01-27 10:47:03 -0600 (Thu, 27 Jan
2011) | 1 line Rerun bootstrap.sh with no changes, so that it is
more obvious what my next commit changes. ........
................
2011-01-26 22:27 +0000 [r304339] Jeff Peeler <jpeeler@digium.com>
* /, main/features.c: Merged revisions 304338 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r304338 | jpeeler | 2011-01-26 16:26:37 -0600 (Wed, 26 Jan 2011)
| 2 lines Change delimiter used internally for GOTO_ON_BLINDXFR
to commas to match 76703. ........
2011-01-26 21:02 +0000 [r304251] Mark Michelson <mmichelson@digium.com>
* main/udptl.c, /: Merged revisions 304250 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r304250 | mmichelson | 2011-01-26 15:02:10 -0600
(Wed, 26 Jan 2011) | 9 lines Merged revisions 304242 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r304242 | mmichelson | 2011-01-26 14:38:37 -0600 (Wed,
26 Jan 2011) | 3 lines Get rid of unused 'verbose' field in
ast_udptl ........ ................
2011-01-26 20:43 +0000 [r304245] Matthew Nicholson <mnicholson@digium.com>
* channels/sip/include/sip.h,
channels/sip/include/reqresp_parser.h, /, channels/chan_sip.c,
channels/sip/reqresp_parser.c: Merged revisions 304244 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r304244 | mnicholson | 2011-01-26 14:42:16 -0600
(Wed, 26 Jan 2011) | 13 lines Merged revisions 304241 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r304241 | mnicholson | 2011-01-26 14:38:22 -0600 (Wed, 26 Jan
2011) | 6 lines This patch modifies chan_sip to route responses
to the address the request came from. It also modifies chan_sip
to respect the maddr parameter in the Via header. ABE-2664
Review: https://reviewboard.asterisk.org/r/1059/ ........
................
2011-01-26 20:23 +0000 [r304186] Sean Bright <sean@malleable.com>
* /, configs/queues.conf.sample: Merged revisions 304181 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r304181 | seanbright | 2011-01-26 15:22:47 -0500
(Wed, 26 Jan 2011) | 9 lines Merged revisions 304159 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r304159 | seanbright | 2011-01-26 15:18:29 -0500 (Wed,
26 Jan 2011) | 1 line Make sure the sample queues.conf is
properly commented. ........ ................
2011-01-26 19:39 +0000 [r304150] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c, /: Merged revisions 304149 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r304149 | rmudgett | 2011-01-26 13:38:38 -0600
(Wed, 26 Jan 2011) | 9 lines Merged revisions 304148 from
https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
.......... r304148 | rmudgett | 2011-01-26 13:23:46 -0600 (Wed,
26 Jan 2011) | 2 lines Update documentation for
DAHDISendCallreroutingFacility() application. ..........
................
2011-01-26 01:26 +0000 [r304097] Sean Bright <sean@malleable.com>
* /, main/file.c: Merged revisions 304096 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r304096 | seanbright | 2011-01-25 20:24:58 -0500 (Tue, 25 Jan
2011) | 12 lines Per the man page, setvbuf() must be called
before any other operation on an open file. We use setvbuf() to
associate a buffer with a stream, but we have already written to
the open file. This works (by chance) on Linux, but fails on
other platforms, such as OpenSolaris. (closes issue #16610)
Reported by: bklang Patches: setvbuf.patch uploaded by crjw
(license 963) Tested by: bklang, asgaroth, efutch ........
2011-01-25 23:28 +0000 [r304007] Richard Mudgett <rmudgett@digium.com>
* /, main/features.c: Merged revisions 304006 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r304006 | rmudgett | 2011-01-25 17:25:32 -0600
(Tue, 25 Jan 2011) | 15 lines Merged revisions 304005 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r304005 | rmudgett | 2011-01-25 17:21:09 -0600 (Tue, 25 Jan 2011)
| 8 lines DTMF attended transfers sometimes fail for no apparent
reason. The loop in feature_request_and_dial() can exit when
Party C has answered without processing an AST_CONTROL_ANSWER.
Also sometimes an AST_CONTROL_ANSWER never happens even though
Party C has answered. Don't hangup Party C if he is up or we
receive an AST_CONTROL_ANSWER. ........ ................
2011-01-25 22:09 +0000 [r303962] Terry Wilson <twilson@digium.com>
* /, channels/chan_sip.c: Merged revisions 303960 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r303960 | twilson | 2011-01-25 16:02:42 -0600
(Tue, 25 Jan 2011) | 23 lines Merged revisions 303906 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r303906 | twilson | 2011-01-25 14:50:59 -0600 (Tue, 25 Jan 2011)
| 16 lines Guard against retransmitting BYEs indefinitely In the
case of an attended transfer (A calls B, A atxfers to C) where A
becomes unreachable before replying to Asterisk's BYE, Asterisk
can sometimes retransmit the BYE indefinitely. This is because
__sip_autodestruct tests p->refer && !ast_test_flag(&p->flags[0],
SIP_ALREADYGONE and will then transmit a BYE. When this BYE times
out, it will not ever be marked as ALREADYGONE, so when
__sip_autodestruct is called again, we end up starting the cycle
over. This patch adds a call to sip_alreadygone(pkt->owner) in
retrans_pkt in the case of a BYE that has timed out. This should
prevent Asterisk from trying to transmit new BYE messages in the
future. Review: https://reviewboard.asterisk.org/r/1077/ ........
................
2011-01-25 20:56 +0000 [r303907] Matthew Nicholson <mnicholson@digium.com>
* include/asterisk/res_fax.h, res/res_fax.c: Reimplemented fax
session reservation to reverse the ABI breakage introduced in
r297486.
2011-01-25 18:55 +0000 [r303860] Tilghman Lesher <tilghman@meg.abyt.es>
* /, channels/chan_sip.c: Merged revisions 303858 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r303858 | tilghman | 2011-01-25 12:41:26 -0600 (Tue, 25 Jan 2011)
| 5 lines Fix "sip show user <tab>", so that it actually shows
results, instead of just completing the last entry. (closes issue
#16675) Reported by: pj ........
2011-01-25 17:49 +0000 [r303771] Richard Mudgett <rmudgett@digium.com>
* channels/sig_pri.c, channels/sig_ss7.c, channels/sig_pri.h,
channels/chan_dahdi.c, channels/sig_ss7.h, /: Merged revisions
303769 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r303769 | rmudgett | 2011-01-25 11:42:42 -0600
(Tue, 25 Jan 2011) | 47 lines Merged revisions 303765 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r303765 | rmudgett | 2011-01-25 11:36:50 -0600 (Tue, 25 Jan 2011)
| 40 lines Sending out unnecessary PROCEEDING messages breaks
overlap dialing. Issue #16789 was a good idea. Unfortunately, it
breaks overlap dialing through Asterisk. There is not enough
information available at this point to know if dialing is
complete. The ast_exists_extension(), ast_matchmore_extension(),
and ast_canmatch_extension() calls are not adequate to detect a
dial through extension pattern of "_9!". Workaround is to use the
dialplan Proceeding() application early in non-dial through
extensions. * Effectively revert issue #16789. * Allow outgoing
overlap dialing to hear dialtone and other early media. A
PROGRESS "inband-information is now available" message is now
sent after the SETUP_ACKNOWLEDGE message for non-digital calls.
An AST_CONTROL_PROGRESS is now generated for incoming
SETUP_ACKNOWLEDGE messages for non-digital calls. * Handling of
the AST_CONTROL_CONGESTION in chan_dahdi/sig_pri was inconsistent
with the cause codes. * Added better protection from sending out
of sequence messages by combining several flags into a single
enum value representing call progress level. * Added diagnostic
messages for deferred overlap digits handling corner cases.
(closes issue #17085) Reported by: shawkris (closes issue #18509)
Reported by: wimpy Patches: issue18509_early_media_v1.8_v3.patch
uploaded by rmudgett (license 664) Expanded upon
issue18509_early_media_v1.8_v3.patch to include analog and SS7
because of backporting requirements. Tested by: wimpy, rmudgett
........ ................
2011-01-25 17:02 +0000 [r303678] Jeff Peeler <jpeeler@digium.com>
* apps/app_voicemail.c, /: Merged revisions 303677 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r303677 | jpeeler | 2011-01-25 10:59:28 -0600
(Tue, 25 Jan 2011) | 26 lines Merged revisions 303676 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r303676 | jpeeler | 2011-01-25 10:58:29 -0600 (Tue, 25 Jan 2011)
| 20 lines Fix voicemail sequencing for file based storage. A
previous change was made to account for when the number of
voicemail messages exceeds the max limit to be handled properly,
but it caused gaps in the messages to not be properly handled.
This has now been resolved. In later non 1.4 branches, it appears
that resequencing wasn't even occurring due from what appears and
accidental code removal. (closes issue #18498) Reported by:
JJCinAZ Patches: bug18498v2.patch uploaded by jpeeler (license
325) (closes issue #18486) Reported by: bluefox Patches:
bug18486.patch uploaded by jpeeler (license 325) ........
................
2011-01-24 20:51 +0000 [r303549] Russell Bryant <russell@digium.com>
* include/asterisk/channel.h, main/channel.c, main/pbx.c, /,
apps/app_meetme.c, main/features.c: Merged revisions 303548 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r303548 | russell | 2011-01-24 14:49:53 -0600
(Mon, 24 Jan 2011) | 38 lines Merged revisions 303546 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r303546 | russell | 2011-01-24 14:32:21 -0600 (Mon, 24 Jan 2011)
| 31 lines Fix channel redirect out of MeetMe() and other issues
with channel softhangup. Mantis issue #18585 reports that a
channel redirect out of MeetMe() stopped working properly. This
issue includes a patch that resolves the issue by removing a call
to ast_check_hangup() from app_meetme.c. I left that in my patch,
as it doesn't need to be there. However, the rest of the patch
fixes this problem with or without the change to app_meetme. The
key difference between what happens before and after this patch
is the effect of the END_OF_Q control frame. After END_OF_Q is
hit in ast_read(), ast_read() will return NULL. With the
ast_check_hangup() removed, app_meetme sees this which causes it
to exit as intended. Checking ast_check_hangup() caused
app_meetme to exit earlier in the process, and the target of the
redirect saw the condition where ast_read() returned NULL.
Removing ast_check_hangup() works around the issue in app_meetme,
but doesn't solve the issue if another application did the same
thing. There are also other edge cases where if an application
finishes at the same time that a redirect happens, the target of
the redirect will think that the channel hung up. So, I made some
changes in pbx.c to resolve it at a deeper level. There are
already places that unset the SOFTHANGUP_ASYNCGOTO flag in an
attempt to abort the hangup process. My patch extends this to
remove the END_OF_Q frame from the channel's read queue, making
the "abort hangup" more complete. This same technique was used in
every place where a softhangup flag was cleared. (closes issue
#18585) Reported by: oej Tested by: oej, wedhorn, russell Review:
https://reviewboard.asterisk.org/r/1082/ ........
................
2011-01-24 17:20 +0000 [r303467] Jason Parker <jparker@digium.com>
* channels/chan_dahdi.c, /: Merged revisions 303285 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r303285 | qwell | 2011-01-21 15:48:09 -0600
(Fri, 21 Jan 2011) | 15 lines Merged revisions 303284 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r303284 | qwell | 2011-01-21 15:45:34 -0600 (Fri, 21 Jan 2011) |
8 lines Reset configuration before parsing users.conf. Some
values configured in chan_dahdi.conf were able to leak in to
users.conf configuration. This was surprising users, and
potentially setting non-sane "defaults". ASTNOW-125 ........
................
2011-01-21 23:11 +0000 [r303286-303375] Jason Parker <jparker@digium.com>
* channels/chan_dahdi.c, /: Temporarily revert r303286
* channels/chan_dahdi.c, /: Merged revisions 303285 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r303285 | qwell | 2011-01-21 15:48:09 -0600
(Fri, 21 Jan 2011) | 15 lines Merged revisions 303284 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r303284 | qwell | 2011-01-21 15:45:34 -0600 (Fri, 21 Jan 2011) |
8 lines Reset configuration before parsing users.conf. Some
values configured in chan_dahdi.conf were able to leak in to
users.conf configuration. This was surprising users, and
potentially setting non-sane "defaults". ASTNOW-125 ........
................
2011-01-20 20:31 +0000 [r303153] Richard Mudgett <rmudgett@digium.com>
* main/ccss.c: Merged revision 303098 from
https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
.......... r303098 | rmudgett | 2011-01-20 12:11:45 -0600 (Thu,
20 Jan 2011) | 15 lines CC_INTERFACES does not get built
correctly with local channels. If local channels are used with
CCSS, CC_INTERFACES gets garbage prepended to it so the CC recall
fails. Also CC_INTERFACES gets "&(null)" appended to it. *
Initialize the buffer to eliminate the prepended garbage. *
Filter out the empty interface strings to eliminate the latter. *
Added a diagnostic message if the CC_INTERFACES is ever empty.
JIRA ABE-2740 JIRA SWP-2848 ..........
2011-01-20 19:57 +0000 [r303107] Shaun Ruffell <sruffell@digium.com>
* /, main/features.c: Merged revisions 303106 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r303106 | sruffell | 2011-01-20 13:56:34 -0600 (Thu, 20 Jan 2011)
| 15 lines main/features: Use POLLPRI when waiting for events on
parked channels. This change resolves a regression in the 1.6.2
when converting from select to poll. The DAHDI timers use POLLPRI
to indicate that the timer fired, but features was not waiting
for that flag. The result was no audio for MOH when a call was
parked and res_timing_dahdi was in use. This patch is slightly
modified from the one on the mantis issue. It does not set an
exception on the channel if the POLLPRI flag is set. (closes
issue #18262) Reported by: francesco_r Patches:
patch_park_moh-trunk-2.txt uploaded by cjacobsen (license 1029)
Tested by: francesco_r, rfrantik, one47 ........
2011-01-20 17:10 +0000 [r303009] Jeff Peeler <jpeeler@digium.com>
* apps/app_queue.c, /, configs/queues.conf.sample: Merged revisions
303008 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r303008 | jpeeler | 2011-01-20 11:07:44 -0600
(Thu, 20 Jan 2011) | 14 lines Merged revisions 303007 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r303007 | jpeeler | 2011-01-20 11:04:08 -0600 (Thu, 20 Jan 2011)
| 8 lines Add new queue strategy to preserve behavior for when
queue members moved to ao2. Add queue strategy called "rrordered"
to mimic old behavior from when queue members were stored in a
linked list. ABE-2707 ........ ................
2011-01-20 16:12 +0000 [r302921] Russell Bryant <russell@digium.com>
* /, apps/app_privacy.c: Merged revisions 302920 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r302920 | russell | 2011-01-20 10:11:58 -0600 (Thu, 20 Jan 2011)
| 2 lines Resolve a compiler warning. ........
2011-01-20 15:45 +0000 [r302918] Leif Madsen <lmadsen@digium.com>
* apps/app_dial.c, /: Merged revisions 302917 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r302917 | lmadsen | 2011-01-20 09:42:05 -0600 (Thu, 20 Jan 2011)
| 8 lines Option L() is milliseconds, not seconds. > Change the
verbose output of option L() to say milliseconds and not seconds
> as the value is in milliseconds. > > (closes issue #18264) >
Reported by: jacco > Patches: > app_dial_patch.txt uploaded by
lmadsen (license 10) ........
2011-01-19 23:56 +0000 [r302837] Russell Bryant <russell@digium.com>
* main/manager.c: Only check container count if it exists.
2011-01-19 23:49 +0000 [r302834] Sean Bright <sean@malleable.com>
* apps/app_voicemail.c, /: Merged revisions 302833 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
........ r302833 | seanbright | 2011-01-19 18:47:22 -0500 (Wed,
19 Jan 2011) | 7 lines Support greetingsfolder as documented in
voicemail.conf.sample. (closes issue #17870) Reported by:
edhorton Patches:
__20100816-app_voicemail-greetingsfolder-support.txt uploaded by
lmadsen (license 10) ........
2011-01-19 23:29 +0000 [r302831] Paul Belanger <pabelanger@digium.com>
* contrib/scripts/install_prereq: Add binutils-dev for
BETTER_BACKTRACES
2011-01-19 23:06 +0000 [r302785-302789] Russell Bryant <russell@digium.com>
* main/manager.c, /: Merged revisions 302788 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r302788 | russell | 2011-01-19 17:06:14 -0600 (Wed, 19 Jan 2011)
| 4 lines Turn a noisy verbose message into a debug message. This
can drown your console if you're using the AMI over HTTP.
........
* main/manager.c: Resolve a memory leak with the manager interface
is disabled. The intent of this check as it stands in previous
versions of Asterisk was to check if there are any active
sessions. If there were no sessions, then the function would
return immediately and not bother with queueing up the manager
event to be processed. Since the conversion of storing sessions
in an astobj2 container, this check will always pass. I changed
it to go back to checking what was intended. The side effect of
this was that if the AMI is disabled, the manager event queue is
populated anyway, but the code that runs to clear out the queue
never runs. A producer with no consumer is a bad thing. Reported
internally by kmorgan.
2011-01-19 21:29 +0000 [r302713] Richard Mudgett <rmudgett@digium.com>
* /, main/features.c: Merged revisions 302693 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r302693 | rmudgett | 2011-01-19 15:25:41 -0600
(Wed, 19 Jan 2011) | 22 lines Merged revisions 302671 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r302671 | rmudgett | 2011-01-19 15:21:56 -0600 (Wed, 19 Jan 2011)
| 15 lines DTMF transfer plays the wrong sounds for wrong number
or other call failure. * Set the default for features.conf.sample
xferfailsound option to "beeperr" as documented instead of
"pbx-invalid" and corrected the use of it in DTMF blind transfer
(#1). * Improved DTMF blind transfer handling of wrong numbers.
Most of the concerns in this issue were taken care of by the
patch for issue 17999: Issues with DTMF triggered attended
transfers. (closes issue #18379) Reported by: gincantalupo Tested
by: rmudgett ........ ................
2011-01-19 21:23 +0000 [r302634-302680] Tilghman Lesher <tilghman@meg.abyt.es>
* include/asterisk/astdb.h, /: Merged revisions 302675 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r302675 | tilghman | 2011-01-19 15:22:45 -0600
(Wed, 19 Jan 2011) | 9 lines Merged revisions 302663 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r302663 | tilghman | 2011-01-19 15:20:28 -0600 (Wed, 19
Jan 2011) | 2 lines Add some API documentation ........
................
* main/app.c, /: Merged revisions 302599 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r302599 | tilghman | 2011-01-19 14:13:24 -0600 (Wed, 19 Jan 2011)
| 15 lines Kill zombies. When we ast_safe_fork() with a non-zero
argument, we're expected to reap our own zombies. On a zero
argument, however, the zombies are only reaped when there aren't
any non-zero forked children alive. At other times, we accumulate
zombies. This code is forward ported from res_agi in 1.4, so that
forked children are always reaped, thus preventing an
accumulation of zombie processes. (closes issue #18515) Reported
by: ernied Patches: 20101221__issue18515.diff.txt uploaded by
tilghman (license 14) Tested by: ernied ........
2011-01-19 20:14 +0000 [r302600] Jason Parker <jparker@digium.com>
* res/res_fax.c: Fix typo pointed out on asterisk-users list.
2011-01-19 19:03 +0000 [r302505-302555] Sean Bright <sean@malleable.com>
* main/utils.c, /: Merged revisions 302554 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r302554 | seanbright | 2011-01-19 14:02:29 -0500 (Wed, 19 Jan
2011) | 7 lines Don't call strlen() when we only need to look at
the next character or two. (closes issue #18042) Reported by:
wdoekes Patches: astsvn-inefficient-ast-uri-decode.patch uploaded
by wdoekes (license 717) ........
* /, main/features.c: Merged revisions 302551 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r302551 | seanbright | 2011-01-19 13:54:03 -0500 (Wed, 19 Jan
2011) | 7 lines Remove an extraneous \r\n at the end of a parking
manager events. (closes issue #18363) Reported by:
clegall_proformatique Patches:
asterisk_1.8_295998_parking_manager_events_format.patch uploaded
by clegall proformatique (license 1139) ........
* /, res/res_agi.c: Merged revisions 302548 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r302548 | seanbright | 2011-01-19 13:37:09 -0500 (Wed, 19 Jan
2011) | 10 lines Properly handle partial reads from fgets() when
handling AGIs. When fgets() failed with EAGAIN, we were
continually decrementing the available space left in our buffer,
resulting in botched command handling. (closes issue #16032)
Reported by: notahat Patches: agi_buffer_patch2.diff uploaded by
fnordian (license 110) ........
* main/utils.c, /: Merged revisions 302504 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r302504 | seanbright | 2011-01-19 12:56:32 -0500 (Wed, 19 Jan
2011) | 7 lines Make sure that h_length is set when we
short-circuit out of ast_gethostbyname. (closes issue #16135)
Reported by: thedavidfactor Patches: utils.patch uploaded by
thedavidfactor (license 903) ........
2011-01-19 17:09 +0000 [r302462] Paul Belanger <pabelanger@digium.com>
* /, res/res_timing_timerfd.c: Merged revisions 302461 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
........ r302461 | pabelanger | 2011-01-19 12:08:01 -0500 (Wed,
19 Jan 2011) | 2 lines Handle 'Resource temporarily unavailable'
error more gracefully. ........
2011-01-19 15:53 +0000 [r302412-302417] Sean Bright <sean@malleable.com>
* configs/extensions.conf.sample, /: Merged revisions 302416 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r302416 | seanbright | 2011-01-19 10:52:44 -0500 (Wed, 19 Jan
2011) | 9 lines Remove references to priorityjumping from the
sample extensions.conf. Priority jumping was removed from
pbx_config in r68970. (closes issue #18622) Reported by: kshumard
Patches: extensions.conf.sample.patch uploaded by kshumard
(license 92) ........
* channels/chan_sip.c: Initialize an uninitialized variable.
(closes issue #18640) Reported by: jcovert Patches:
chan_sip.c.patch uploaded by jcovert (license 551)
* channels/chan_local.c: Use appropriate type for requested format
in chan_local. We were passing and storing the requested format
as an int instead of format_t resulting in truncation. (closes
issue #18238) Reported by: whizemen Patches:
0018238_speex16.patch uploaded by whizemen (license 1143)
2011-01-18 22:04 +0000 [r302318] Richard Mudgett <rmudgett@digium.com>
* main/features.c: Use the expanded format type instead of plain
int.
2011-01-18 21:43 +0000 [r302314] Matthew Nicholson <mnicholson@digium.com>
* /, channels/chan_sip.c: Merged revisions 302313 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r302313 | mnicholson | 2011-01-18 15:40:03 -0600
(Tue, 18 Jan 2011) | 11 lines Merged revisions 302311 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r302311 | mnicholson | 2011-01-18 15:35:03 -0600 (Tue, 18 Jan
2011) | 4 lines URI encode the user part of the contact header.
ABE-2705 ........ ................
2011-01-18 20:19 +0000 [r302267] Russell Bryant <russell@digium.com>
* main/astobj2.c: Don't enable AO2_DEBUG by default if AST_DEVMODE
is on. AO2_DEBUG is not important and is causing a false compiler
warning to be generated on my Ubuntu Natty dev box.
2011-01-18 20:19 +0000 [r302266] Jeff Peeler <jpeeler@digium.com>
* main/pbx.c, /: Merged revisions 302265 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r302265 | jpeeler | 2011-01-18 14:13:52 -0600 (Tue, 18 Jan 2011)
| 27 lines Convert device state callbacks to ao2 objects to fix a
deadlock in chan_sip. Lock scenario presented here: Thread 1
holds ast_rdlock_contexts &conlock holds handle_statechange hints
holds handle_statechange hint waiting for cb_extensionstate
Locked Here: chan_sip.c line 7428 (find_call) Thread 2 holds
handle_request_do &netlock holds find_call sip_pvt_ptr waiting
for ast_rdlock_contexts &conlock Locked Here: pbx.c line 9911
(ast_rdlock_contexts) Chan_sip has an established locking order
of locking the sip_pvt and then getting the context lock. So the
as stated by the summary, the operations in thread 2 have been
modified to no longer require the context lock. (closes issue
#18310) Reported by: one47 Patches: statecbs_ao2.mk2.patch
uploaded by one47 (license 23), modified by me Review:
https://reviewboard.asterisk.org/r/1072/ ........
2011-01-18 18:11 +0000 [r302174] Richard Mudgett <rmudgett@digium.com>
* /, main/features.c: Merged revisions 302173 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r302173 | rmudgett | 2011-01-18 12:07:15 -0600
(Tue, 18 Jan 2011) | 95 lines Merged revisions 302172 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r302172 | rmudgett | 2011-01-18 12:04:36 -0600 (Tue, 18 Jan 2011)
| 88 lines Issues with DTMF triggered attended transfers. Issue
#17999 1) A calls B. B answers. 2) B using DTMF dial *2 (code in
features.conf for attended transfer). 3) A hears MOH. B dial
number C 4) C ringing. A hears MOH. 5) B hangup. A still hears
MOH. C ringing. 6) A hangup. C still ringing until
"atxfernoanswertimeout" expires. For v1.4 C will ring forever
until C answers the dead line. (Issue #17096) Problem: When A and
B hangup, C is still ringing. Issue #18395 SIP call limit of B is
1 1. A call B, B answered 2. B *2(atxfer) call C 3. B hangup, C
ringing 4. Timeout waiting for C to answer 5. Recall to B fails
because B has reached its call limit. Because B reached its call
limit, it cannot do anything until the transfer it started
completes. Issue #17273 Same scenario as issue 18395 but party B
is an FXS port. Party B cannot do anything until the transfer it
started completes. If B goes back off hook before C answers, B
hears ringback instead of the expected dialtone. ********** Note
for the issue #17273 and #18395 fix: DTMF attended transfer works
within the channel bridge. Unfortunately, when either party A or
B in the channel bridge hangs up, that channel is not completely
hung up until the transfer completes. This is a real problem
depending upon the channel technology involved. For chan_dahdi,
the channel is crippled until the hangup is complete. Either the
channel is not useable (analog) or the protocol disconnect
messages are held up (PRI/BRI/SS7) and the media is not released.
For chan_sip, a call limit of one is going to block that endpoint
from any further calls until the hangup is complete. For party A
this is a minor problem. The party A channel will only be in this
condition while party B is dialing and when party B and C are
conferring. The conversation between party B and C is expected to
be a short one. Party B is either asking a question of party C or
announcing party A. Also party A does not have much incentive to
hangup at this point. For party B this can be a major problem
during a blonde transfer. (A blonde transfer is our term for an
attended transfer that is converted into a blind transfer. :))
Party B could be the operator. When party B hangs up, he assumes
that he is out of the original call entirely. The party B channel
will be in this condition while party C is ringing, while
attempting to recall party B, and while waiting between call
attempts. WARNING: The ATXFER_NULL_TECH conditional is a hack to
fix the problem. It will replace the party B channel technology
with a NULL channel driver to complete hanging up the party B
channel technology. The consequences of this code is that the 'h'
extension will not be able to access any channel technology
specific information like SIP statistics for the call.
ATXFER_NULL_TECH is not defined by default. ********** (closes
issue #17999) Reported by: iskatel Tested by: rmudgett JIRA
SWP-2246 (closes issue #17096) Reported by: gelo Tested by:
rmudgett JIRA SWP-1192 (closes issue #18395) Reported by:
shihchuan Tested by: rmudgett (closes issue #17273) Reported by:
grecco Tested by: rmudgett Review:
https://reviewboard.asterisk.org/r/1047/ ........
................
2011-02-22 Leif Madsen <lmadsen@digium.com>
* Asterisk 1.8.3 Released.
* Merged changes related to AST-2011-002
2011-02-16 Leif Madsen <lmadsen@digium.com>
* Asterisk 1.8.3-rc3 Released.
------------------------------------------------------------------------
r301790 | jpeeler | 2011-01-14 11:32:53 -0600 (Fri, 14 Jan 2011) | 42 lines
Resolve deadlock involving REFER.
(closes issue 0018403)
Reported by: jthurman
Patches:
20110103-blind_deadlock.diff uploaded by jthurman (license 614)
issue18403.patch uploaded by jpeeler (license 325)
Tested by: jthurman
------------------------------------------------------------------------
------------------------------------------------------------------------
r308002 | qwell | 2011-02-15 17:32:21 -0600 (Tue, 15 Feb 2011) | 10
lines
Fix regression that changed behavior of queues when ringing a queue
member.
This reverts r298596, which was to fix a highly bizarre and contrived
issue with a queue member that called into his own queue being
transferred back into his own queue. I couldn't reproduce that issue in
any way. I think one of the other recent transfer fixes actually fixed
this.
(closes issue 0018747)
Reported by: vrban
------------------------------------------------------------------------
2011-01-20 Leif Madsen <lmadsen@digium.com>
* Asterisk 1.8.3-rc2 Released.
------------------------------------------------------------------------
r303907 | mnicholson | 2011-01-25 14:56:12 -0600 (Tue, 25 Jan 2011) | 2
lines
Reimplemented fax session reservation to reverse the ABI breakage
introduced in r297486.
------------------------------------------------------------------------
------------------------------------------------------------------------
r303106 | sruffell | 2011-01-20 13:56:35 -0600 (Thu, 20 Jan 2011) | 15
lines
main/features: Use POLLPRI when waiting for events on parked channels.
This change resolves a regression in the 1.6.2 when converting from
select to poll. The DAHDI timers use POLLPRI to indicate that the
timer
fired, but features was not waiting for that flag. The result was no
audio for MOH when a call was parked and res_timing_dahdi was in use.
This patch is slightly modified from the one on the mantis issue. It
does
not set an exception on the channel if the POLLPRI flag is set.
(closes issue 0018262)
Reported by: francesco_r
Patches:
patch_park_moh-trunk-2.txt uploaded by cjacobsen (license 1029)
Tested by: francesco_r, rfrantik, one47
------------------------------------------------------------------------
------------------------------------------------------------------------
r302785 | russell | 2011-01-19 16:35:15 -0600 (Wed, 19 Jan 2011) | 15
lines
Resolve a memory leak with the manager interface is disabled.
The intent of this check as it stands in previous versions of Asterisk
was to
check if there are any active sessions. If there were no sessions,
then the
function would return immediately and not bother with queueing up the
manager
event to be processed. Since the conversion of storing sessions in an
astobj2
container, this check will always pass. I changed it to go back to
checking
what was intended.
The side effect of this was that if the AMI is disabled, the manager
event
queue is populated anyway, but the code that runs to clear out the
queue
never runs. A producer with no consumer is a bad thing.
Reported internally by kmorgan.
------------------------------------------------------------------------
------------------------------------------------------------------------
r302837 | russell | 2011-01-19 17:56:48 -0600 (Wed, 19 Jan 2011) | 2
lines
Only check container count if it exists.
------------------------------------------------------------------------
2011-01-17 Leif Madsen <lmadsen@digium.com>
* Asterisk 1.8.3-rc1 Released.
2011-01-18 18:11 +0000 [r302174] Richard Mudgett <rmudgett@digium.com>
* /, main/features.c: Merged revisions 302173 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r302173 | rmudgett | 2011-01-18 12:07:15 -0600
(Tue, 18 Jan 2011) | 95 lines Merged revisions 302172 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r302172 | rmudgett | 2011-01-18 12:04:36 -0600 (Tue, 18 Jan 2011)
| 88 lines Issues with DTMF triggered attended transfers. Issue
#17999 1) A calls B. B answers. 2) B using DTMF dial *2 (code in
features.conf for attended transfer). 3) A hears MOH. B dial
number C 4) C ringing. A hears MOH. 5) B hangup. A still hears
MOH. C ringing. 6) A hangup. C still ringing until
"atxfernoanswertimeout" expires. For v1.4 C will ring forever
until C answers the dead line. (Issue #17096) Problem: When A and
B hangup, C is still ringing. Issue #18395 SIP call limit of B is
1 1. A call B, B answered 2. B *2(atxfer) call C 3. B hangup, C
ringing 4. Timeout waiting for C to answer 5. Recall to B fails
because B has reached its call limit. Because B reached its call
limit, it cannot do anything until the transfer it started
completes. Issue #17273 Same scenario as issue 18395 but party B
is an FXS port. Party B cannot do anything until the transfer it
started completes. If B goes back off hook before C answers, B
hears ringback instead of the expected dialtone. ********** Note
for the issue #17273 and #18395 fix: DTMF attended transfer works
within the channel bridge. Unfortunately, when either party A or
B in the channel bridge hangs up, that channel is not completely
hung up until the transfer completes. This is a real problem
depending upon the channel technology involved. For chan_dahdi,
the channel is crippled until the hangup is complete. Either the
channel is not useable (analog) or the protocol disconnect
messages are held up (PRI/BRI/SS7) and the media is not released.
For chan_sip, a call limit of one is going to block that endpoint
from any further calls until the hangup is complete. For party A
this is a minor problem. The party A channel will only be in this
condition while party B is dialing and when party B and C are
conferring. The conversation between party B and C is expected to
be a short one. Party B is either asking a question of party C or
announcing party A. Also party A does not have much incentive to
hangup at this point. For party B this can be a major problem
during a blonde transfer. (A blonde transfer is our term for an
attended transfer that is converted into a blind transfer. :))
Party B could be the operator. When party B hangs up, he assumes
that he is out of the original call entirely. The party B channel
will be in this condition while party C is ringing, while
attempting to recall party B, and while waiting between call
attempts. WARNING: The ATXFER_NULL_TECH conditional is a hack to
fix the problem. It will replace the party B channel technology
with a NULL channel driver to complete hanging up the party B
channel technology. The consequences of this code is that the 'h'
extension will not be able to access any channel technology
specific information like SIP statistics for the call.
ATXFER_NULL_TECH is not defined by default. ********** (closes
issue #17999) Reported by: iskatel Tested by: rmudgett JIRA
SWP-2246 (closes issue #17096) Reported by: gelo Tested by:
rmudgett JIRA SWP-1192 (closes issue #18395) Reported by:
shihchuan Tested by: rmudgett (closes issue #17273) Reported by:
grecco Tested by: rmudgett Review:
https://reviewboard.asterisk.org/r/1047/ ........
................
2011-01-17 15:04 +0000 [r302005] Terry Wilson <twilson@digium.com>
* configs/sip.conf.sample: Document "encryption" option in
sip.conf.sample
2011-01-14 21:09 +0000 [r301946] Richard Mudgett <rmudgett@digium.com>
* channels/sig_pri.c: Deadlock between dahdi_request() and
pri_dchannel() processing an incomming call. The
sig_pri_new_ast_channel() is called with the channel private lock
held when pri_dchannel() calls it and no channel private lock
held when dahdi_request() calls it. The use of pri_grab() in
sig_pri_new_ast_channel() could leave the channel private lock
held when it returns if the lock was not held before calling it.
Make sig_pri_new_ast_channel() just lock the PRI span lock
instead of using pri_grab(). It is safe to do this because
dahdi_request() does not have the channel private lock and the
deadlock potential with the PRI span lock is only between
pri_dchannel() and other threads.
2011-01-14 20:11 +0000 [r301851] Brett Bryant <bbryant@digium.com>
* channels/chan_multicast_rtp.c: Changing previous revisions
301845/301847 to use ast_sockaddr_setnull() instead of setting
the field manually to avoid uninitialized data. Review:
https://reviewboard.asterisk.org/r/1076/
2011-01-14 20:05 +0000 [r301849] Andrew Latham <lathama@gmail.com>
* funcs/func_base64.c, /, funcs/func_aes.c: Add relationships to
function documentation. Fix amatuer type mistake
2011-01-14 19:35 +0000 [r301845] Brett Bryant <bbryant@digium.com>
* channels/chan_multicast_rtp.c: Fix for a consistent MulticastRTP
channel driver crash due to use of unitilized data. (closes issue
#18290) (closes issue #18602) Reported by: voipgate, wybecom
Review: https://reviewboard.asterisk.org/r/1076/
2011-01-14 19:35 +0000 [r301844] Andrew Latham <lathama@gmail.com>
* funcs/func_base64.c, /, funcs/func_aes.c: Add relationships to
function documentation.
2011-01-14 17:32 +0000 [r301790] Jeff Peeler <jpeeler@digium.com>
* channels/chan_sip.c: Resolve deadlock involving REFER. Two fixes:
1) One must always have the private unlocked before calling
pbx_builtin_setvar_helper to not invalidate locking order since
it locks the channel. 2) Unlock the channel before calling
pbx_find_extension, which starts and stops autoservice during the
lookup. The problem scenario as illustrated by the reporter:
Thread: do_monitor ----------------------- handle_request_do
handle_incoming handle_request_refer ast_parking_ext_valid
pbx_find_extension ast_autoservice_stop while (chan_list_state ==
as_chan_list_state) { usleep(1000); } Thread: autoservice_run
----------------------- autoservice_run chan = ast_waitfor_n
ast_waitfor_nandfds ast_waitfor_nandfds_classic / simple /
complex (depending on your system) ast_channel_lock(c[x]);
handle_request_do and schedule_process_request_queue locks the
owner if it exists. The autoservice thread is waiting for the
channel lock, which wasn't ever released since the do_monitor
thread was waiting for autoservice operations to complete. Solved
by unlocking the channel but keeping a reference to guarantee
safety. (closes issue #18403) Reported by: jthurman Patches:
20110103-blind_deadlock.diff uploaded by jthurman (license 614)
issue18403.patch uploaded by jpeeler (license 325) Tested by:
jthurman
2011-01-13 17:01 +0000 [r301731] Leif Madsen <lmadsen@digium.com>
* configs/phoneprov.conf.sample, /: Merged revisions 301730 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r301730 | lmadsen | 2011-01-13 11:01:11 -0600 (Thu, 13 Jan 2011)
| 7 lines Add static entry for split Polycom 332 firmware.
(closes issue #18607) Reported by: cjacobsen Patches:
polycom_331.diff uploaded by cjacobsen (license 1029) Tested by:
lathama ........
2011-01-12 21:19 +0000 [r301683] Terry Wilson <twilson@digium.com>
* /, channels/chan_sip.c: Merged revisions 301682 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r301682 | twilson | 2011-01-12 15:05:02 -0600 (Wed, 12 Jan 2011)
| 9 lines Don't reject all SUBSCRIBE auth requests When merging
another SUBSCRIBE fix from 1.4, some braces were put in the wrong
place. This patch fixes that. (closes issue #18597) Reported by:
thsgmbh ........
2011-01-12 18:51 +0000 [r301595] Matthew Nicholson <mnicholson@digium.com>
* main/manager.c, /: Merged revisions 301594 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r301594 | mnicholson | 2011-01-12 12:50:31 -0600
(Wed, 12 Jan 2011) | 15 lines Removed a usleep(1) that shouldn't
be necessary in session_do, and removed the ms_t member from the
mansession_session structure. Merged revisions 301591 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r301591 | mnicholson | 2011-01-12 12:39:03 -0600 (Wed, 12 Jan
2011) | 5 lines Don't store the thread id for the manager session
in the structure we pass to the thread for the manager session.
ABE-2543 ........ ................
2011-01-12 18:12 +0000 [r301504] Jeff Peeler <jpeeler@digium.com>
* main/channel.c, /: Merged revisions 301503 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r301503 | jpeeler | 2011-01-12 12:11:49 -0600
(Wed, 12 Jan 2011) | 19 lines Merged revisions 301502 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r301502 | jpeeler | 2011-01-12 12:10:42 -0600 (Wed, 12 Jan 2011)
| 12 lines Fix CPU spike when pressing DTMF after agent login.
The problem here is that DTMF was being continuously deferred and
requeued since ast_safe_sleep is called in a loop. There are
serveral other places in the code that sleeps and then loops in a
similar fashion. Because of this fact I opted to not defer DTMF
any more, which will not affect the original fix:
https://reviewboard.asterisk.org/r/674 (closes issue #18130)
Reported by: rgj ........ ................
2011-01-12 16:05 +0000 [r301446] David Vossel <dvossel@digium.com>
* main/file.c: Removal of unused variables so Asterisk will
compile.
2011-01-12 15:57 +0000 [r301444] Stefan Schmidt <sst@sil.at>
* Makefile: fix wrong text of rerun menuselect after user interface
warning the warning, if no user interface for menuselect warning
was found is not right. you have to rerun configure before make
menuselect after installing a proper user interface. (closes
issue #18594) Reported by: Dovid
2011-01-12 00:26 +0000 [r301402] Tilghman Lesher <tilghman@meg.abyt.es>
* main/file.c: Call execl() directly for a better solution for
paths with spaces. (closes issue #18600) Reported by: ebroad
Patches: 20110111__issue18600__2.diff.txt uploaded by tilghman
(license 14)
2011-01-11 19:16 +0000 [r301311] Paul Belanger <pabelanger@digium.com>
* configs/extensions.conf.sample, /: Merged revisions 301310 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r301310 | pabelanger | 2011-01-11 14:14:31 -0500 (Tue, 11 Jan
2011) | 2 lines Fix a logic issue when passing context ARG
........
2011-01-11 18:51 +0000 [r301308] Matthew Nicholson <mnicholson@digium.com>
* main/utils.c, /: Merged revisions 301307 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r301307 | mnicholson | 2011-01-11 12:42:05 -0600
(Tue, 11 Jan 2011) | 11 lines Merged revisions 301305 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r301305 | mnicholson | 2011-01-11 12:34:40 -0600 (Tue, 11 Jan
2011) | 4 lines Prevent buffer overflows in ast_uri_encode()
ABE-2705 ........ ................
2011-01-10 22:39 +0000 [r301263] Tilghman Lesher <tilghman@meg.abyt.es>
* main/strcompat.c: Little endian machines were not converted
properly. (closes issue #18583) Reported by: jcovert Patches:
20110110__issue18583.diff.txt uploaded by tilghman (license 14)
Tested by: jcovert
2011-01-09 21:40 +0000 [r301177-301221] Paul Belanger <pabelanger@digium.com>
* autoconf/ast_ext_lib.m4, /, configure, configure.ac: Merged
revisions 301220 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r301220 | pabelanger | 2011-01-09 16:38:24 -0500 (Sun, 09 Jan
2011) | 14 lines SOUND_CACHE_DIR now defaults to empty Sounds
files included in the Asterisk tarball were being ignored and
re-downloaded. Users wanting to cache the files can still
override the setting using the --with-sounds-cache option.
(closes issue #18589) Reported by: pabelanger Patches:
issue18589.patch uploaded by pabelanger (license 224) Tested by:
pabelanger Review: https://reviewboard.asterisk.org/r/1074/
........
* apps/app_verbose.c, /: Merged revisions 301176 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r301176 | pabelanger | 2011-01-08 16:58:24 -0500 (Sat, 08 Jan
2011) | 7 lines Indicate log level argument for Log() is not
optional (closes issue #18586) Reported by: kshumard Patches:
app_verbose.c.patch uploaded by kshumard (license 92) ........
2011-01-08 01:11 +0000 [r301134] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c: The DTMF attended transfer feature cannot
callback a chan_dahdi BRI phone. The DAHDI ISDN channel name is
not dialable. Make a channel name like DAHDI/i3/400-12 dialable
when the sequence number is stripped off of the name.
2011-01-07 20:53 +0000 [r301090] Jason Parker <jparker@digium.com>
* /, apps/app_meetme.c: Merged revisions 301089 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r301089 | qwell | 2011-01-07 14:52:00 -0600 (Fri, 07 Jan 2011) |
8 lines Initialize useropts/adminopts in case there is no column
in the realtime DB. (closes issue #18182) Reported by: dimas
Patches: v1-18182.patch uploaded by dimas (license 88) Tested by:
dimas ........
2011-01-07 19:58 +0000 [r300955-301047] Jeff Peeler <jpeeler@digium.com>
* apps/app_voicemail.c, /: Merged revisions 301046 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
........ r301046 | jpeeler | 2011-01-07 13:57:42 -0600 (Fri, 07
Jan 2011) | 8 lines Fix regression causing forwarding voicemails
to not work with file storage. I had actually already fixed this
in 295200 in 1.4 and thought it wasn't missing in the other
branches for some reason. (closes issue #18358) Reported by:
cabal95 ........
* apps/app_voicemail.c, /: Merged revisions 300951 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r300951 | jpeeler | 2011-01-07 11:23:37 -0600
(Fri, 07 Jan 2011) | 14 lines Merged revisions 300918 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r300918 | jpeeler | 2011-01-07 11:13:21 -0600 (Fri, 07 Jan 2011)
| 7 lines Ensure good bye prompt in voicemail is played at the
correct time. Specifically in the case of timing out but not
leaving voicemail nothing should be heard. And when leaving
voicemail it should be heard. ABE-2647 ........ ................
2011-01-06 06:28 +0000 [r300798] Tilghman Lesher <tilghman@meg.abyt.es>
* addons/res_config_mysql.c: Don't destroy handle not created by
use (because the caller will). (closes issue #18526) Reported by:
makoto Patches: res-config-mysql-include.patch uploaded by makoto
(license 38) Tested by: makoto
2011-01-05 20:54 +0000 [r300714] Richard Mudgett <rmudgett@digium.com>
* channels/sig_pri.c: Merged revision 300711 from
https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
.......... r300711 | rmudgett | 2011-01-05 13:43:55 -0600 (Wed,
05 Jan 2011) | 14 lines A call retrieved from hold may wind up
with no audio. If the retrieved call is natively bridged then the
call may not have any audio path. The following warning message
is given: "Failed to add <dfd> to conference <chan>/<chan>:
Invalid argument". * Open the media on a B channel when
pri_fixup_principle() moves the call from a no_b_channel channel
to a real channel. * Added lock protection while
pri_fixup_principle() moves a call from one private structure to
another. * Made some pri_fixup_principle() messages more
meaningful. ..........
2011-01-05 18:56 +0000 [r300623] Tilghman Lesher <tilghman@meg.abyt.es>
* res/res_odbc.c, /: Merged revisions 300622 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r300622 | tilghman | 2011-01-05 12:54:58 -0600
(Wed, 05 Jan 2011) | 17 lines Merged revisions 300621 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r300621 | tilghman | 2011-01-05 12:47:46 -0600 (Wed, 05 Jan 2011)
| 10 lines Use the sanity check in place of the
disconnect/connect cycle. The disconnect/connect cycle has the
potential to cause random crashes. (closes issue #18243) Reported
by: ks3 Patches: res_odbc.patch uploaded by ks3 (license 1147)
Tested by: ks3 ........ ................
2011-01-05 16:29 +0000 [r300575] Paul Belanger <pabelanger@digium.com>
* /, cdr/cdr_sqlite.c: Merged revisions 300574 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r300574 | pabelanger | 2011-01-05 11:28:07 -0500 (Wed, 05 Jan
2011) | 6 lines Change deprecated message to LOG_WARNING Also
removed latter part of message Discussed on #asterisk-dev
........
2011-01-04 21:53 +0000 [r300433-300521] Leif Madsen <lmadsen@digium.com>
* channels/chan_iax2.c, main/xmldoc.c, /, channels/chan_sip.c,
channels/chan_agent.c: Merged revisions 300520 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r300520 | lmadsen | 2011-01-04 15:52:41 -0600 (Tue, 04 Jan 2011)
| 9 lines Fix backwards and broken XML documentation. (closes
issue #18547) Reported by: jcovert Patches: xmldoc.c.patch
uploaded by jcovert (license 551) chan_iax2.c.doc.patch uploaded
by jcovert (license 551) chan_sip.c.patch uploaded by jcovert
(license 551) chan_agent.c.patch uploaded by jcovert (license
551) ........
* configs/users.conf.sample, /: Merged revisions 300431 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r300431 | lmadsen | 2011-01-04 15:00:29 -0600 (Tue, 04 Jan 2011)
| 7 lines Add some documentation to users.conf.sample. (closes
issue #18531) Reported by: lathama Patches:
users.conf.sample2.diff uploaded by lathama (license 1028) Tested
by: lathama ........
2011-01-04 21:00 +0000 [r300430] Russell Bryant <russell@digium.com>
* contrib/scripts/autosupport, /, contrib/scripts/autosupport.8:
Merged revisions 300429 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r300429 | russell | 2011-01-04 14:59:56 -0600
(Tue, 04 Jan 2011) | 11 lines Merged revisions 300428 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r300428 | russell | 2011-01-04 14:56:04 -0600 (Tue, 04 Jan 2011)
| 4 lines Update the autosupport script from Digium support.
(closes AST-395) ........ ................
2011-01-04 19:45 +0000 [r300384] Leif Madsen <lmadsen@digium.com>
* phoneprov/000000000000.cfg: Update STAT() to use the comma
instead of the pipe. (closes issue #18503) Reported by: cjacobsen
Patches: old_separator.diff uploaded by cjacobsen (license 1029)
Tested by: lathama
2011-01-04 17:54 +0000 [r300301] Terry Wilson <twilson@digium.com>
* /, channels/chan_sip.c: Merged revisions 300298 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r300298 | twilson | 2011-01-04 11:37:26 -0600
(Tue, 04 Jan 2011) | 22 lines Merged revisions 300216 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r300216 | twilson | 2011-01-04 11:11:48 -0600 (Tue, 04 Jan 2011)
| 15 lines Don't authenticate SUBSCRIBE re-transmissions This
only skips authentication on retransmissions that are already
authenticated. A similar method is already used for INVITES. This
is the kind of thing we end up having to do when we don't have a
transaction layer... (closes issue #18075) Reported by: mdu113
Patches: diff.txt uploaded by twilson (license 396) Tested by:
twilson, mdu113 Review: https://reviewboard.asterisk.org/r/1005/
........ ................
2011-01-04 17:01 +0000 [r300214] Jan Kalab <pitlicek@gmail.com>
* res/res_calendar_exchange.c, res/res_calendar_icalendar.c: Memory
leaking in calendars ne_request_destroy() was missing in
icalendar and exchange calendar modules, causing memory leak.
(closes issue #18521) Review:
https://reviewboard.asterisk.org/r/1068/
2011-01-03 23:14 +0000 [r300166] Richard Mudgett <rmudgett@digium.com>
* /, main/features.c: Merged revisions 300165 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r300165 | rmudgett | 2011-01-03 17:02:13 -0600 (Mon, 03 Jan 2011)
| 4 lines Use correct variable for atxfercallbackretries config
option. * Misc formatting changes. ........
2011-01-03 13:14 +0000 [r300082] Leif Madsen <lmadsen@digium.com>
* pbx/pbx_dundi.c: Increase side of mapping response field. I've
increased the size of the response field in a DUNDi mapping
because of some documentation I'm writing. Previously it was set
to AST_MAX_EXTENSION which is only 80 characters, which is far
too small when you're using some dialplan functions to craft a
response. The example I'm using is: extensions =>
RegisteredDevices,0,SIP,dundi:very_awesome_password/${IF($[${DB_EXISTS(phones/${NUMBER}/device)}]?${DB(phones/${NUMBER}/device)}:None)},nopartial
2010-12-29 22:02 +0000 [r299989] Tilghman Lesher <tilghman@meg.abyt.es>
* apps/app_voicemail.c, main/file.c: Quote arguments, just in case
there's a space in a pathname. (Diagnosed by pabelanger on
#asterisk-dev, fixed by me.)
2010-12-29 19:28 +0000 [r299865-299948] Paul Belanger <pabelanger@digium.com>
* sounds/Makefile: Only remove /tmp/astdatadir, not
/var/lib/asterisk
* build_tools/make_sample_voicemail, sounds/Makefile, Makefile:
Properly quote varibles for MAC OS X
* apps/app_chanspy.c, /: Merged revisions 299864 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r299864 | pabelanger | 2010-12-28 13:51:13 -0500 (Tue, 28 Dec
2010) | 2 lines Documentation typo ........
2010-12-27 21:23 +0000 [r299752-299820] Tilghman Lesher <tilghman@meg.abyt.es>
* sounds/Makefile: More space-in-pathname issues.
* sounds/Makefile, Makefile, Makefile.moddir_rules: Mac OS X
spaces-in-pathnames fix.
* configure: Regen configure
* configure.ac: Properly quote path on Darwin.
2010-12-25 16:12 +0000 [r299711] Alexandr Anikin <may@telecom-service.ru>
* addons/ooh323c/src/oochannels.c, addons/ooh323c/src/ooq931.c,
addons/ooh323c/src/ooh323.c, addons/ooh323c/src/ooh245.c: Change
order of sending TCS and MSD packets Change order of sending
Terminal Capability Set and MasterSlave Determination packets,
MSD send when TCS exchange procedure is done (we send tcs ack to
remote and we have remote tcs ack already or we receive tcs ack
from remote and we have send our tcs ack to remote already). Some
endpoints can work in this sequence only, i suggest they can't
work with both (tcs and msd) exchange procedures simultaneously.
Also changed StartH245 facility message sending. It send on
incoming calls only due to some endpoints can't proccess properly
this facility messages on their incoming calls. (issue #18433)
Reported by: MrHanMan Patches: tcs-msd-h245-3.patch uploaded by
may213 (license 454) Tested by: MrHanMan, may213
2010-12-25 10:07 +0000 [r299583-299626] Tilghman Lesher <tilghman@meg.abyt.es>
* channels/chan_local.c, /: Merged revisions 299625 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r299625 | tilghman | 2010-12-25 04:05:00 -0600
(Sat, 25 Dec 2010) | 12 lines Merged revisions 299624 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r299624 | tilghman | 2010-12-25 04:04:06 -0600 (Sat, 25 Dec 2010)
| 5 lines Move check for extension existence below variable
inheritance, due to the possible use of an eswitch. (closes issue
#16228) Reported by: jlaguilar ........ ................
* addons/res_config_mysql.c: Reset 'first' variable after usage.
(closes issue #18525) Reported by: makoto Patches:
res-config-mysql-update2.patch uploaded by makoto (license 38)
2010-12-23 02:53 +0000 [r299531] Moises Silva <moises.silva@gmail.com>
* channels/chan_dahdi.c, /: Merged revisions 299530 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
........ r299530 | moy | 2010-12-22 21:28:37 -0500 (Wed, 22 Dec
2010) | 7 lines Enqueue AST_CONTROL_PROGRESS after
AST_CONTROL_RINGING when MFC-R2 calls are accepted (closes issue
#18438) Reported by: mariner7 Tested by: moy ........
2010-12-22 20:05 +0000 [r299449] Tilghman Lesher <tilghman@meg.abyt.es>
* pbx/ael/ael-test/ref.ael-test19,
pbx/ael/ael-test/ref.ael-vtest13, res/ael/pval.c,
pbx/ael/ael-test/ref.ael-vtest25,
pbx/ael/ael-test/ref.ael-vtest17, /,
pbx/ael/ael-test/ref.ael-test3: Merged revisions 299448 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r299448 | tilghman | 2010-12-22 14:03:30 -0600 (Wed, 22 Dec 2010)
| 8 lines Resolve warnings by disambiguating the "s" extension as
used by chan_dahdi from the "s" extension as used by the AEL
macros. (closes issue #18480) Reported by: nivek Patches:
20101215__issue18480__2.diff.txt uploaded by tilghman (license
14) Tested by: nivek ........
2010-12-22 02:10 +0000 [r299405] Richard Mudgett <rmudgett@digium.com>
* channels/sig_pri.c: Chan_dahdi sends an empty COLP on the bridged
channel. Chan_dahdi always inserts a connected party IE when you
call from one dahdi channel to another dahdi channel, even if no
such information was received on the 2nd channel. This clears the
display of many phones. * Removed leftover artifact from before
the valid flag was added. * Updated all of the channel's caller
id information with the new connected line information instead of
just the string parts. (closes issue #18508) Reported by: wimpy
Patches: issue18508_trunk.patch uploaded by rmudgett (license
664) Tested by: wimpy, rmudgett
2010-12-21 15:25 +0000 [r299353] Matthew Nicholson <mnicholson@digium.com>
* /, channels/chan_sip.c: Merged revisions 299242 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r299242 | mnicholson | 2010-12-20 15:25:35 -0600
(Mon, 20 Dec 2010) | 23 lines Merged revisions
299194,299198,299220 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r299194 | mnicholson | 2010-12-20 14:45:38 -0600 (Mon, 20 Dec
2010) | 6 lines Respond as soon as possible with a 202 Accepted
to refer requests. This change also plugs a few memory leaks that
can occur when parking sip calls. ABE-2656 ........ r299198 |
mnicholson | 2010-12-20 15:00:44 -0600 (Mon, 20 Dec 2010) | 2
lines Remove changes to via processing that were not supposed to
go into the last commit. ........ r299220 | mnicholson |
2010-12-20 15:21:39 -0600 (Mon, 20 Dec 2010) | 4 lines Use
ast_free() instead of free() ABE-2656 ........ ................
2010-12-21 00:44 +0000 [r299312] Paul Belanger <pabelanger@digium.com>
* configs/cel.conf.sample: Correct typo with USER_DEFINED event.
(closes issue #18461) Reported by: joscas Patches:
cel.conf.sample.diff uploaded by lathama (license 1028) Tested
by: lathama, joscas
2010-12-20 21:38 +0000 [r299248] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Fix a couple of CCSS issues. * Make sure to
allocate a cc_params structure when creating autopeers. * Use
sip_uri_cmp when retrieving SIP CC agents and monitors in case
parameters appear in the URI. (closes issue #18504) Reported by:
kkm (closes issue #18338) Reported by: GeorgeKonopacki Patches:
18338.diff uploaded by mmichelson (license 60) Tested by:
GeorgeKonopacki
2010-12-20 18:17 +0000 [r299131-299138] Tilghman Lesher <tilghman@meg.abyt.es>
* sample.call, /: Merged revisions 299136 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r299136 | tilghman | 2010-12-20 12:16:37 -0600 (Mon, 20 Dec 2010)
| 2 lines Documentation fix ........
* cdr/cdr_pgsql.c, /: Merged revisions 299130 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r299130 | tilghman | 2010-12-20 11:41:24 -0600 (Mon, 20 Dec 2010)
| 11 lines If a call was not answered, then the billsec was
calculated unusually large. Also, due to a copy and paste error,
a request for the answer field would have given the start value,
instead. (closes issue #18460) Reported by: joscas Patches:
20101215__issue18460.diff.txt uploaded by tilghman (license 14)
Tested by: joscas ........
2010-12-20 16:18 +0000 [r299088] Leif Madsen <lmadsen@digium.com>
* /, main/features.c: Merged revisions 299087 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r299087 | lmadsen | 2010-12-20 10:18:03 -0600 (Mon, 20 Dec 2010)
| 5 lines Note that Park() timeout is milliseconds. (closes issue
#15758) Reported by: mmurdock Tested by: mmurdock, seanbright
........
2010-12-20 09:14 +0000 [r299004] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
* main/aoc.c, channels/sig_pri.h, channels/chan_sip.c: Typos:
recieved => received
2010-12-18 00:09 +0000 [r298818-298963] Tilghman Lesher <tilghman@meg.abyt.es>
* /, main/say.c: Merged revisions 298962 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r298962 | tilghman | 2010-12-17 18:08:57 -0600 (Fri, 17 Dec 2010)
| 2 lines Remove backtrace used for testing merge process
........
* main/utils.c, main/astobj2.c, utils/conf2ael.c,
include/asterisk/logger.h, configure,
build_tools/menuselect-deps.in, main/logger.c, utils/ael_main.c,
utils/hashtest2.c, makeopts.in, utils/check_expr.c,
utils/refcounter.c, include/asterisk/utils.h,
build_tools/cflags-devmode.xml, /,
include/asterisk/autoconfig.h.in, main/Makefile, main/say.c,
configure.ac, utils/hashtest.c: Merged revisions 298957 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r298957 | tilghman | 2010-12-17 17:30:55 -0600
(Fri, 17 Dec 2010) | 13 lines Merged revisions 298905 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r298905 | tilghman | 2010-12-17 15:40:56 -0600 (Fri, 17 Dec 2010)
| 6 lines Let Asterisk find better backtrace information with
libbfd. The menuselect option BETTER_BACKTRACES, if enabled, will
use libbfd to search for better symbol information within both
the Asterisk binary, as well as loaded modules, to assist when
using inline backtraces to track down problems. ........
................
* contrib/init.d/rc.debian.asterisk: -v implies -f, so override
with -F. (closes issue #18446) Reported by: lathama Patches:
rc.debian.asterisk.diff uploaded by lathama (license 1028) Tested
by: lathama
* /, configure, configure.ac: Merged revisions 298817 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
........ r298817 | tilghman | 2010-12-17 15:03:06 -0600 (Fri, 17
Dec 2010) | 8 lines Also include PTHREAD_LIBS and PTHREAD_CFLAGS
for SQLite 3, as it's needed on some platforms. (closes issue
#18493) Reported by: pprindeville Patches:
asterisk-1.8-sqlite3.patch uploaded by pprindeville (license 347)
Tested by: pprindeville ........
2010-12-17 17:26 +0000 [r298773] Brad Watkins <Marquis42@gmail.com>
* configs/sip.conf.sample, channels/chan_sip.c: Fix parsing of mwi
=> lines in sip.conf Reworking parsing of mwi => lines to resolve
a segfault. Also add a set of unit tests for the function that
does the parsing. (closes issue #18350) Reported by: gbour Tested
by: Marquis, gbour Review:
https://reviewboard.asterisk.org/r/1053/
2010-12-16 23:31 +0000 [r298598-298685] Jeff Peeler <jpeeler@digium.com>
* apps/app_voicemail.c, /: Merged revisions 298684 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r298684 | jpeeler | 2010-12-16 17:30:59 -0600
(Thu, 16 Dec 2010) | 9 lines Merged revisions 298683 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r298683 | jpeeler | 2010-12-16 17:29:30 -0600 (Thu, 16
Dec 2010) | 2 lines After recording only silence for a voicemail
prepending, restore backup files. ........ ................
* apps/app_queue.c, /: Merged revisions 298597 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r298597 | jpeeler | 2010-12-16 14:49:33 -0600
(Thu, 16 Dec 2010) | 14 lines Merged revisions 298596 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r298596 | jpeeler | 2010-12-16 14:46:52 -0600 (Thu, 16 Dec 2010)
| 7 lines Fix improper hangup when doing an attended transfer to
queue. Had to indicate ringing in wait_for_answer so the attended
transfer code would not try and hang up the local channel it
created, which would kill the call. ABE-2624 ........
................
2010-12-16 09:28 +0000 [r298394-298539] Tilghman Lesher <tilghman@meg.abyt.es>
* channels/chan_sip.c: Ensure the ipaddr field in realtime is large
enough to handle IPv6 addresses. (closes issue #18464) Reported
by: IgorG Patches: realtime_ipv6store.diff uploaded by IgorG
(license 20) (plus a few additional lines by tilghman)
* res/res_config_odbc.c, /: Merged revisions 298481 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r298481 | tilghman | 2010-12-16 03:04:38 -0600
(Thu, 16 Dec 2010) | 21 lines Merged revisions 298480 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r298480 | tilghman | 2010-12-16 03:03:40 -0600 (Thu, 16 Dec 2010)
| 14 lines Only increment the pointer once per loop, otherwise we
corrupt the value. (closes issue #18251) Reported by: bcnit
Patches: 20101110__issue18251.diff.txt uploaded by tilghman
(license 14) Tested by: trev, jthurman, elguero (closes issue
#18279) Reported by: zerohalo Patches:
20101109__issue18279.diff.txt uploaded by tilghman (license 14)
Tested by: zerohalo ........ ................
* /, funcs/func_dialgroup.c: Merged revisions 298477 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
........ r298477 | tilghman | 2010-12-16 02:54:23 -0600 (Thu, 16
Dec 2010) | 8 lines Eliminate duplicates from container. (closes
issue #18091) Reported by: bunny Patches:
20101006__issue18091.diff.txt uploaded by tilghman (license 14)
Tested by: bunny ........
* /, cdr/cdr_sqlite.c: Merged revisions 298393 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r298393 | tilghman | 2010-12-15 18:29:10 -0600
(Wed, 15 Dec 2010) | 15 lines Merged revisions 298392 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r298392 | tilghman | 2010-12-15 18:28:04 -0600 (Wed, 15 Dec 2010)
| 8 lines Unregister before shutting down the connection, to
avoid a race. (closes issue #18481) Reported by: pabelanger
Patches: 20101215__issue18481.diff.txt uploaded by tilghman
(license 14) Tested by: pabelanger ........ ................
2010-12-13 17:11 +0000 [r298195] Richard Mudgett <rmudgett@digium.com>
* channels/sig_pri.c, channels/chan_dahdi.c, /: Merged revisions
298194 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r298194 | rmudgett | 2010-12-13 11:04:41 -0600
(Mon, 13 Dec 2010) | 26 lines Merged revisions 298193 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r298193 | rmudgett | 2010-12-13 10:56:07 -0600 (Mon, 13 Dec 2010)
| 19 lines Outgoing PRI/BRI calls cannot do DTMF triggered
transfers. Outgoing PRI/BRI calls cannot do DTMF triggered
transfers if a PROCEEDING message is not received. The debug
output shows that the DTMF begin event is seen, but the DTMF end
event is missing. When the DTMF begin happens, the call is muted
so we now have one way audio (until a DTMF end event is somehow
seen). * Made set the proceeding flag when the PRI_EVENT_ANSWER
event is received. * Made absorb the DTMF begin and DTMF end
events if we are overlap dialing and have not seen a PROCEEDING
message. * Added a debug message when absorbing a DTMF event.
JIRA SWP-2690 JIRA ABE-2697 ........ ................
2011-01-12 Leif Madsen <lmadsen@digium.com>
* Asterisk 1.8.2 Released.
* Merge in a change in the configure script to fix an issue for
Debian packagers.
------------------------------------------------------------------------
r301221 | pabelanger | 2011-01-09 15:40:35 -0600 (Sun, 09 Jan 2011)
| 21 lines
Merged revisions 301220 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 [^]
........
r301220 | pabelanger | 2011-01-09 16:38:24 -0500 (Sun, 09 Jan
2011) | 14 lines
SOUND_CACHE_DIR now defaults to empty
Sounds files included in the Asterisk tarball were being
ignored and
re-downloaded. Users wanting to cache the files can
still override the setting
using the --with-sounds-cache option.
(closes issue 0018589)
Reported by: pabelanger
Patches:
issue18589.patch uploaded by
pabelanger (license 224)
Tested by: pabelanger
Review:
https://reviewboard.asterisk.org/r/1074/
------------------------------------------------------------------------
2010-12-13 Leif Madsen <lmadsen@digium.com>
* Asterisk 1.8.2-rc1 Released.
2010-12-11 21:45 +0000 [r298099] Alexandr Anikin <may@telecom-service.ru>
* addons/ooh323c/src/ooGkClient.c: Correction to work with
gatekeeper which don't send GK ID Don't use GK ID if it's not
presented in GK replies Extract GK ID not only in GK confirm but
in GK register confirm also (issue #18401) Reported by: MrHanMan
Patches: no-gkid-2.patch uploaded by may213 (license 454) Tested
by: may213, MrHanMan
2010-12-10 16:52 +0000 [r298054] Matthew Nicholson <mnicholson@digium.com>
* res/res_fax.c: Prevent a memcpy overlap in
GENERIC_FAX_EXEC_SET_VARS
2010-12-10 16:26 +0000 [r298051] Tilghman Lesher <tlesher@digium.com>
* main/netsock.c, /, configure, include/asterisk/autoconfig.h.in,
configure.ac: Merged revisions 298050 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r298050 | tilghman | 2010-12-10 10:24:13 -0600 (Fri, 10 Dec 2010)
| 11 lines Portability issue on OpenSolaris. Also detect the
required structure element, because OpenSolaris defines
SIOCGIFHWADDR, but without support for IP sockets. (closes issue
#18442) Reported by: ranjtech Patches:
20101209__issue18442.diff.txt uploaded by tilghman (license 14)
Tested by: ranjtech ........
2010-12-09 22:18 +0000 [r297965] Terry Wilson <twilson@digium.com>
* /, channels/chan_sip.c: Merged revisions 297960 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r297960 | twilson | 2010-12-09 16:10:31 -0600
(Thu, 09 Dec 2010) | 21 lines Merged revisions 297959 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r297959 | twilson | 2010-12-09 16:00:30 -0600 (Thu, 09 Dec 2010)
| 14 lines Ignore spurious REGISTER requests If a REGISTER
request with a Call-ID matching an existing transaction is
received it was possible that the REGISTER request would
overwrite the initreq of the private structure. This info is used
to generate messages for other responses in the transaction. This
patch ignores REGISTER requests that match non-REGISTER
transactions. (closes issue #18051) Reported by: eeman Tested by:
twilson Review: https://reviewboard.asterisk.org/r/1050/ ........
................
2010-12-09 21:32 +0000 [r297957] David Vossel <dvossel@digium.com>
* channels/chan_gtalk.c: Fixes issue with outbound google voice
calls not working. Thanks to az1234 and nevermind_quack for their
input in helping debug the issue. (closes issue #18412) Reported
by: nevermind_quack Patches: fix uploaded by dvossel (license
671)
2010-12-09 20:48 +0000 [r297952] Terry Wilson <twilson@digium.com>
* main/features.c: Don't crash after Set(CDR(userfield)=...) in
ast_bridge_call Instead of setting peer->cdr = NULL, set it to
not post. (closes issue #18415) Reported by: macbrody Patches:
patch-18415 uploaded by jsolares (license 1167) Tested by:
jsolares, twilson
2010-12-08 18:06 +0000 [r297909] Tilghman Lesher <tlesher@digium.com>
* configs/extensions.conf.sample, /: Merged revisions 297908 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r297908 | tilghman | 2010-12-08 12:04:38 -0600 (Wed, 08 Dec 2010)
| 4 lines Use inheritance to get correct results for
SIPFROMDOMAIN. (from an internal Digium discussion) ........
2010-12-08 16:12 +0000 [r297905] Matthew Nicholson <mnicholson@digium.com>
* res/res_fax.c: Display the capabilities requested when requesting
a fax session fails instead of displaying a hex value. Tweak the
way fax stats are calculated so that all fax attempts and
faliures are logged. Also make ensure faxes are either counted as
completed or falied and never both. FAX-210
2010-12-07 22:59 +0000 [r297825] Jeff Peeler <jpeeler@digium.com>
* main/channel.c, /: Merged revisions 297824 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r297824 | jpeeler | 2010-12-07 16:58:54 -0600
(Tue, 07 Dec 2010) | 19 lines Merged revisions 297823 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r297823 | jpeeler | 2010-12-07 16:57:48 -0600 (Tue, 07 Dec 2010)
| 12 lines Revert code that changed SSRC for DTMF. Some previous
behavior was attempted to be restored, but mistakingly I did not
realize that the previous behavior was incorrect. This fixes DTMF
not being detected since DTMF shouldn't cause the SSRC to change.
(related to issue #17404) (closes issue #18189) (closes issue
#18352) Reported by: marcbou Tested by: cmbaker82 ........
................
2010-12-07 22:51 +0000 [r297733-297821] Tilghman Lesher <tlesher@digium.com>
* contrib/init.d/org.asterisk.muted.plist (added), Makefile,
contrib/init.d/org.asterisk.asterisk.plist, utils/muted.c, /:
Merged revisions 297819 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r297819 | tilghman | 2010-12-07 16:40:45 -0600
(Tue, 07 Dec 2010) | 11 lines Merged revisions 297818 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r297818 | tilghman | 2010-12-07 16:35:50 -0600 (Tue, 07 Dec 2010)
| 4 lines Use non-deprecated APIs for CoreAudio Review:
https://reviewboard.asterisk.org/r/1040/ ........
................
* apps/app_followme.c, /: Merged revisions 297713 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r297713 | tilghman | 2010-12-06 18:21:50 -0600
(Mon, 06 Dec 2010) | 15 lines Merged revisions 297689 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r297689 | tilghman | 2010-12-06 18:07:37 -0600 (Mon, 06 Dec 2010)
| 8 lines Don't create a Local channel if the target extension
does not exist. (closes issue #18126) Reported by: junky Patches:
followme.diff uploaded by junky (license 177) (partially
restructured by me to avoid a possible memory leak) ........
................
2010-12-06 22:06 +0000 [r297607] Jeff Peeler <jpeeler@digium.com>
* /, channels/chan_sip.c: Merged revisions 297605 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r297605 | jpeeler | 2010-12-06 16:03:04 -0600
(Mon, 06 Dec 2010) | 18 lines Merged revisions 297603 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r297603 | jpeeler | 2010-12-06 15:57:15 -0600 (Mon, 06 Dec 2010)
| 12 lines Improve handling of REGISTER requests with multiple
contact headers. The changes here attempt to more strictly follow
RFC 3261 section 10.3. Basically the following will now cause a
400 Bad Response to be returned, if: - multiple Contact headers
are present with one set to expire all bindings ("*") - wildcard
parameter is specified for Contact without Expires header or
Expires header is not set to zero. ABE-2442 ABE-2443 ........
................
2010-12-03 17:41 +0000 [r297535] Sean Bright <sean@malleable.com>
* channels/chan_console.c, /: Merged revisions 297534 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
........ r297534 | seanbright | 2010-12-03 12:40:52 -0500 (Fri,
03 Dec 2010) | 3 lines The CLI command should not contain
<placeholder>s, these are for descriptions. ........
2010-12-03 15:21 +0000 [r297486-297495] Matthew Nicholson <mnicholson@digium.com>
* res/res_fax.c: Print a DEBUG message instead of a WARNING message
when the selected fax tech does not support reserving sessions.
Answer the channel before quering it for t.38 support. This is
necessary for the query to work properly over local channels.
* include/asterisk/res_fax.h, res/res_fax.c: Add support for
reserving a fax session before answering the channel. Note: this
change breaks ABI compatibility. FAX-217
2010-12-02 20:09 +0000 [r297406] Paul Belanger <pabelanger@digium.com>
* Makefile, /: Merged revisions 297405 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r297405 | pabelanger | 2010-12-02 15:06:43 -0500
(Thu, 02 Dec 2010) | 14 lines Merged revisions 297404 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r297404 | pabelanger | 2010-12-02 15:01:08 -0500 (Thu, 02 Dec
2010) | 7 lines Resolve compile error under FreeBSD We now set
_ASTCFLAGS+=-march=i686 for i386 processors, still allowing
ASTCFLAGS to override the setting. Review:
https://reviewboard.asterisk.org/r/1043/ ........
................
2010-12-02 18:13 +0000 [r297312] Terry Wilson <twilson@digium.com>
* /, main/abstract_jb.c: Merged revisions 297311 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r297311 | twilson | 2010-12-02 12:07:39 -0600
(Thu, 02 Dec 2010) | 21 lines Merged revisions 297310 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r297310 | twilson | 2010-12-02 12:00:27 -0600 (Thu, 02 Dec 2010)
| 12 lines Initialize offset for adaptive jitter buffer When the
adaptive jitter buffer is enabled in sip.conf, the first frame
placed in the jitter buffer fails with something like:
jb_warning_output: Resyncing the jb. last_delay 0, this delay
-215886466, threshold 1000, new offset 215886466 This happens
because the offset is not initialized before calling jb_put().
This patch modifies jb_put_first_adaptive() to set the offset to
the frame's timestamp. Review:
https://reviewboard.asterisk.org/r/1041/ ........
................
2010-12-02 13:20 +0000 [r297245] Russell Bryant <russell@digium.com>
* /, apps/app_meetme.c: Merged revisions 297229 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r297229 | russell | 2010-12-02 07:16:47 -0600
(Thu, 02 Dec 2010) | 13 lines Merged revisions 297228 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r297228 | russell | 2010-12-02 07:16:15 -0600 (Thu, 02 Dec 2010)
| 6 lines Add "DAHDI" to a couple of app_meetme error messages.
This is in response to some questions on IRC. To the user, there
was nothing that made it obvious that this error had anything to
do with DAHDI not being loaded. ........ ................
2010-12-01 19:47 +0000 [r297157] Matthew Nicholson <mnicholson@digium.com>
* res/res_fax.c: Changed some NOTICE and WARNING messages to DEBUG
messages.
2010-12-01 17:53 +0000 [r297075] Jeff Peeler <jpeeler@digium.com>
* /, channels/chan_sip.c: Merged revisions 297073 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r297073 | jpeeler | 2010-12-01 11:52:46 -0600
(Wed, 01 Dec 2010) | 30 lines Merged revisions 297072 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r297072 | jpeeler | 2010-12-01 11:50:09 -0600 (Wed, 01 Dec 2010)
| 23 lines Fix not stopping MOH when transfered local channel
queue member is answered. The problem here is only present when
local channels are used with the MOH passthru option as well as
no optimization (/nm). I will describe the slightly bizarre
scenario that was used to test, where phones B and C are queue
members: Phone A dials into a queue with two members using local
channels and the above options. Phone B answers. Phone A blind
transfers phone B into the same queue. Phone A hangs up. Phone C
answers, but phone B didn't stop playing MOH. In this scenario,
the unhold frame that should have gotten to phone B never arrived
due to the masquerade from the blind transfer. This is usually
fine since app_queue manages the starting and stopping of MOH.
However, with the passthrough option enabled when app_queue
attempts to stop MOH it tries to do so on the local channel
rather than the real channel. The easiest solution was to just
make sure to send an unhold frame during the transfer since it
wouldn't make sense to have MOH playing after a transfer anyway.
This only modifies SIP transfers, but the other transfers did not
seem to be a problem. If DTMF based transfers were a problem it
might be okay to add ast_moh_stop to finishup, but I didn't want
to have to add that unless required. ABE-2624 ........
................
2010-12-01 17:01 +0000 [r296951-296992] Tilghman Lesher <tlesher@digium.com>
* include/asterisk/frame.h, /: Merged revisions 296991 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r296991 | tilghman | 2010-12-01 11:01:00 -0600
(Wed, 01 Dec 2010) | 12 lines Merged revisions 296990 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r296990 | tilghman | 2010-12-01 10:59:26 -0600 (Wed, 01 Dec 2010)
| 5 lines Clarify documentation on how we store codec preference
lists. (closes issue #18397) Reported by: birgita ........
................
* channels/chan_iax2.c, /: Merged revisions 296950 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
........ r296950 | tilghman | 2010-11-30 19:38:19 -0600 (Tue, 30
Nov 2010) | 2 lines Missed initializations caused startup errors
on Mac OS X (and possibly others, too). ........
2010-12-01 00:28 +0000 [r296870] Jeff Peeler <jpeeler@digium.com>
* apps/app_voicemail.c, /: Merged revisions 296869 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r296869 | jpeeler | 2010-11-30 18:24:58 -0600
(Tue, 30 Nov 2010) | 11 lines Merged revisions 296868 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r296868 | jpeeler | 2010-11-30 18:23:19 -0600 (Tue, 30 Nov 2010)
| 4 lines Properly restore backup information file when hanging
up during message prepending. ABE-2654 ........ ................
2010-11-30 19:12 +0000 [r296787] Tilghman Lesher <tlesher@digium.com>
* apps/app_meetme.c: DOC: Conference number can be omitted; if
omitted, all users in a meetme are listed.
2010-11-29 23:05 +0000 [r296673] Paul Belanger <pabelanger@digium.com>
* channels/chan_iax2.c, /: Merged revisions 296671 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r296671 | pabelanger | 2010-11-29 17:54:14 -0500
(Mon, 29 Nov 2010) | 12 lines Merged revisions 296670 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r296670 | pabelanger | 2010-11-29 17:49:39 -0500 (Mon, 29 Nov
2010) | 5 lines Make sure nothing else is needed before
destroying the scheduler. (closes issue #18398) Reported by:
pabelanger ........ ................
2010-11-29 21:26 +0000 [r296628] Russell Bryant <russell@digium.com>
* channels/chan_sip.c: Complete some error handling in
transmit_publish() in chan_sip.c. This error handling block
caught my eye. It was missing a couple of things, but it should
be safe now. Thanks to mmichelson for the quick peer review on
IRC.
2010-11-29 20:46 +0000 [r296582] Richard Mudgett <rmudgett@digium.com>
* channels/misdn/isdn_msg_parser.c, channels/chan_misdn.c: Merged
revision 296575 from
https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
.......... r296575 | rmudgett | 2010-11-29 14:27:37 -0600 (Mon,
29 Nov 2010) | 13 lines Invalid mISDN PTMP redirecting signaling
as TE towards NT. The mISDN PTMP redirection signaling (NOTIFY
redirecting number and notification code, SETUP redirecting
number) is also sent in PTMP/TE mode. It should only apply in
PTMP/NT mode. The call setup proceeds but the network (Deutsche
Telekom) reacts with ugly ISDN STATUS messages. Also don't send
the redirecting number ie when PTP is also sending the
DivertingLegInformation2 facility. The redirecting number ie is
redundant and the network (Deutsche Telekom) complains about it.
Patches: abe_2651_v4.patch uploaded by rmudgett (license 664)
JIRA ABE-2651 JIRA SWP-2537 ..........
2010-11-29 07:28 +0000 [r296534] Tilghman Lesher <tlesher@digium.com>
* main/asterisk.c, /, configure, include/asterisk/autoconfig.h.in,
configure.ac: Merged revisions 296533 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r296533 | tilghman | 2010-11-29 01:27:09 -0600 (Mon, 29 Nov 2010)
| 13 lines I love standards. There are so many to choose from.
Except when there isn't one. Linux and *BSD disagree on the
elements within the ucred structure. Detect which one is in use
on the system. (closes issue #18384) Reported by: bjm Patches:
cred-diffs uploaded by bjm (license 473)
20101127__issue18384__1.6.2.diff.txt uploaded by tilghman
(license 14) 20101127__issue18384__1.8.diff.txt uploaded by
tilghman (license 14) Tested by: tilghman, bjm ........
2010-11-27 10:40 +0000 [r296429-296467] Tilghman Lesher <tlesher@digium.com>
* /, apps/app_meetme.c: Merged revisions 296466 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r296466 | tilghman | 2010-11-27 04:39:01 -0600 (Sat, 27 Nov 2010)
| 5 lines 18 characters is too short for most date/times (20 is
the usual, but we add more in case of greater precision). (closes
issue #18369) Reported by: tnakonz ........
* include/asterisk.h: Also don't build DEBUG_FD_LEAKS when
STANDALONE2 is defined. (closes issue #18385) Reported by: cmaj
2010-11-26 21:37 +0000 [r296391] Olle Johansson <oej@edvina.net>
* main/say.c: Merged revisions 296351 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r296351 | oej | 2010-11-26 13:23:03 +0100 (Fre,
26 Nov 2010) | 17 lines Merged revisions 296309 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r296309 | oej | 2010-11-26 10:53:31 +0100 (Fre, 26 Nov 2010) | 11
lines Fix bugs in saying numbers using the Swedish language
syntax (closes issue #18355) Reported by: oej Patch by: oej Much
help from Peter Lindahl. Testing by the ClearIT team during a
coffee break. Review: https://reviewboard.asterisk.org/r/1033/
........ ................
2010-11-26 18:31 +0000 [r296352-296354] Brad Watkins <Marquis42@gmail.com>
* res/res_jabber.c: Fix XMPP PubSub-based distributed device state.
Initialize pubsubflags to 0 so res_jabber doesn't think there is
already an XMPP connection sending device state. Also clean up
CLI commands a bit. (closes issue #18272) Reported by: klaus3000
Patches: res_jabber_fix_pubsubflags_and_CLI-patch.txt uploaded by
klaus3000 (license 65) Tested by: klaus3000, Marquis Review:
https://reviewboard.asterisk.org/r/1030/
* channels/chan_sip.c: Fix reloading of peer when a user is
requested. Prevent peer reloading from causing multiple MWI
subscriptions to be created when using realtime. This had the
effect of sending one NOTIFY for every time a sip peer made a
call, in one case eventually overwhelming the phone and causing
it to reboot. (closes issue #18342) Reported by: nivek Patches:
issue0018342p1.patch uploaded by nivek (license 636) Tested by:
nivek Review: https://reviewboard.asterisk.org/r/1029/
2010-11-24 23:29 +0000 [r296230] Russell Bryant <russell@digium.com>
* main/channel.c, /: Merged revisions 296221 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r296221 | russell | 2010-11-24 17:28:19 -0600
(Wed, 24 Nov 2010) | 13 lines Merged revisions 296213 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r296213 | russell | 2010-11-24 17:26:43 -0600 (Wed, 24 Nov 2010)
| 6 lines Make Asterisk less crashy. Since we might not put a new
translation path on the channel, go ahead and set it to NULL
right after destroying the old one to ensure we don't try to free
an invalid translation path later on. ........ ................
2010-11-24 22:49 +0000 [r296167] Richard Mudgett <rmudgett@digium.com>
* channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_analog.c,
/, channels/sig_analog.h: Merged revisions 296166 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r296166 | rmudgett | 2010-11-24 16:42:45 -0600
(Wed, 24 Nov 2010) | 50 lines Merged revisions 296165 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r296165 | rmudgett | 2010-11-24 16:41:07 -0600 (Wed, 24 Nov 2010)
| 43 lines Oneway audio to SIP phone from FXS port after FXS port
gets a CallWaiting pip. The FXS connected phone has to have
CW/CID support to fail, as it will send back a DTMF 'A' or 'D'
when it's ready to receive CallerID. A normal phone with no CID
never fails. Also the SIP phone does not hear MOH when the CW
call is answered. The DTMF end frame is suppressed when the phone
acknowledges the CW signal for CID. The problem is the DTMF begin
frame needs to be suppressed as well. The DTMF begin frame is
causing SIP to start sending the DTMF RTP frames. Since the DTMF
end frame is suppressed, SIP will not stop sending those DTMF RTP
packets. * Suppress the DTMF begin and end frames when the
channel driver is looking for DTMF digits. * Fixed a couple
issues caused by not cleaning up the CID spill if you answer the
CW call while it is sending the CID spill. * Fixed not sending
CW/CID spill to the phone when the call is natively bridged.
(Fixed by not using native bridge if CW/CID is possible.) *
Suppress received audio when sending CW/CID spills. The other
parties involved do not need to hear the CW/CID spills and may be
confused if the CW call is for them. (closes issue #18129)
Reported by: alecdavis Patches: issue_18129_v1.8_v3.patch
uploaded by rmudgett (license 664) Tested by: alecdavis, rmudgett
NOTE: * v1.4 does not have the main problem fixed by suppressing
the DTMF start frames. The other three items fixed are relevant.
* If you really must restore native bridging between analog
ports, you need to disable CW/CID either by configuring
chan_dahdi.conf callwaitingcallerid=no or dialing *70 before
dialing the number to temporarily disable CW. ........
................
2010-11-24 20:23 +0000 [r296002-296084] Russell Bryant <russell@digium.com>
* main/channel.c, /: Merged revisions 296083 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r296083 | russell | 2010-11-24 14:23:11 -0600
(Wed, 24 Nov 2010) | 19 lines Merged revisions 296082 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r296082 | russell | 2010-11-24 14:22:32 -0600 (Wed, 24 Nov 2010)
| 12 lines Fix false reporting of an error by set_format(). In
the case that the native format was able to be changed to match
the new requested format, the code proceeded to attempt to build
a translation path, anyway. The result would be NULL, since no
translation path is necessary and resulted in this function
thinking an error has occurred. This case is now specifically
caught and no attempt to build a translation path is attempted.
Thanks to our automated tests and bamboo.asterisk.org for
catching this problem and making a whole lot of noise when things
started failing. :-) ........ ................
* apps/app_dial.c, main/channel.c, /: Merged revisions 296001 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r296001 | russell | 2010-11-24 11:03:16 -0600
(Wed, 24 Nov 2010) | 45 lines Merged revisions 296000 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r296000 | russell | 2010-11-24 10:48:39 -0600 (Wed, 24 Nov 2010)
| 38 lines Handle failures building translation paths more
effectively. The problem scenario occurred on a heavily loaded
system that was using the codec_dahdi module and exceeded the
hardware transcoding capacity. The failure mode at that point was
not good. The report came in to us as an Asterisk lock-up. The
"core show locks" shows a ton of threads locked up (but no
obvious deadlock). Upon deeper investigation, when the system is
in this state, the CPU was maxed out. The CPU was being consumed
by the Asterisk logger spewing messages on every audio frame for
calls set up after transcoder capacity was reached. The purpose
of this patch is to make Asterisk handle failures to create a
translation path in a more graceful manner. If we can't
translate, then the call just needs to be dropped, as it's not
going to work. These are the changes: 1) In set_format() of
channel.c (which is called by set_read_format() and
set_write_format()), it was ignoring if
ast_translator_build_path() failed and returned NULL. It now pays
attention to that case and returns a result reflecting failure.
With this change in place, the bridging code will immediately
detect a failure and end the bridge instead of proceeding to try
to bridge frames that can't be translated and making channel
drivers freak out by sending them frames in a format they weren't
expecting. 2) In ast_indicate_data() of channel.c, failure of
ast_playtones_start() was ignored. It is now reflected in the
return value of the function. This didn't turn out to have any
affect on the bug, but seemed like a good change to leave in. 3)
In app_dial(), when only sending a call to a single endpoint, it
will attempt to do some bridging of its own of early audio. It
uses make_compatible() when it's going to do this. However, it
ignored failure from make compatible. So, even with the fix from
#1, if there was early audio going through app_dial, there would
still be a period of invalid frames passing through. After
detecting failure here, Dial() exits. ABE-2658 ........
................
2010-11-23 10:30 +0000 [r295949] Olle Johansson <oej@edvina.net>
* /, main/say.c: Merged revisions 295907 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r295907 | oej | 2010-11-23 10:36:38 +0100 (Tis,
23 Nov 2010) | 14 lines Merged revisions 295906 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r295906 | oej | 2010-11-23 10:28:14 +0100 (Tis, 23 Nov 2010) | 8
lines Fix support of saynumber(1,n) in the Swedish language
(closes issue #18353) Reported by: oej Review:
https://reviewboard.asterisk.org/r/1031/ ........
................
2010-11-22 20:03 +0000 [r295869] Sean Bright <sean@malleable.com>
* configs/chan_dahdi.conf.sample, /: Merged revisions 295868 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r295868 | seanbright | 2010-11-22 15:02:37 -0500 (Mon, 22 Nov
2010) | 2 lines Change some documentation to suggest
dahdi_monitor instead of ztmonitor. ........
2010-11-22 19:36 +0000 [r295866] Richard Mudgett <rmudgett@digium.com>
* apps/app_macro.c, include/asterisk/channel.h,
include/asterisk/frame.h, main/channel.c, main/pbx.c, /: Merged
revisions 295843 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r295843 | rmudgett | 2010-11-22 13:28:23 -0600
(Mon, 22 Nov 2010) | 53 lines Merged revisions 295790 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r295790 | rmudgett | 2010-11-22 12:46:26 -0600 (Mon, 22 Nov 2010)
| 46 lines The channel redirect function (CLI or AMI) hangs up
the call instead of redirecting the call. To recreate the
problem: 1) Party A calls Party B 2) Invoke CLI "channel
redirect" command to redirect channel call leg associated with A.
3) All associated channels are hung up. Note that if the CLI
command were done on the channel call leg associated with B it
works. This regression was a result of the fix for issue #16946
(https://reviewboard.asterisk.org/r/740/). The regression affects
all features that use an async goto to execute the dialplan
because of an external event: Channel redirect, AMI redirect, SIP
REFER, and FAX detection. The struct ast_channel._softhangup code
is a mess. The variable is used for several purposes that do not
necessarily result in the call being hung up. I have added
doxygen comments to describe how the various _softhangup bits are
used. I have corrected all the places where the variable was
tested in a non-bit oriented manner. The primary fix is the new
AST_CONTROL_END_OF_Q frame. It acts as a weak hangup request so
the soft hangup requests that do not normally result in a hangup
do not hangup. JIRA SWP-2470 JIRA SWP-2489 (closes issue #18171)
Reported by: SantaFox (closes issue #18185) Reported by:
kwemheuer (closes issue #18211) Reported by: zahir_koradia
(closes issue #18230) Reported by: vmarrone (closes issue #18299)
Reported by: mbrevda (closes issue #18322) Reported by: nerbos
Review: https://reviewboard.asterisk.org/r/1013/ ........
................
2010-11-20 03:11 +0000 [r295747] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c, channels/sig_analog.c,
channels/sig_analog.h: One way audio before answering call
waiting call on analog port. * Analog call waiting Caller ID
spills could get stuck resulting in one way audio until the
waiting call is answered. This only happens on the second (and
later) call waiting call if the active call is not the first
call. * The CLI/AMI "dahdi show channel" command could report the
wrong channel information. Must keep the struct analog_pvt.owner
and struct dahdi_pvt.owner pointer in sync.
2010-11-20 00:50 +0000 [r295711] Russell Bryant <russell@digium.com>
* main/event.c, include/asterisk/event.h, /: Merged revisions
295710 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r295710 | russell | 2010-11-19 18:45:51 -0600 (Fri, 19 Nov 2010)
| 29 lines Fix cache of device state changes for multiple
servers. This patch addresses a regression where device states
across multiple servers were not being processing completely
correctly. The code works to determine the overall state by
looking at the last known state of a device on each server.
However, there was a regression due to some invasive rewrites of
how the cache works that led to the cache only storing the last
device state change for a device, regardless of which server it
was on. The code is set up to cache device state change events by
ensuring that each event in the cache has a unique device name +
entity ID (server ID). The code that was responsible for
comparing raw information elements (which EID is) always returned
a match due to a memcmp() with a length of 0. There isn't much
code to fix the actual bug. This patch also introduces a new CLI
command that was very useful for debugging this problem. The
command allows you to dump the contents of the event cache.
(closes issue #18284) Reported by: klaus3000 Patches:
issue18284.rev1.txt uploaded by russell (license 2) Tested by:
russell, klaus3000 (closes issue #18280) Reported by: klaus3000
Review: https://reviewboard.asterisk.org/r/1012/ ........
2010-11-19 22:06 +0000 [r295673] Terry Wilson <twilson@digium.com>
* /, channels/chan_sip.c: Merged revisions 295672 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r295672 | twilson | 2010-11-19 13:55:48 -0800
(Fri, 19 Nov 2010) | 15 lines Merged revisions 295628 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r295628 | twilson | 2010-11-19 12:53:36 -0800 (Fri, 19 Nov 2010)
| 8 lines Discard responses with more than one Via This is not a
perfect solution as headers that are joined via commas are not
detected. This is a parsing issue that to fix "correctly" would
necessitate a new SIP parser. Review:
https://reviewboard.asterisk.org/r/1019/ ........
................
2010-11-19 21:40 +0000 [r295670] Brett Bryant <bbryant@digium.com>
* apps/app_queue.c: Patch for deadlock from ordering issue between
channel/queue locks in app_queue (set_queue_variables). (closes
issue #18031) Reported by: rain Review:
https://reviewboard.asterisk.org/r/1018/
2010-11-19 16:47 +0000 [r295516] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c, channels/sig_analog.c,
channels/sig_analog.h: Bring sig_analog extraction more into
alignment with orig-trunk/v1.6.2 chan_dahdi. * Restore SMDI
support. * Fixed initial value of struct analog_pvt.use_callerid.
It may get forced on depending upon other config options. * Call
analog_dnd() instead of manual inlined code. * Removed unused
struct analog_pvt.usedistinctiveringdetection. * Removed the
struct analog_pvt.unknown_alarm flag. It was really the struct
analog_pvt.inalarm flag. * Use ast_debug() instead of
ast_log(LOG_DEBUG). * Rename several function's index variable to
idx. * Some formatting tweaks.
2010-11-18 20:30 +0000 [r295477] Leif Madsen <lmadsen@digium.com>
* configs/sip_notify.conf.sample: 'sip notify clear-mwi' needs
terminating CRLF. (closes issue #18275) Reported by: klaus3000
Patches: fix_body_CRLF_patch.txt uploaded by klaus3000 (license
65)
2010-11-18 18:02 +0000 [r295361-295441] Paul Belanger <pabelanger@digium.com>
* res/res_jabber.c, /, include/asterisk/jabber.h: Merged revisions
295440 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r295440 | pabelanger | 2010-11-18 12:51:34 -0500 (Thu, 18 Nov
2010) | 4 lines Fix compiler warnings when using openssl-dev
1.0.0+ Review: https://reviewboard.asterisk.org/r/1016/ ........
* contrib/scripts/install_prereq: Add RedHat specific dependencies
* configs/res_curl.conf.sample: Uncomment settings under [global],
to surpress warning when loading Asterisk.
2010-11-16 23:02 +0000 [r295282] Richard Mudgett <rmudgett@digium.com>
* main/channel.c, /: Merged revisions 295281 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r295281 | rmudgett | 2010-11-16 16:57:07 -0600
(Tue, 16 Nov 2010) | 9 lines Merged revisions 295280 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r295280 | rmudgett | 2010-11-16 16:52:06 -0600 (Tue, 16
Nov 2010) | 1 line Dead code elimination in
channel.c:ast_channel_bridge() variable who. ........
................
2010-11-16 22:41 +0000 [r295164-295278] Russell Bryant <russell@digium.com>
* build_tools/prep_tarball: Check for pdftotext and give a useful
error if not found.
* build_tools/prep_tarball: Remove intentional typo I had added
when testing the check. oops.
* build_tools/prep_tarball: Check for wikiexport.py in PATH and
give a useful error message if not found.
2010-12-02 Leif Madsen <lmadsen@digium.com>
* Asterisk 1.8.1 Released.
2010-11-16 Leif Madsen <lmadsen@digium.com>
* Asterisk 1.8.1-rc1 Released.
2010-11-15 18:30 +0000 [r294989-295078] Tilghman Lesher <tlesher@digium.com>
* tests/test_expr.c (added), /: Merged revisions 295062 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r295062 | tilghman | 2010-11-15 12:24:02 -0600
(Mon, 15 Nov 2010) | 9 lines Merged revisions 295026 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r295026 | tilghman | 2010-11-15 11:58:37 -0600 (Mon, 15
Nov 2010) | 2 lines Create test verifying results of expression
parser ........ ................
* funcs/func_curl.c, /: Merged revisions 294988 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r294988 | tilghman | 2010-11-15 01:42:39 -0600 (Mon, 15 Nov 2010)
| 8 lines It is possible to crash Asterisk by feeding the curl
engine invalid data. (closes issue #18161) Reported by: wdoekes
Patches: 20101029__issue18161.diff.txt uploaded by tilghman
(license 14) Tested by: tilghman ........
2010-11-12 21:14 +0000 [r294905-294911] Jeff Peeler <jpeeler@digium.com>
* apps/app_voicemail.c, /: Merged revisions 294910 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
........ r294910 | jpeeler | 2010-11-12 15:14:23 -0600 (Fri, 12
Nov 2010) | 4 lines Return correct error code if lock path fails.
The recent changes to open_mailbox actually caused it to be
fixed, but let's be consistent. Reported by alecdavis in
asterisk-dev. ........
* apps/app_voicemail.c, /: Merged revisions 294904 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r294904 | jpeeler | 2010-11-12 14:51:15 -0600
(Fri, 12 Nov 2010) | 23 lines Merged revisions 294903 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r294903 | jpeeler | 2010-11-12 14:49:09 -0600 (Fri, 12 Nov 2010)
| 16 lines Fix regression causing abort in voicemail after
opening a mailbox with no mesgs. In order to be more safe, some
error handling code was changed to respect more error conditions
including the potential memory allocation failure for deleted and
heard message tracking introduced in 293004. However,
last_message_index returns -1 for zero messages (perhaps as
expected) and was triggering the stricter error checking. Because
last_message_index is only called directly in one place, just
return 0 from open_mailbox (for file based storage) when no
messages are detected unless a real error has occurred. (closes
issue #18240) Reported by: leobrown Patches:
bug18240.1-6-2.diff.txt uploaded by alecdavis (license 585)
Tested by: pabelanger ........ ................
2010-11-12 02:45 +0000 [r294823] Richard Mudgett <rmudgett@digium.com>
* channels/sig_pri.c, channels/sig_pri.h, /: Merged revisions
294822 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r294822 | rmudgett | 2010-11-11 20:44:12 -0600
(Thu, 11 Nov 2010) | 18 lines Merged revisions 294821 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r294821 | rmudgett | 2010-11-11 20:41:13 -0600 (Thu, 11 Nov 2010)
| 11 lines Asterisk is getting a "No D-channels available!"
warning message every 4 seconds. Asterisk is just whining too
much with this message: "No D-channels available! Using Primary
channel XXX as D-channel anyway!". Filtered the message so it
only comes out once if there is no D channel available without an
intervening D channel available period. (closes issue #17270)
Reported by: jmls ........ ................
2010-11-11 22:17 +0000 [r294740-294745] Russell Bryant <russell@digium.com>
* doc/CCSS_architecture.pdf (removed): Remove CCSS architecture
PDF. It has been moved to:
https://wiki.asterisk.org/wiki/display/AST/CCSS+Architecture
* doc/digium-mib.txt (removed), doc/followme.txt (removed),
doc/building_queues.txt (removed), doc/timing.txt (removed),
doc/advice_of_charge.txt (removed), doc/unistim.txt (removed),
doc/video_console.txt (removed), doc/macroexclusive.txt
(removed), doc/google-soc2009-ideas.txt (removed), doc/README.txt
(added), doc/callfiles.txt (removed), doc/externalivr.txt
(removed), doc/codec-64bit.txt (removed),
build_tools/prep_tarball, doc/video.txt (removed), doc/jingle.txt
(removed), doc/modules.txt (removed), doc/manager_1_1.txt
(removed), doc/PEERING (removed), doc/snmp.txt (removed),
doc/siptls.txt (removed), doc/HOWTO_collect_debug_information.txt
(removed), doc/ldap.txt (removed), doc/sip-retransmit.txt
(removed), doc/distributed_devstate.txt (removed),
doc/voicemail_odbc_postgresql.txt (removed), doc/tex (removed),
doc/queue.txt (removed), doc/jabber.txt (removed),
doc/chan_sip-perf-testing.txt (removed), Makefile,
doc/asterisk-mib.txt (removed), doc/database_transactions.txt
(removed), doc/smdi.txt (removed), doc/janitor-projects.txt
(removed), doc/hoard.txt (removed), doc/res_config_sqlite.txt
(removed), doc/osp.txt (removed), doc/speechrec.txt (removed),
doc/sms.txt (removed), doc/distributed_devstate-XMPP.txt
(removed), doc/valgrind.txt (removed), doc/realtimetext.txt
(removed), doc/cli.txt (removed), doc/rtp-packetization.txt
(removed), doc/datastores.txt (removed), doc/CODING-GUIDELINES
(removed), doc/ss7.txt (removed), doc/backtrace.txt (removed),
doc/India-CID.txt (removed): Remove most of the contents of the
doc dir in favor of the wiki content. This merge does the
following things: * Removes most of the contents from the doc/
directory in favor of the wiki - http://wiki.asterisk.org/ *
Updates the build_tools/prep_tarball script to know how to export
the contents of the wiki in both PDF and plain text formats so
that the documentation is still included in Asterisk release
tarballs.
2010-11-11 21:58 +0000 [r294640-294734] Jeff Peeler <jpeeler@digium.com>
* /, channels/chan_sip.c: Merged revisions 294733 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r294733 | jpeeler | 2010-11-11 15:57:22 -0600
(Thu, 11 Nov 2010) | 25 lines Merged revisions 294688 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r294688 | jpeeler | 2010-11-11 15:12:27 -0600 (Thu, 11 Nov 2010)
| 18 lines Fix problem with qualify option packets for realtime
peers never stopping. The option packets not only never stopped,
but if a realtime peer was not in the peer list multiple options
dialogs could accumulate over time. This scenario has the
potential to progress to the point of saturating a link just from
options packets. The fix was to ensure that the poke scheduler
checks to see if a peer is in the peer list before continuing to
poke. The reason a peer must be in the peer list to be able to
properly manage an options dialog is because otherwise the call
pointer is lost when the peer is regenerated from the database,
which is how existing qualify dialogs are detected. (closes issue
#16382) (closes issue #17779) Reported by: lftsy Patches:
bug16382-3.patch uploaded by jpeeler (license 325) Tested by:
zerohalo ........ ................
* main/asterisk.c, include/asterisk.h, main/pbx.c, /: Merged
revisions 294639 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r294639 | jpeeler | 2010-11-11 13:31:00 -0600
(Thu, 11 Nov 2010) | 53 lines Merged revisions 294384 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r294384 | jpeeler | 2010-11-09 11:37:59 -0600 (Tue, 09 Nov 2010)
| 47 lines Fix a deadlock in device state change processing.
Copied from some notes from the original author (Russell):
Deadlock scenario: Thread 1: device state change thread Holds -
rdlock on contexts Holds - hints lock Waiting on channels
container lock Thread 2: SIP monitor thread Holds the "iflock"
Holds a sip_pvt lock Holds channel container lock Waiting for a
channel lock Thread 3: A channel thread (chan_local in this case)
Holds 2 channel locks acquired within app_dial Holds a 3rd
channel lock it got inside of chan_local Holds a local_pvt lock
Waiting on a rdlock of the contexts lock A bunch of other threads
waiting on a wrlock of the contexts lock To address this
deadlock, some locking order rules must be put in place and
enforced. Existing relevant rules: 1) channel lock before a pvt
lock 2) contexts lock before hints lock 3) channels container
before a channel What's missing is some enforcement of the order
when you involve more than any two. To fix this problem, I put in
some code that ensures that (at least in the code paths involved
in this bug) the locks in (3) come before the locks in (2). To
change the operation of thread 1 to comply, I converted the
storage of hints to an astobj2 container. This allows processing
of hints without holding the hints container lock. So, in the
code path that led to thread 1's state, it no longer holds either
the contexts or hints lock while it attempts to lock the channels
container. (closes issue #18165) Reported by: antonio ABE-2583
........ ................
2010-11-10 23:26 +0000 [r294569-294605] Tilghman Lesher <tlesher@digium.com>
* pbx/pbx_spool.c: Fixing the Mac OS X build (bamboo warning)
* pbx/pbx_spool.c: Properly queue files with inotify(7). (closes
issue #18089) Reported by: abelbeck Patches:
20101021__issue18089.diff.txt uploaded by tilghman (license 14)
Tested by: tilghman
2010-11-10 14:14 +0000 [r294501-294535] Russell Bryant <russell@digium.com>
* UPGRADE.txt, res/ais/clm.c, res/ais/evt.c: Tweak a couple of CLI
commands back to their original form. The "module" in this case
is two parts, so there are two words before the verb of the CLI
command.
* main/devicestate.c, /: Merged revisions 294500 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r294500 | russell | 2010-11-10 06:41:41 -0600 (Wed, 10 Nov 2010)
| 7 lines Improve a debug message to be more readable and
consistent. (closes issue #18282) Reported by: klaus3000 Patches:
ast_devstate2str-patch.txt uploaded by klaus3000 (license 65)
........
2010-11-09 22:46 +0000 [r294466] Richard Mudgett <rmudgett@digium.com>
* main/channel.c: Allow ast_do_masquerade() failure to be reported
again.
2010-11-09 20:33 +0000 [r294430] Tilghman Lesher <tlesher@digium.com>
* /, configure, include/asterisk/autoconfig.h.in, configure.ac:
Merged revisions 294429 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r294429 | tilghman | 2010-11-09 14:27:23 -0600 (Tue, 09 Nov 2010)
| 8 lines Detect GMime properly on systems where gmime flags and
libs are configured with pkg-config. (closes issue #16155)
Reported by: jcollie Patches: 20100917__issue16155.diff.txt
uploaded by tilghman (license 14) Tested by: tilghman ........
2010-11-09 16:55 +0000 [r294349] Richard Mudgett <rmudgett@digium.com>
* include/asterisk/channel.h, channels/sig_pri.c, main/channel.c,
channels/chan_misdn.c, channels/sig_analog.c: Analog lines do not
transfer CONNECTED LINE or execute the interception macros. Add
connected line update for sig_analog transfers and simplify the
corresponding sig_pri and chan_misdn transfer code. Note that if
you create a three-way call in sig_analog before transferring the
call, the distinction of the caller/callee interception macros
make little sense. The interception macro writer needs to be
prepared for either caller/callee macro to be executed. The
current implementation swaps which caller/callee interception
macro is executed after a three-way call is created. Review:
https://reviewboard.asterisk.org/r/996/ JIRA ABE-2589 JIRA
SWP-2372
2010-11-08 22:32 +0000 [r294278-294313] Jeff Peeler <jpeeler@digium.com>
* /, res/res_timing_timerfd.c: Merged revisions 294312 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
........ r294312 | jpeeler | 2010-11-08 16:30:49 -0600 (Mon, 08
Nov 2010) | 1 line add missing unlock not present in 294277
........
* include/asterisk/timing.h, main/timing.c, main/channel.c, /,
res/res_timing_timerfd.c: Merged revisions 294277 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
........ r294277 | jpeeler | 2010-11-08 15:58:13 -0600 (Mon, 08
Nov 2010) | 16 lines Fix playback failure when using IAX with the
timerfd module. To fix this issue the alert pipe will now be used
when the timerfd module is in use. There appeared to be a race
that was not solved by adding locking in the timerfd module, but
needed to be there anyway. The race was between the timer being
put in non-continuous mode in ast_read on the channel thread and
the IAX frame scheduler queuing a frame which would enable
continuous mode before the non-continuous mode event was read.
This race for now is simply avoided. (closes issue #18110)
Reported by: tpanton Tested by: tpanton I put tested by tpanton
because it was tested on his hardware. Thanks for the remote
access to debug this issue! ........
2010-11-08 20:56 +0000 [r294243] Matthew Nicholson <mnicholson@digium.com>
* /, channels/chan_sip.c: Merged revisions 294242 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r294242 | mnicholson | 2010-11-08 14:50:21 -0600 (Mon, 08 Nov
2010) | 8 lines Go off hold when we get an empty reinvite telling
us to. (closes issue 0014448) Reported by: frawd (closes issue
#17878) Reported by: frawd ........
2010-11-08 19:56 +0000 [r294207] Terry Wilson <twilson@digium.com>
* configs/calendar.conf.sample, res/res_calendar.c: Set a default
waittime, and make sure to convert it to milliseconds
2010-11-08 17:16 +0000 [r294125] Richard Mudgett <rmudgett@digium.com>
* channels/chan_misdn.c: valgrind reported references to freed
memory during a mISDN hangup collision. Bad things have been
happening in chan_misdn because the chan_misdn channel private
struct chan_list is not protected from reentrancy. Hangup
collisions have be causing read and write accesses to freed
memory. Converted chan_misdn struct chan_list to an ao2 object
for its reference counting feature. ********** Removed an
impediment to converting chan_list to an ao2 object. The use of
the other_ch member in chan_list is shaky at best. It is set if
the incoming and outgoing call legs are mISDN. The use of the
other_ch member goes against the Asterisk architecture and can
even cause problems. 1) It is used to disable echo cancellation.
This could be bad if the call is forked and the winning call leg
is not mISDN or the winning call leg is not the last mISDN
channel called by the fork. The other_ch would become a dangling
pointer. 2) It is used when the far end is alerting to hear the
far end's inband audio instead of Asterisk's generated ringback
tone. This is bad if the call is forked. You would only hear the
last forked mISDN channel and it may not be ringing yet. The
other_ch would become a dangling pointer if the call is later
transferred. ********** JIRA SWP-2423 JIRA ABE-2614
2010-11-05 22:03 +0000 [r294084] Brett Bryant <bbryant@digium.com>
* channels/chan_sip.c: Fixed deadlock avoidance issues while
locking channel when adding the Max-Forwards header to a request.
(closes issue #17949) (closes issue #18200) Reported by: bwg
Review: https://reviewboard.asterisk.org/r/997/
2010-11-05 16:05 +0000 [r294047-294049] Terry Wilson <twilson@digium.com>
* contrib/scripts/ast_tls_cert: Corret spelling and example
* contrib/scripts/ast_tls_cert: Tell people to use the correct
common name for clients as well
2010-11-05 00:07 +0000 [r293970] Shaun Ruffell <sruffell@digium.com>
* codecs/codec_dahdi.c, /: Merged revisions 293969 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r293969 | sruffell | 2010-11-04 19:06:02 -0500
(Thu, 04 Nov 2010) | 25 lines Merged revisions 293968 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r293968 | sruffell | 2010-11-04 19:02:53 -0500 (Thu, 04 Nov 2010)
| 17 lines codecs/codec_dahdi: Prevent "choppy" audio when
receiving unexpected frame sizes. dahdi-linux 2.4.0 (specifically
commit 9034) added the capability for the wctc4xxp to return more
than a single packet of data in response to a read. However, when
decoding packets, codec_dahdi was still assuming that the default
number of samples was in each read. In other words, each packet
your provider sent you, regardless of size, would result in 20 ms
of decoded data (30 ms if decoding G723). If your provider was
sending 60 ms packets then codec_dahdi would end up stripping 40
ms of data from each transcoded frame resulting in "choppy"
audio. This would only affect systems where G729 packets are
arriving in sizes greater than 20ms or G723 packets arriving in
sizes greater than 30ms. DAHDI-744. ........ ................
2010-11-04 21:39 +0000 [r293924] David Vossel <dvossel@digium.com>
* channels/chan_sip.c: Fixes ringback tone on sip semi-attended
transfer. ABE-2168
2010-11-04 13:27 +0000 [r293887] Paul Belanger <paul.belanger@polybeacon.com>
* channels/chan_sip.c: Do not output port in IPaddress for AMI
sippeers. (closes issue #18248) Reported by: orn Patches:
ami_sippeers.patch uploaded by pabelanger (license 224) Tested
by: orn
2010-11-03 18:35 +0000 [r293807] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c, channels/sig_analog.c, /: Merged revisions
293806 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r293806 | rmudgett | 2010-11-03 13:31:57 -0500
(Wed, 03 Nov 2010) | 27 lines Merged revisions 293805 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r293805 | rmudgett | 2010-11-03 13:23:04 -0500 (Wed, 03 Nov 2010)
| 20 lines Party A in an analog 3-way call would continue to hear
ringback after party C answers. All parties are analog FXS ports.
1) A calls B. 2) A flash hooks to call C. 3) A flash hooks to
bring C into 3-way call before C answers. (A and B hear ringback)
4) C answers 5) A continues to hear ringback during the 3-way
call. (All parties can hear each other.) * Fixed use of wrong
variable in dahdi_bridge() that stopped ringback on the wrong
subchannel. * Made several debug messages have more information.
A similar issue happens if B and C are SIP channels. B continues
to hear ringback. For some reason this only affects v1.8 and
trunk. * Don't start ringback on the real and 3-way subchannels
when creating the 3-way conference. Removing this code is benign
on v1.6.2 and earlier. ........ ................
2010-11-03 18:05 +0000 [r293803] Terry Wilson <twilson@digium.com>
* include/asterisk/rtp_engine.h, main/rtp_engine.c,
channels/chan_sip.c: Avoid valgrind warnings for
ast_rtp_instance_get_xxx_address The documentation for
ast_rtp_instance_get_(local/remote)_address stated that they
returned 0 for success and -1 on failure. Instead, they returned
0 if the address structure passed in was already equivalent to
the address instance local/remote address or 1 otherwise. 90% of
the calls to these functions completely ignored the return
address and passed in an uninitialized struct, which would make
valgrind complain even though the operation was technically safe.
This patch fixes the documentation and converts the
get_xxx_address functions to void since all they really do is
copy the address and cannot fail. Additionally two new functions
(ast_rtp_instance_get_and_cmp_(local/remote)_address) are created
for the 3 times where the return value was actually checked. The
get_and_cmp_local_address function is currently unused, but
exists for the sake of symmetry. The only functional change as a
result of this change is that we will not do an
ast_sockaddr_cmp() on (mostly uninitialized) addresses before
doing the ast_sockaddr_copy() in the get_*_address functions. So,
even though it is an API change, it shouldn't have a noticeable
change in behavior. Review:
https://reviewboard.asterisk.org/r/995/
2010-11-02 23:09 +0000 [r293724] Jeff Peeler <jpeeler@digium.com>
* /, channels/chan_sip.c: Merged revisions 293723 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r293723 | jpeeler | 2010-11-02 18:07:13 -0500
(Tue, 02 Nov 2010) | 15 lines Merged revisions 293722 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r293722 | jpeeler | 2010-11-02 18:02:51 -0500 (Tue, 02 Nov 2010)
| 8 lines Add enabled/disabled information for rtautoclear sip
show settings output. When setting to zero/"no", the numeric
default was shown making it not obvious the disabled setting was
respected. (closes issue #18123) Reported by: zerohalo ........
................
2010-11-02 21:29 +0000 [r293648] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c, channels/sig_analog.c, /: Merged revisions
293647 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r293647 | rmudgett | 2010-11-02 16:26:30 -0500
(Tue, 02 Nov 2010) | 13 lines Merged revisions 293639 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r293639 | rmudgett | 2010-11-02 16:24:13 -0500 (Tue, 02 Nov 2010)
| 6 lines Make warning message have more useful information in
it. Change "Unable to get index, and nullok is not asserted" to
"Unable to get index for '<channel-name>' on channel <number>
(<function>(), line <number>)". ........ ................
2010-11-02 20:45 +0000 [r293611] Paul Belanger <paul.belanger@polybeacon.com>
* main/manager.c: If manager and tls are disabled, do not display
TCP/TLS Bindaddress.
2010-11-01 17:29 +0000 [r293530] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c, channels/sig_analog.c,
channels/sig_analog.h: Analog 3-way call would not connect all
parties if one was using sig_pri. Also the "dahdi show channel"
would not show the correct 3-way call status. * Synchronized the
inthreeway flag between chan_dahdi and sig_analog. * Fixed a
my_set_linear_mode() sign error and made take an analog sub
channel enum.
2010-11-01 16:09 +0000 [r293496] Paul Belanger <paul.belanger@polybeacon.com>
* channels/chan_iax2.c: Use ast_sockaddr_from_sin function not
memcpy This resolves some IAX2 registration issue report on the
asterisk-users mailing list. (closes issue #18202) Reported by:
pabelanger Patches: update_registry.patch.v2 uploaded by
pabelanger (license 224) Tested by: pabelanger, Nic Colledge
(mailing list) Review: https://reviewboard.asterisk.org/r/993
2010-11-01 14:58 +0000 [r293493] Terry Wilson <twilson@digium.com>
* channels/chan_sip.c: Only offer codecs both sides support for
directmedia When using directmedia, Asterisk needs to limit the
codecs offered to just the ones that both sides recognize,
otherwise they may end up sending audio that the other side
doesn't understand. (closes issue #17403) Reported by: one47
Patches: sip_codecs_simplified4 uploaded by one47 (license 23)
Tested by: one47, falves11 Review:
https://reviewboard.asterisk.org/r/967/
2010-10-30 01:53 +0000 [r293341-293418] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c, channels/sig_analog.c, /: Merged revisions
293417 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r293417 | rmudgett | 2010-10-29 20:49:15 -0500
(Fri, 29 Oct 2010) | 9 lines Merged revisions 293416 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r293416 | rmudgett | 2010-10-29 20:45:49 -0500 (Fri, 29
Oct 2010) | 1 line Remove some more code that serves no purpose.
........ ................
* channels/chan_dahdi.c, channels/sig_analog.c, /: Merged revisions
293340 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r293340 | rmudgett | 2010-10-29 19:40:10 -0500
(Fri, 29 Oct 2010) | 9 lines Merged revisions 293339 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r293339 | rmudgett | 2010-10-29 19:34:12 -0500 (Fri, 29
Oct 2010) | 1 line Remove some code that serves no purpose.
........ ................
2010-10-29 21:48 +0000 [r293305] Jeff Peeler <jpeeler@digium.com>
* channels/chan_sip.c: Modify sip_setoption to not complain about
unknown options. This now behaves just like the other setoption
callbacks. For the curious the offending option for the reporter
was AST_OPTION_CHANNEL_WRITE which was getting passed due to a
fix for chan_local in 286189. (closes issue #17985) Reported by:
globalnetinc
2010-10-28 20:00 +0000 [r293197] Tilghman Lesher <tlesher@digium.com>
* res/ael/ael.tab.h, main/ast_expr2.c, /, main/ast_expr2.h,
res/ael/ael.tab.c, main/ast_expr2.y, res/ael/ael_lex.c: Merged
revisions 293195-293196 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r293195 | tilghman | 2010-10-28 14:52:52 -0500
(Thu, 28 Oct 2010) | 12 lines Merged revisions 293194 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r293194 | tilghman | 2010-10-28 14:44:37 -0500 (Thu, 28 Oct 2010)
| 5 lines "!00" evaluated as false, which is incorrect. Fixing.
Reported (though the reporter did not understand he was reporting
a bug) on the asterisk-users list:
http://lists.digium.com/pipermail/asterisk-users/2010-October/255505.html
........ ................ r293196 | tilghman | 2010-10-28
14:54:34 -0500 (Thu, 28 Oct 2010) | 12 lines Merged revisions
293194 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r293194 | tilghman | 2010-10-28 14:44:37 -0500 (Thu, 28 Oct 2010)
| 5 lines "!00" evaluated as false, which is incorrect. Fixing.
Reported (though the reporter did not understand he was reporting
a bug) on the asterisk-users list:
http://lists.digium.com/pipermail/asterisk-users/2010-October/255505.html
........ ................
2010-10-28 16:11 +0000 [r293159] Jeff Peeler <jpeeler@digium.com>
* /, funcs/func_strings.c: Merged revisions 293158 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
........ r293158 | jpeeler | 2010-10-28 11:09:40 -0500 (Thu, 28
Oct 2010) | 11 lines Fix infinite loop in FILTER(). Specifically
when you're using characters above \x7f or invalid character
escapes (e.g. \xgg). (closes issue #18060) Reported by: wdoekes
Patches: issue18060_func_strings_filter_infinite_loop.patch
uploaded by wdoekes (license 717) Tested by: wdoekes ........
2010-10-26 18:49 +0000 [r293119] Jeff Peeler <jpeeler@digium.com>
* apps/app_voicemail.c, /: Merged revisions 293118 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r293118 | jpeeler | 2010-10-26 13:33:24 -0500
(Tue, 26 Oct 2010) | 36 lines Merged revisions 293004 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r293004 | jpeeler | 2010-10-25 17:55:28 -0500 (Mon, 25 Oct 2010)
| 29 lines Fix inprocess_container in voicemail to correctly
restrict max messages. The comparison function logic was off, so
the number of sessions for a given mailbox were not being
incremented properly. This problem caused the maximum number of
messages per folder to not be respected when simultaneously
leaving multiple voicemails just below the threshold. These
problems should be fixed by the above, but just in case: Fixed
resequence_mailbox to rely on the actual number of detected
number of files in a directory rather than just assuming only 10
messages more than the maximum had been left. Also if more
messages than the maximum are deleted they are actually removed
now. The second purpose of this commit should have been separated
out probably, but is related to the above. Again, if the number
of messages in a given voicemail folder exceeds the maximum set
limit make sure to allocate enough space for the deleted and
heard index tracking array. A few random fixes: There was a
forgotten decrement of the inprocess count in imap_store_file.
When using IMAP storage, do not look in the directory where file
based storage messages may still reside and influence the message
count. Ensure to use only the first format in sendmail. ABE-2516
........ ................
2010-10-26 16:32 +0000 [r293046-293081] Richard Mudgett <rmudgett@digium.com>
* channels/sig_pri.c: No need to define the struct if there are no
users.
* channels/sig_pri.c, configure, include/asterisk/autoconfig.h.in,
configure.ac: Allow the DAHDI driver to compile, even with a
sufficiently older version of libpri. Fixes our Bamboo builds.
2010-10-25 21:15 +0000 [r292906-292969] Tilghman Lesher <tlesher@digium.com>
* channels/sig_pri.c: Several more defines that need to be altered
for compiling against an older version of libpri
* channels/sig_pri.c, configure, include/asterisk/autoconfig.h.in,
configure.ac: Allow the DAHDI driver to compile, even with a
sufficiently older version of libpri. Fixes our Bamboo builds.
2010-10-25 19:07 +0000 [r292868] David Vossel <dvossel@digium.com>
* channels/chan_local.c, /: Merged revisions 292867 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r292867 | dvossel | 2010-10-25 14:06:21 -0500
(Mon, 25 Oct 2010) | 32 lines Merged revisions 292866 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r292866 | dvossel | 2010-10-25 14:05:07 -0500 (Mon, 25 Oct 2010)
| 27 lines This patch turns chan_local pvts into astobj2 objects.
chan_local does some dangerous things involving deadlock
avoidance. tech_pvt functions like hangup and queue_frame are
provided with a locked channel upon entry. Those functions are
completely safe as long as you don't attempt to give up that
channel lock, but that is impossible to guarantee due to the
required deadlock avoidance necessary to lock both the tech_pvt
and both channels involved. In the past, we have tried to account
for this by doing things like setting a "glare" flag that
indicates what function should destroy the pvt. This was used in
local_hangup and local_queue_frame to decided who should destroy
the pvt if they collided in separate threads. I have removed the
need to do this by converting all chan_local tech_pvts to
astobj2. This means we can ref a pvt before deadlock avoidance
and not have to worry about that pvt possibly getting destroyed
under us. It also cleans up where we destroy the tech_pvt. The
only unlink from the tech_pvt container occurs in local_hangup
now, which is where it should occur. Since there still may be
thread collisions on some functions like local_hangup after
deadlock avoidance, I have added some checks to detect those
collisions and exit appropriately. I think this patch is going to
solve quite a bit of weirdness we have had with local channels in
the past. ........ ................
2010-10-22 22:35 +0000 [r292794-292825] Terry Wilson <twilson@digium.com>
* contrib/scripts/ast_tls_cert: Don't create directories without at
least o+x Also, making files that you are going to modify
read-only is dumb.
* contrib/scripts/ast_tls_cert: Make files readable only by the
owner
2010-10-22 21:28 +0000 [r292787] Leif Madsen <lmadsen@digium.com>
* configs/res_ldap.conf.sample, contrib/scripts/asterisk.ldif, /,
channels/chan_sip.c: Merged revisions 292786 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r292786 | lmadsen | 2010-10-22 16:16:12 -0500 (Fri, 22 Oct 2010)
| 13 lines Update the LDIF file for LDAP. The LDIF file
asterisk.ldif was quite a bit out of date from the
asterisk.ldap-schema file, so I've now updated that to be in
sync. The asterisk.ldif file being out of sync was a problem on
my systems where I was doing an ldapadd to import the schema into
the LDAP database, and the existing file would cause problems and
ERROR messages when registering. Additional documention has been
added based on feedback in the issue I'm closing. (closes issue
#13861) Reported by: scramatte Patches: ldap-update.txt uploaded
by lmadsen (license 10) Tested by: lmadsen, jcovert, suretec,
rgenthner ........
2010-10-22 17:09 +0000 [r292741] Mark Michelson <mmichelson@digium.com>
* tests/test_event.c: Prevent multiple runs of event_sub_test from
producing false failure results. The array of test subscriptions
was declared "static," meaning that the data.count field would
retain its value between runs of the test. After the first test
run, this would result in false reports of test failures. I chose
to just remove the "static" keyword from the structure since it's
not a huge deal to construct this structure during each run of
the test. Another alternative would have been to zero out the
data.count fields of each test subscription instead.
2010-10-22 16:49 +0000 [r292740] Terry Wilson <twilson@digium.com>
* contrib/scripts/ast_tls_cert (added): Add TLS cert helper script
This script is useful for quickly generating self-signed CA,
server, and client certificates for use with Asterisk. It is
still recommended to obtain certificates from a recognized
Certificate Authority and to develop an understanding how SSL
certificates work. Real security is hard work. OPTIONS: -h Show
this message -m Type of cert "client" or "server". Defaults to
server. -f Config filename (openssl config file format) -c CA
cert filename (creates new CA cert/key as ca.crt/ca.key if not
passed) -k CA key filename -C Common name (cert field) For a
server cert, this should be the same address that clients attempt
to connect to. Usually this will be the Fully Qualified Domain
Name, but might be the IP of the server. For a CA or client cert,
it is merely informational. Make sure your certs have unique
common names. -O Org name (cert field) An informational string
(company name) -o Output filename base (defaults to asterisk) -d
Output directory (defaults to the current directory) Example: To
create a CA and a server (pbx.mycompany.com) cert with output in
/tmp: ast_tls_cert -C pbx.mycompany.com -O "My Company" -d /tmp
This will create a CA cert and key as well as asterisk.pem and
the the two files that it is made from: asterisk.crt and
asterisk.key. Copy asterisk.pem and ca.crt somewhere (like
/etc/asterisk) and set tlscertfile=/etc/asterisk.pem and
tlscafile=/etc/ca.crt. Since this is a self-signed key, many
devices will require you to import the ca.crt file as a trusted
cert. To create a client cert using the CA cert created by the
example above: ast_tls_cert -m client -c /tmp/ca.crt -k
/tmp/ca.key -C "Joe User" -O \ "My Company" -d /tmp -o joe_user
This will create client.crt/key/pem in /tmp. Use this if your
device supports a client certificate. Make sure that you have the
ca.crt file set up as a tlscafile in the necessary Asterisk
configs. Make backups of all .key files in case you need them
later.
2010-10-22 15:47 +0000 [r292704] Richard Mudgett <rmudgett@digium.com>
* channels/sig_pri.c, main/channel.c, channels/chan_misdn.c:
Connected line is not updated when chan_dahdi/sig_pri or
chan_misdn transfers a call. When a call is transfered by ECT or
implicitly by disconnect in sig_pri or implicitly by disconnect
in chan_misdn, the connected line information is not exchanged.
The connected line interception macros also need to be executed
if defined. The CALLER interception macro is executed for the
held call. The CALLEE interception macro is executed for the
active/ringing call. JIRA ABE-2589 JIRA SWP-2296 Patches:
abe_2589_c3bier.patch uploaded by rmudgett (license 664)
abe_2589_v1.8_v2.patch uploaded by rmudgett (license 664) Review:
https://reviewboard.asterisk.org/r/958/
2010-10-21 22:09 +0000 [r292667] Tilghman Lesher <tlesher@digium.com>
* channels/misdn/ie.c: Compile correctly on Linux
(asterisk/localtime.h depends upon asterisk/autoconfig.h loading
first).
2010-10-21 18:13 +0000 [r292628] Paul Belanger <paul.belanger@polybeacon.com>
* contrib/init.d/rc.suse.asterisk: Fix typo in SUSE init script.
Reported by: Dave Cotton on asterisk-users list.
2010-10-21 16:14 +0000 [r292595] David Vossel <dvossel@digium.com>
* main/manager.c: Fixes recursive lock problem in manager.c It is
possible for a AMI session to freeze because of invalid use of
recursive locks during the EVENT processing. This patch removes
the unnecessary locks. (closes issue #18167) Reported by: sustav
Patches: manager_locking_v1.diff uploaded by dvossel (license
671) Tested by: sustav
2010-10-21 13:12 +0000 [r292557] Leif Madsen <lmadsen@digium.com>
* configs/res_ldap.conf.sample, /: Merged revisions 292556 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r292556 | lmadsen | 2010-10-21 08:11:52 -0500 (Thu, 21 Oct 2010)
| 6 lines Change res_ldap.sample.conf to match the schema.
(closes issue #17376) Reported by: jcovert Patches:
res_ldap.conf.sample.patch uploaded by jcovert (license 551)
........
2010-10-21 11:36 +0000 [r292523] Russell Bryant <russell@digium.com>
* res/res_config_ldap.c: Add var=value to log message on update
failure, and add newline. ... just for you, Leif.
2010-10-21 01:02 +0000 [r292489] Richard Mudgett <rmudgett@digium.com>
* channels/sig_pri.c: Send CONNECT_ACKNOWLEDGE for CIS calls too.
The originator of the Q.SIG call completion signaling link was
not changed to the active state when the CONNECT message came in.
The T309 processing would immediately kill the signaling link
because it was not in the active state.
2010-10-21 00:21 +0000 [r292413-292436] Paul Belanger <paul.belanger@polybeacon.com>
* apps/app_voicemail.c: Application not properly unregister in
voicemail (closes issue #18128) Reported by: junky Patches:
vm_unregister.diff uploaded by junky (license 177) Tested by:
pabelanger, lmadsen
* apps/app_dial.c, /: Merged revisions 292412 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r292412 | pabelanger | 2010-10-20 20:05:45 -0400
(Wed, 20 Oct 2010) | 17 lines Merged revisions 292411 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r292411 | pabelanger | 2010-10-20 20:00:51 -0400 (Wed, 20 Oct
2010) | 10 lines Record priv-recordintro as sln, not gsm This
removes the gsm->sln step when transcoding priv-recordintro.
(closes issue #18176) Reported by: pabelanger Patches:
chan_sip.diff uploaded by pabelanger (license 224) ........
................
2010-10-20 00:40 +0000 [r292376] Tilghman Lesher <tlesher@digium.com>
* res/res_musiconhold.c: Oops. This module uses the generic timer
and no longer uses DAHDI. This causes a problem with the Solaris
and other system builds that have gcc 4.1 (where optional_api is
non-optional).
2010-10-19 22:14 +0000 [r292343] Paul Belanger <paul.belanger@polybeacon.com>
* contrib/scripts/install_prereq: Add resample and imap_tk
dependencies.
2010-10-19 19:27 +0000 [r292309] Terry Wilson <twilson@digium.com>
* res/res_srtp.c, channels/chan_sip.c: Add sip show peer info about
crypto and remove dated comment This patch adds information about
the encryption setting to 'sip show peers' and removes an
out-of-date comment from res_srtp.c and instead directs users to
the proper documentation. (closes issue #18140) Reported by:
chodorenko
2010-10-21 Leif Madsen <lmadsen@digium.com>
* Asterisk 1.8.0 Released.
2010-10-18 Leif Madsen <lmadsen@digium.com>
* Asterisk 1.8.0-rc5 Released.
2010-10-18 22:02 +0000 [r292230] Leif Madsen <lmadsen@digium.com>
* sounds/Makefile, /: Merged revisions 292229 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r292229 | lmadsen | 2010-10-18 17:01:16 -0500 (Mon, 18 Oct 2010)
| 3 lines Fix typo in the sounds/Makefile. (Issue #17426)
........
2010-10-18 21:55 +0000 [r292227] Jeff Peeler <jpeeler@digium.com>
* apps/app_voicemail.c, /: Merged revisions 292226 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r292226 | jpeeler | 2010-10-18 16:54:38 -0500
(Mon, 18 Oct 2010) | 18 lines Merged revisions 292223 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r292223 | jpeeler | 2010-10-18 16:50:30 -0500 (Mon, 18 Oct 2010)
| 11 lines Fix improper operator key acceptance and clean up temp
recording files. This is a fix for when pressing the operator key
after recording an unavailable, busy, name, or temporary message
in mailbox options. The operator key should not be accepted here,
but should be allowed during the message recording. If the
operator key is pressed during ensure the file is saved or
deleted as apporopriate. Also, ensure removal of temporary
recorded files after an early hang up or when message acceptance
confirmation times out. ABE-2518 ........ ................
2010-10-18 21:51 +0000 [r292225] Leif Madsen <lmadsen@digium.com>
* sounds/sounds.xml, sounds/Makefile, /: Merged revisions 292224
via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r292224 | lmadsen | 2010-10-18 16:50:47 -0500
(Mon, 18 Oct 2010) | 17 lines Merged revisions 292222 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r292222 | lmadsen | 2010-10-18 16:47:25 -0500 (Mon, 18 Oct 2010)
| 9 lines Add support for the new English (Australian Accent)
sound files. (closes issue #17426) Reported by: camsown Patches:
core-sounds-en_AU.txt uploaded by camsown (license 1050)
add_AU_sounds.patch.txt uploaded by lmadsen (license 10) Tested
by: camsown, lmadsen, jtodd, qwell ........ ................
2010-10-18 19:50 +0000 [r292188] Russell Bryant <russell@digium.com>
* main/netsock2.c: Resolve some compiler errors in
ast_sockaddr_is_any(). These errors came up once this function
was used from within netsock2.c. The errors were like the
following: netsock2.c:393: error: dereferencing pointer
‘({anonymous})’ does break strict-aliasing rules The usage of a
union here avoids this problem.
2010-10-18 19:16 +0000 [r292155] David Vossel <dvossel@digium.com>
* main/netsock2.c: Fixes build error for systems not supporting
IPV6_TCLASS.
2010-10-18 17:15 +0000 [r292122] Matthew Nicholson <mnicholson@digium.com>
* addons/chan_mobile.c: Fix the cmgr parser. (closes issue 0018152)
Reported by: menschentier
2010-10-18 Leif Madsen <lmadsen@digium.com>
* Asterisk 1.8.0-rc4 Released
2010-10-18 16:02 +0000 [r292085] David Vossel <dvossel@digium.com>
* main/netsock2.c: Fixes qos settings for sockets bound to any IPv6
or IPv4 address. (closes issue #18099) Reported by: jamesnet
Patches: issues_18099_v3.diff uploaded by dvossel (license 671
2010-10-18 15:32 +0000 [r292083] Jeff Peeler <jpeeler@digium.com>
* pbx/pbx_spool.c: Disable use of inotify for call file handling as
it is not working properly. (related to #18089)
2010-10-16 10:47 +0000 [r292050] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
* res/res_musiconhold.c, /, configs/musiconhold.conf.sample: Merged
revisions 292049 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r292049 | tzafrir | 2010-10-16 12:03:04 +0200 (ש', 16 אוק 2010) |
15 lines Base directory for MOH should be ASTDATADIR If the
directive 'directory' is relative, make it relative to the
datadir, rather than to the varlibdir. In the sample
configuration it is relative ('moh'). This has no effect unless
you have actively set the datadir explicitly (at build time or at
run time). (closes issue #16906) Patches: moh_datadir uploaded by
tzafrir (license 46) Review:
https://reviewboard.asterisk.org/r/974/ ........
2010-10-15 21:40 +0000 [r292016] Terry Wilson <twilson@digium.com>
* res/res_srtp.c: Ref/unref res_srtp when we create/destroy a
session This avoids unhappy crashing when we try to 'core stop
gracefully' and res_srtp tries to unload before chan_sip does.
Thanks, Russell! (closes issue #18085) Reported by: st
2010-10-15 20:12 +0000 [r291942] David Vossel <dvossel@digium.com>
* channels/chan_sip.c: Fixes peer's host port information being
lost on sip reload. (closes issue #18135) Reported by: lmadsen
Patches: crazy_ports_v2.diff uploaded by dvossel (license 671)
Tested by: lmadsen
2010-10-15 19:50 +0000 [r291940] Paul Belanger <paul.belanger@polybeacon.com>
* configs/gtalk.conf.sample, /: Merged revisions 291939 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r291939 | pabelanger | 2010-10-15 15:35:20 -0400
(Fri, 15 Oct 2010) | 9 lines Merged revisions 291938 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r291938 | pabelanger | 2010-10-15 15:30:41 -0400 (Fri,
15 Oct 2010) | 2 lines Clean up formatting. ........
................
2010-10-15 16:39 +0000 [r291905] Terry Wilson <twilson@digium.com>
* res/res_jabber.c, /: Merged revisions 291904 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r291904 | twilson | 2010-10-15 09:16:57 -0700 (Fri, 15 Oct 2010)
| 7 lines Don't crash or deadlock on module unload We can't hold
the lock while pthread_join is called since aji_log_hook will
attempt to lock from the other therad. We reorder the
pthread_join and ast_aji_disconnect so that we don't do an
SSL_read() while SSL_shutdown is running, causing a crash.
........
2010-10-14 22:09 +0000 [r291827-291829] David Vossel <dvossel@digium.com>
* main/netsock2.c: Set TCLASS field of IPv6 header when sip qos
options are set. (closes issue #18099) Reported by: jamesnet
Patches: issues_18099_v2.diff uploaded by dvossel (license 671)
Tested by: dvossel, jamesnet
* channels/chan_gtalk.c: Safer xml parsing, treat all clients the
same, and better local candidate selection. The gtalk channel
driver was doing several unsafe operations in regards to how it
parsed incoming XML messages. I have cleaned that code up so it
should be much safer now. We now treat all clients types the
same. We have no reason to distinguish between GMAIL and GOOGLE
VOICE clients anymore because they all work the same way. I also
modified how the local ip is found. If no bindaddress is provided
in the config file, we attempt to determine the local ip we would
use to connect to google.com. If that fails, then we fall back to
the ast_find_ourip() function as a last resort. Using the new
method makes it much less likely that we would ever advertise a
local RTP candidate as a loopback address.
2010-10-14 18:45 +0000 [r291791] Jeff Peeler <jpeeler@digium.com>
* main/stdtime/localtime.c: Add missing ifdefs for test framework
and new locale code. (closes issue #18137) Reported by: ovi
Patches: 18137_test_framework_ifdef.patch uploaded by wdoekes
(license 717) 18137_localelist_warning.patch uploaded by wdoekes
(license 717) Tested by: ovi
2010-10-14 15:15 +0000 [r291758] Paul Belanger <paul.belanger@polybeacon.com>
* channels/chan_gtalk.c, channels/chan_jingle.c,
include/asterisk/acl.h, channels/chan_sip.c,
channels/chan_h323.c, main/acl.c: Add the ability for
ast_find_ourip to return IPv4, IPv6 or both. While testing
chan_gtalk I noticed jabber was using my IPv6 address and not
IPv4. When using bindaddr=0.0.0.0 it is possible for
ast_find_ourip() to return both IPv6 and IPv4 results. Adding a
family parameter gives you the ablility to choose. Since
jabber/gtalk/h323 do not support IPv6, we should only return IPv4
results. Review: https://reviewboard.asterisk.org/r/973/
2010-10-14 12:08 +0000 [r291725] Russell Bryant <russell@digium.com>
* doc/tex/secure-calls.tex: Fix a typo - s/seucre/secure/
2010-10-13 23:45 +0000 [r291656] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c, channels/sig_analog.c, /,
channels/sig_analog.h: Merged revisions 291655 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r291655 | rmudgett | 2010-10-13 18:36:50 -0500
(Wed, 13 Oct 2010) | 27 lines Merged revisions 291643 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r291643 | rmudgett | 2010-10-13 18:29:58 -0500 (Wed, 13 Oct 2010)
| 20 lines Deadlock between dahdi_exception() and
dahdi_indicate(). There is a deadlock between dahdi_exception()
and dahdi_indicate() for analog ports. The call-waiting and
three-way-calling feature can experience deadlock if these
features are trying to do something and an event from the bridged
channel happens at the same time. Deadlock avoidance code added
to obtain necessary channel locks before attemting an operation
with call-waiting and three-way-calling. (closes issue #16847)
Reported by: shin-shoryuken Patches: issue_16847_v1.4.patch
uploaded by rmudgett (license 664) issue_16847_v1.6.2.patch
uploaded by rmudgett (license 664) issue_16847_v1.8_v2.patch
uploaded by rmudgett (license 664) Tested by: alecdavis, rmudgett
Review: https://reviewboard.asterisk.org/r/971/ ........
................
2010-10-13 23:01 +0000 [r291581] Terry Wilson <twilson@digium.com>
* main/channel.c, /: Merged revisions 291580 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r291580 | twilson | 2010-10-13 15:58:43 -0700
(Wed, 13 Oct 2010) | 28 lines Merged revisions 291577 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r291577 | twilson | 2010-10-13 15:45:15 -0700 (Wed, 13 Oct 2010)
| 21 lines Don't ignore frames that have been queued when
softhangup'd When an outgoing call is answered and hung up by the
far end *very* quickly, we may not read any frames and therefor
end up with a call that displays the wrong
disposition/DIALSTATUS. The reason is because ast_queue_hangup()
immediately sets the _softhangup flag on the channel and then
queues the HANGUP control frame, but __ast_read refuses to read
any frames if ast_check_hangup() indicates that a hangup request
has been made (which it will if _softhangup is set). So, we end
up losing control frames. This change makes __ast_read continue
to read frames even if a soft hangup has been requested. It
queues a hangup frame to make sure that __ast_read() will still
eventually return NULL. Much thanks to David Vossel for all of
the reviews, discussion, and help! (closes issue #16946) Reported
by: davidw Review: https://reviewboard.asterisk.org/r/740/
........ ................
2010-10-13 22:46 +0000 [r291578] David Vossel <dvossel@digium.com>
* channels/chan_gtalk.c: More fixup for chan_gtalk. This patch
makes the xml parsing safer.
2010-10-13 22:24 +0000 [r291575] Terry Wilson <twilson@digium.com>
* Makefile, static-http/mantest.html (added): Add a simple AMI
client web page This patch uses the XML docs to parse all of the
available AMI commands and allows you to enter the command name
and be presented with a form with the available fields. You can
then rapidly tab through the fields and submit the command and
view the response. It is much faster/easier than having to use
telnet for testing purposes.
2010-10-13 20:21 +0000 [r291469-291541] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c: The chan_dahdi faxdetect option only works
for the first FAX call. The chan_dahdi faxdetect option only
works for the first call. After that the option no longer works.
The struct dahdi_pvt.callprogress member is the encoded user
config setting for the callprogress and faxdetect config options.
Changing this value alters the configuration for all following
calls until the chan_dahdi.conf file is reloaded. * Fixed the
chan_dahdi ast_channel_setoption callback to not change the users
faxdetect config setting except for the current call. * Fixed the
chan_dahdi ast_channel_queryoption callback to read the active
DSP setting of the faxdetect option. * Made actually disable the
active faxdetect DSP setting for the current call on the analog
port. my_handle_dtmfup() is used for normal analog ports.
dahdi_handle_dtmfup() is the legacy code and is no longer used
unless in a radio mode. (closes issue #18116) Reported by:
seandarcy Patches: issue18116_v1.8.patch uploaded by rmudgett
(license 664) Review: https://reviewboard.asterisk.org/r/972/
* channels/chan_misdn.c: Merged revision 291504 from
https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
.......... r291504 | rmudgett | 2010-10-13 13:30:21 -0500 (Wed,
13 Oct 2010) | 11 lines Hold off ast_hangup() from destroying the
ast_channel. Must get the ast_channel lock before proceeding with
release_chan() and release_chan_early() to hold off ast_hangup()
from destroying the ast_channel. Missed this change for -r291468.
JIRA ABE-2598 JIRA SWP-2317 ..........
* channels/chan_misdn.c: Merge revision 291468 from
https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
.......... r291468 | rmudgett | 2010-10-13 12:39:02 -0500 (Wed,
13 Oct 2010) | 16 lines Memory overwrites when releasing mISDN
call. Phone <--> Asterisk <-- ALERTING --> DISCONNECT <-- RELEASE
--> RELEASE_COMPLETE * Add lock protection around channel list
for find/add/delete operations. * Protect misdn_hangup() from
release_chan() and vise versa using the release_lock. JIRA
ABE-2598 JIRA SWP-2317 ..........
2010-10-13 15:46 +0000 [r291394] Russell Bryant <russell@digium.com>
* /, channels/chan_sip.c: Merged revisions 291393 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r291393 | russell | 2010-10-13 10:29:21 -0500
(Wed, 13 Oct 2010) | 13 lines Merged revisions 291392 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r291392 | russell | 2010-10-13 10:23:19 -0500 (Wed, 13 Oct 2010)
| 6 lines Lock pvt so pvt->owner can't disappear when queueing up
a frame. This fixes a crash due to a hangup race condition.
ABE-2601 ........ ................
2010-10-12 17:20 +0000 [r291284] Leif Madsen <lmadsen@digium.com>
* configs/phoneprov.conf.sample, /: Merged revisions 291280 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r291280 | lmadsen | 2010-10-12 12:20:02 -0500 (Tue, 12 Oct 2010)
| 7 lines Add undocumented variables to phoneprov.conf.sample
(closes issue #18107) Reported by: lathama Patches:
phoneprov.conf.sample.diff uploaded by lathama (license 1028)
........
2010-10-12 17:06 +0000 [r291265] Tilghman Lesher <tlesher@digium.com>
* /, main/acl.c: Merged revisions 291264 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r291264 | tilghman | 2010-10-12 12:05:31 -0500
(Tue, 12 Oct 2010) | 9 lines Merged revisions 291263 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r291263 | tilghman | 2010-10-12 11:55:30 -0500 (Tue, 12
Oct 2010) | 2 lines Oops, incorrect range (although unallocated
at ARIN) ........ ................
2010-10-12 16:08 +0000 [r291230] Leif Madsen <lmadsen@digium.com>
* configs/manager.conf.sample, /: Merged revisions 291229 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r291229 | lmadsen | 2010-10-12 11:07:28 -0500 (Tue, 12 Oct 2010)
| 2 lines Add documention that mentions options are defined but
not used. (Issue #18101) ........
2010-10-12 15:58 +0000 [r291192-291227] David Vossel <dvossel@digium.com>
* main/manager.c: Fixes manager.c crash. This issue was caused by
improper use of the mansession lock and manession_session lock.
These two structures are confusing to begin with so I'm not
surprised this occurred. I fixed this by consistently making sure
we use each of these locks only to protect the data in the
corresponding structure. We had mismatched usage of these locks
which resulted in no mutual exclusivity occurring at all. (closes
issue #17994) Reported by: vrban Patches:
mansession_locking_fix.diff uploaded by dvossel (license 671)
Tested by: vrban
* CHANGES: Update CHANGES to reflect new gtalk.conf options.
* channels/chan_gtalk.c, include/asterisk/stun.h,
configs/gtalk.conf.sample, res/res_stun_monitor.c: Gtalk
enhancements and general code cleanup. This patch includes
several chan_gtalk enhancements. Two new gtalk.conf options have
been added, externip and stunadd. Setting externip allows us to
manually specify what the external IP address is outside of a NAT
environment. Setting the stunaddr option to a valid stun server
allows for that external ip to be retrieved via a STUN server
automatically. This external IP is then advertised during call
setup as a possible candidate. I have also attempted to clean up
chan_gtalk's code so it meets our coding guidelines. During this
cleanup I noticed several things that need to be done in the code
and made a TODO section at the top of the file.
2010-10-11 18:51 +0000 [r291075-291113] Richard Mudgett <rmudgett@digium.com>
* channels/chan_sip.c: Move declaration closer to where now used.
* /, channels/chan_sip.c: Merged revisions 291110-291111 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r291110 | rmudgett | 2010-10-11 13:34:22 -0500
(Mon, 11 Oct 2010) | 9 lines Merged revisions 291109 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r291109 | rmudgett | 2010-10-11 13:29:43 -0500 (Mon, 11
Oct 2010) | 1 line Add missing unlock to an exception condition
in reload_config(). ........ ................ r291111 | rmudgett
| 2010-10-11 13:39:06 -0500 (Mon, 11 Oct 2010) | 1 line Make exit
from handle_request_do() consistent. ................
* main/cli.c, /: Merged revisions 291073 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r291073 | rmudgett | 2010-10-11 11:39:17 -0500 (Mon, 11 Oct 2010)
| 15 lines Fixed infinite loop in verbose/debug message output.
Setting the module/filename specific message level and then
changing it resulted in the linked list being looped on itself.
Traversing this linked list is an infinite loop if what you are
looking for is not in the list. Also plugged some CLI parsing
holes in the associated CLI command: * Removing a nonexistent
module from the list actually added it with a level of zero. *
Setting the non-module specific level to zero is now equivalent
to setting it to "off" as documented. ........
2010-10-09 23:25 +0000 [r291038] Tilghman Lesher <tlesher@digium.com>
* cdr/cdr_pgsql.c, configs/cdr_pgsql.conf.sample: Add missing
option to set calls to be logged in GMT/UTC.
2010-10-09 15:00 +0000 [r291005-291037] Alexandr Anikin <may@telecom-service.ru>
* addons/ooh323c/src/oochannels.c: small correction for verbose
print h.323 packets
* addons/ooh323c/src/ooh323.c, addons/chan_ooh323.c,
addons/ooh323c/src/ooh245.c: Added fast start and h.245 tunneling
options per user and peer. Added options for faststart/h.245
tunneling per user/peer, properly handle these and global
options, correction of handling fs/tunneling fields in signalling
responses (issue #17972) Reported by: salecha Patches:
fs-tunnel-per-point-3.patch uploaded by may213 (license 454)
Tested by: may213, salecha
2010-10-08 20:44 +0000 [r290973] David Vossel <dvossel@digium.com>
* channels/chan_gtalk.c: Make outbound Google Voice calls. This
patch allows for outbound Google Voice calls to be dialed from
Asterisk using chan_gtalk. Below is an example dialstring. exten
-> blah,1,Dial(Gtalk/asterisk/+15552225555@voice.google.com,,) In
this example, 'asterisk' is the jabber.conf profile configured to
connect to your gmail account. In order to receive Google Voice
calls make sure to enable 'allowguest=yes' in gtalk.conf.
2010-10-08 15:49 +0000 [r290937-290938] Erin Spiceland <erin@thespicelands.com>
* addons/res_config_mysql.c: Parentheses around assignment used as
truth value, introduced in r290937.
* addons/res_config_mysql.c, addons/app_mysql.c,
configs/res_config_mysql.conf.sample: Add option to
res_config_mysql and app_mysql to specify a character set that
MySQL should use. (closes issue 17948) Reported by qmax.
2010-10-08 02:56 +0000 [r290864] Jeff Peeler <jpeeler@digium.com>
* main/asterisk.c, /: Merged revisions 290863 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r290863 | jpeeler | 2010-10-07 21:45:44 -0500
(Thu, 07 Oct 2010) | 16 lines Merged revisions 290862 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r290862 | jpeeler | 2010-10-07 21:35:29 -0500 (Thu, 07 Oct 2010)
| 9 lines Ensure editline cleanup occurs when Ctrl-C is pressed
at control console. A recent change was made to avoid a race
condition on shutdown which only called the end functions from
the console thread. However, when pressing Ctrl-C the quit
handler is called from the signal handler thread. (closes issue
#17698) Reported by: jmls ........ ................
2010-10-07 22:38 +0000 [r290828-290829] David Vossel <dvossel@digium.com>
* channels/chan_gtalk.c: Add Philippe Sultan to chan_gtalk author
list. Philippe has made some notable contributions to the gtalk
channel driver. His name deserves to be listed amoung the authors
of that file. Thanks Philippe!
* channels/chan_gtalk.c: Outbound gtalk calls now work correctly.
There was a problem with how the candidates were being built on
an outbound call. This patch fixes that.
2010-10-07 20:58 +0000 [r290752] Jason Parker <jparker@digium.com>
* autoconf/ast_ext_lib.m4, /, configure,
include/asterisk/autoconfig.h.in: Merged revisions 290751 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r290751 | qwell | 2010-10-07 15:57:14 -0500
(Thu, 07 Oct 2010) | 16 lines Merged revisions 290750 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r290750 | qwell | 2010-10-07 15:56:04 -0500 (Thu, 07 Oct 2010) |
9 lines Allow PRI to build properly when using --with-pri. Use
the directories found for the parent when using lib dependencies.
(closes issue #17314) Reported by: tzafrir Patches:
17314-withdeps.diff uploaded by qwell (license 4) ........
................
2010-10-07 Leif Madsen <lmadsen@digium.com>
* Asterisk 1.8.0-rc3 Released.
2010-10-07 11:00 +0000 [r290713] Russell Bryant <russell@digium.com>
* main/pbx.c, /: Merged revisions 290712 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r290712 | russell | 2010-10-07 12:53:56 +0200 (Thu, 07 Oct 2010)
| 4 lines Don't crash when Set() is called without a value.
Review: https://reviewboard.asterisk.org/r/949/ ........
2010-10-06 21:22 +0000 [r290648-290674] David Vossel <dvossel@digium.com>
* channels/chan_gtalk.c: Fixes commented out code to use #if 0
instead. Thanks to rmudgett for catching this!
* channels/chan_gtalk.c: Fixes gtalk outbound DTMF to work
properly. Outbound DTMF with gtalk needs to be done within the
RTP stream. I discovered this after investigating a packet
capture from the gmail client. Instead of performing jingle
signaling DTMF, the gtalk servers expect all DTMF to arrive on
the RTP stream using RFC2833 way of doing things. Chan_gtalk also
had an issue with negotiating RTP payload type 106 for the
telephony-event and then sending DTMF as payload 101. This has
been resolved by always negotiating 101 as the payload type like
we do everywhere else. With this patch, incoming google voice
calls forwarded to Asterisk via gtalk work.
2010-10-06 18:50 +0000 [r290614] Richard Mudgett <rmudgett@digium.com>
* apps/app_dial.c: Merged revision 290613 from
https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
.......... r290613 | rmudgett | 2010-10-06 13:42:41 -0500 (Wed,
06 Oct 2010) | 5 lines Eliminate a redundant test for
AST_CONTROL_REDIRECTING. Eliminate redundant test for
AST_CONTROL_REDIRECTING that prevents running the redirecting
interception macro if it is defined. ..........
2010-10-06 13:49 +0000 [r290576] Tilghman Lesher <tlesher@digium.com>
* /, main/file.c: Merged revisions 290575 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r290575 | tilghman | 2010-10-06 08:48:27 -0500 (Wed, 06 Oct 2010)
| 8 lines Allow streaming audio from a pipe. (closes issue
#18001) Reported by: jamicque Patches:
20100926__issue18001.diff.txt uploaded by tilghman (license 14)
Tested by: jamicque ........
2010-10-06 04:35 +0000 [r290542] Terry Wilson <twilson@digium.com>
* res/res_rtp_asterisk.c: Don't try to send RTP when remote_address
is null It is possible for ast_rtp_stop() to be called which will
clear the remote address and cause the sendto to fail and spam
warnings. Don't send in this case.
2010-10-05 22:23 +0000 [r290479-290506] David Vossel <dvossel@digium.com>
* channels/chan_iax2.c: Fixes uninitialized memory problem in 'iax2
set debug peer' option.
* include/asterisk/jingle.h, channels/chan_gtalk.c,
res/res_jabber.c, include/asterisk/jabber.h: Fixes chan_gtalk to
work with gmail client This patch was written by Philippe Sultan
(phsultan). Thanks for keeping this up to date!
2010-10-05 20:23 +0000 [r290408] Tilghman Lesher <tlesher@digium.com>
* res/res_jabber.c, /: Merged revisions 290396 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r290396 | tilghman | 2010-10-05 15:21:02 -0500
(Tue, 05 Oct 2010) | 15 lines Merged revisions 290392 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r290392 | tilghman | 2010-10-05 15:20:07 -0500 (Tue, 05 Oct 2010)
| 8 lines Fix a crash by ensuring that we don't alter memory
after it's freed. (closes issue #17387) Reported by: jmls
Patches: 20100726__issue17387.diff.txt uploaded by tilghman
(license 14) Tested by: jmls ........ ................
2010-10-05 20:09 +0000 [r290376-290378] David Vossel <dvossel@digium.com>
* channels/chan_iax2.c: Resolves dnsmgr memory corruption in
chan_iax2. (closes issue #17902) Reported by: afried Patches:
issue_17902.rev1.txt uploaded by russell (license 2) Tested by:
afried, russell, dvossel Review:
https://reviewboard.asterisk.org/r/965/
* /, apps/app_directed_pickup.c: Merged revisions 290375 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r290375 | dvossel | 2010-10-05 14:54:50 -0500 (Tue, 05 Oct 2010)
| 10 lines Fixes PickupChan() not working with full channel name.
(closes issue #18011) Reported by: schern Patches:
app_directed_pickup.c.2.patch uploaded by schern (license 995)
app_directed_pickup.c.trunk.patch uploaded by schern (license
995) Tested by: schern, dvossel ........
2010-10-05 14:15 +0000 [r290066-290289] Tilghman Lesher <tlesher@digium.com>
* configure, configure.ac: Restore run directory for OS X, as well
as standardizing some other paths to Mac OS X.
* pbx/ael/ael-test/ref.ael-test4, pbx/ael/ael-test/ref.ael-test5,
pbx/ael/ael-test/ref.ael-test19,
pbx/ael/ael-test/ref.ael-vtest13, res/ael/pval.c, main/pbx.c,
pbx/ael/ael-test/ref.ael-vtest17, /,
pbx/ael/ael-test/ref.ael-ntest10, pbx/ael/ael-test/ref.ael-test1,
pbx/ael/ael-test/ref.ael-test2, pbx/ael/ael-test/ref.ael-test3:
Merged revisions 290254 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r290254 | tilghman | 2010-10-04 18:14:59 -0500 (Mon, 04 Oct 2010)
| 11 lines Change new pattern matcher to regard dashes the same
as the old pattern matcher -- as visual candy to be ignored. Also
change the AEL parser to not generate dashes within extensions,
as those dashes would be ignored. Update the AEL tests to match
this behavior. (closes issue #17366) Reported by: murf Patches:
20100727__issue17366.diff.txt uploaded by tilghman (license 14)
Tested by: tilghman ........
* /, configure, configure.ac: Merged revisions 290201 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r290201 | tilghman | 2010-10-04 15:22:03 -0500
(Mon, 04 Oct 2010) | 9 lines Merged revisions 290177 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r290177 | tilghman | 2010-10-04 15:15:26 -0500 (Mon, 04
Oct 2010) | 2 lines Fixing Mac OS X auto-builder. ........
................
* /, configure, configure.ac: Merged revisions 290101 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r290101 | tilghman | 2010-10-03 16:06:58 -0500
(Sun, 03 Oct 2010) | 9 lines Merged revisions 290100 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r290100 | tilghman | 2010-10-03 16:04:29 -0500 (Sun, 03
Oct 2010) | 2 lines Automatically re-run configure test for
menuselect, when the relevant makeopts settings change. ........
................
* pbx/pbx_spool.c: Get notification only when file is closed, not
when created. (closes issue #17924) Reported by: mkeuter Patches:
asterisk-1.8-bugid17924.patch uploaded by abelbeck (license 946)
Tested by: abelbeck
2010-10-02 17:57 +0000 [r290026] Kevin P. Fleming <kpfleming@digium.com>
* contrib/scripts/get_mp3_source.sh: Allow users to pass additional
arguments to the Subversion command that obtains the MP-3 source
code. (reported on IRC by jmls)
2010-10-02 08:56 +0000 [r289951] Olle Johansson <oej@edvina.net>
* main/manager.c, /: Merged revisions 289950 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r289950 | oej | 2010-10-02 10:52:03 +0200 (Lör,
02 Okt 2010) | 9 lines Merged revisions 289949 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r289949 | oej | 2010-10-02 10:50:05 +0200 (Lör, 02 Okt 2010) | 2
lines Add documentation for undocumented option to AMI action
originate ........ ................
2010-10-02 04:46 +0000 [r289875] Tilghman Lesher <tlesher@digium.com>
* apps/app_voicemail.c, /: Merged revisions 289874 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r289874 | tilghman | 2010-10-01 23:45:49 -0500
(Fri, 01 Oct 2010) | 15 lines Merged revisions 289873 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r289873 | tilghman | 2010-10-01 23:42:08 -0500 (Fri, 01 Oct 2010)
| 8 lines When forwarding a message, a prepend means that the
filesystem will always have a better copy. (closes issue #17803)
Reported by: dpetersen Patches: 20100923__issue17803.diff.txt
uploaded by tilghman (license 14) Tested by: dpetersen ........
................
2010-10-02 02:43 +0000 [r289840] Jeff Peeler <jpeeler@digium.com>
* include/asterisk/rtp_engine.h, res/res_rtp_asterisk.c,
main/rtp_engine.c, /, channels/chan_sip.c: Merged revisions
289798 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r289798 | jpeeler | 2010-10-01 18:01:31 -0500
(Fri, 01 Oct 2010) | 22 lines Merged revisions 289797 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r289797 | jpeeler | 2010-10-01 17:58:38 -0500 (Fri, 01 Oct 2010)
| 15 lines Change RFC2833 DTMF event duration on end to report
actual elapsed time. The scenario here is with a non P2P early
media session. The reported time length of DTMF presses are
coming up short when sending to the remote side. Currently the
event duration is a running total that is incremented when
sending continuation packets. These continuation packets are only
triggered upon incoming media from the remote side, which means
that the running total probably is not going to end up matching
the actual length of time Asterisk received DTMF. This patch
changes the end event duration to be lengthened if it is detected
that the end event is going to come up short. Review:
https://reviewboard.asterisk.org/r/957/ ABE-2476 ........
................
2010-10-01 17:19 +0000 [r289718] Paul Belanger <paul.belanger@polybeacon.com>
* res/res_jabber.c, /, configs/jabber.conf.sample: Merged revisions
289704 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r289704 | pabelanger | 2010-10-01 13:09:03 -0400
(Fri, 01 Oct 2010) | 13 lines Merged revisions 289703 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r289703 | pabelanger | 2010-10-01 13:03:11 -0400 (Fri, 01 Oct
2010) | 6 lines Disable debugging by default and reformat .config
file. Review: https://reviewboard.asterisk.org/r/929/ ........
................
2010-10-01 16:22 +0000 [r289701] Jeff Peeler <jpeeler@digium.com>
* /, channels/chan_sip.c: Merged revisions 289700 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r289700 | jpeeler | 2010-10-01 11:21:04 -0500
(Fri, 01 Oct 2010) | 21 lines Merged revisions 289699 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r289699 | jpeeler | 2010-10-01 11:20:00 -0500 (Fri, 01 Oct 2010)
| 14 lines Ensure user portion of SIP URI matches dialplan when
using encoded characters. This commit takes a simliar approach to
288112 and checks the dialplan to determine the proper action for
an incoming contact header as to whether or not it should be
decoded or not. sip_new was blindly always decoding the
extension, which also caused the outgoing contact header to be
incorrect as well as failing to match the encoded extension in
the dialplan. (closes issue #17892) Reported by: wdoekes Patches:
bug17892-1.patch uploaded by jpeeler (license 325) Tested by:
wdoekes ........ ................
2010-10-01 09:42 +0000 [r289622] Stefan Schmidt <sst@sil.at>
* channels/chan_sip.c: don't iterate through all dialogs to find
and delete old subscribes On every incoming subscribe there is a
iteration through all dialogs to find old subscribes and delete
them. This is slow and not RFC conform. This was only needed in
1.2 cause a subscribe was not deleted when a dialog was
destroyed, after 1.4 a subscribe get removed when its dialog is
destroyed. (closes issue #17950) Reported by: schmidts Tested by:
schmidts Review: https://reviewboard.asterisk.org/r/901/
2010-09-30 20:23 +0000 [r289581] Tilghman Lesher <tlesher@digium.com>
* funcs/func_env.c: Solaris fixes.
2010-09-30 19:53 +0000 [r289554] Matthew Nicholson <mnicholson@digium.com>
* /, channels/chan_sip.c: Merged revisions 289553 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r289553 | mnicholson | 2010-09-30 14:51:27 -0500 (Thu, 30 Sep
2010) | 4 lines Properly handle channel allocation failures duing
invites with replaces. ABE-2588 ........
2010-09-30 19:28 +0000 [r289549] Richard Mudgett <rmudgett@digium.com>
* channels/chan_misdn.c: Merged revision 289547 from
https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
.......... r289547 | rmudgett | 2010-09-30 14:16:36 -0500 (Thu,
30 Sep 2010) | 10 lines In chan_misdn, the
DivertingLegInformation2 DivertingNr is garbage when the number
is restricted. The same thing happens with
DivertingLegInformation1 DivertedTo number. The
misdn_PresentedNumberUnscreened_extract() extracted the
Unscreened PartyNumber field unconditionally. It now checks the
presented number unscreened type to see if the PartyNumber was
even present. JIRA ABE-2595 ..........
2010-09-30 17:50 +0000 [r289543] Tilghman Lesher <tlesher@digium.com>
* include/asterisk/localtime.h, main/stdtime/localtime.c,
tests/test_time.c, tests/test_utils.c, res/res_agi.c: More
Solaris compatibility fixes
2010-09-30 15:39 +0000 [r289426] Russell Bryant <russell@digium.com>
* apps/app_sms.c, /: Merged revisions 289425 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r289425 | russell | 2010-09-30 10:37:29 -0500
(Thu, 30 Sep 2010) | 15 lines Merged revisions 289424 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r289424 | russell | 2010-09-30 10:34:29 -0500 (Thu, 30 Sep 2010)
| 8 lines Fix a crash in app_sms. Since the data being passed to
the generator callback is on the stack of the SMS() application,
we must ensure that the generator is stopped before the
application exits. ABE-2587 ........ ................
2010-09-29 21:12 +0000 [r289340] Jason Parker <jparker@digium.com>
* main/channel.c, /, main/features.c: Merged revisions 289339 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r289339 | qwell | 2010-09-29 16:03:47 -0500
(Wed, 29 Sep 2010) | 15 lines Merged revisions 289338 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r289338 | qwell | 2010-09-29 15:56:26 -0500 (Wed, 29 Sep 2010) |
8 lines Allow a manager originate to succeed on forwarded
devices. The timeout to wait for an answer was being set to 0
when a device forwarded to another extension. We don't always
need the timeout set like this, so make it an optional parameter,
and don't use it in this case. ABE-2544 ........ ................
2010-09-29 20:27 +0000 [r289336] Leif Madsen <lmadsen@digium.com>
* configs/res_ldap.conf.sample, /: Merged revisions 289334 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r289334 | lmadsen | 2010-09-29 15:24:47 -0500 (Wed, 29 Sep 2010)
| 1 line Update sample documentation to note md5secret
requirements. ........
2010-09-29 20:20 +0000 [r289333] Russell Bryant <russell@digium.com>
* res/res_config_ldap.c, /: Merged revisions 289332 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
........ r289332 | russell | 2010-09-29 15:15:57 -0500 (Wed, 29
Sep 2010) | 4 lines Don't completely ignore md5secret from LDAP
if the value does not begin with {md5}. This fixes a problem that
lmadsen ran in to where md5secret was not working for him.
........
2010-09-29 17:53 +0000 [r289268-289300] Matthew Nicholson <mnicholson@digium.com>
* configs/res_fax.conf.sample: Add 'ecm' to the sample fax config
file
* main/channel.c: Update the CDR record when
ast_channel_set_caller_event() is called (related to issue
#17569) Reported by: tbelder
2010-09-29 16:16 +0000 [r289253] Richard Mudgett <rmudgett@digium.com>
* main/channel.c: Make development error message indicate which
channel.
2010-09-29 15:04 +0000 [r289179] Matthew Nicholson <mnicholson@digium.com>
* main/channel.c, /: Merged revisions 289178 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r289178 | mnicholson | 2010-09-29 10:04:11 -0500
(Wed, 29 Sep 2010) | 15 lines Merged revisions 289177 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r289177 | mnicholson | 2010-09-29 10:03:27 -0500 (Wed, 29 Sep
2010) | 8 lines Set the caller id on CDRs when it is set on the
parent channel. (closes issue #17569) Reported by: tbelder
Patches: 17569.diff uploaded by tbelder (license 618) Tested by:
tbelder ........ ................
2010-09-28 18:18 +0000 [r289104] Tilghman Lesher <tlesher@digium.com>
* makeopts.in, apps/app_voicemail.c, Makefile, tests/test_time.c,
configure, include/asterisk/autoconfig.h.in,
include/asterisk/compat.h, main/strcompat.c, tests/test_utils.c,
configure.ac: Solaris compatibility fixes Review:
https://reviewboard.asterisk.org/r/942/
2010-09-28 18:18 +0000 [r289099] Brett Bryant <bbryant@digium.com>
* main/channel.c, /: Merged revisions 289095 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r289095 | bbryant | 2010-09-28 14:14:19 -0400
(Tue, 28 Sep 2010) | 21 lines Merged revisions 289094 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r289094 | bbryant | 2010-09-28 14:10:19 -0400 (Tue, 28 Sep 2010)
| 14 lines Fixes an issue with the Newchannel AMI event during
the Masquerading process. Fixes an issue with the Newchannel AMI
event during the Masquerading process, where no Newchannel AMI
event was generated for the psuedo channel used during the
masquerading process. (closes issue #17987) Reported by:
RadicAlish Patches: newchannel.patch.txt uploaded by RadicAlish
(license 1122) Tested by: RadicAlish Review:
https://reviewboard.asterisk.org/r/937/ ........ ................
2010-09-28 01:04 +0000 [r289054-289057] Richard Mudgett <rmudgett@digium.com>
* channels/sig_pri.c: Avoid deadlock processing incoming AOC-E
messages. Deadlock avoidance for the owner channel was not done
when processing incoming AOC-E messages.
* channels/sig_pri.c: Revert stuff not ready for commit in
-r289054.
* channels/sig_pri.c, channels/chan_sip.c: Break up long
ast_manager_event_multichan() event lines.
2010-09-27 18:37 +0000 [r288961] Tilghman Lesher <tlesher@digium.com>
* channels/chan_sip.c: Still build SIP, even if res_crypto cannot
be built (use, not depend). (closes issue #18062) Reported by: a
user on the mailing list
2010-09-27 13:03 +0000 [r288925-288927] Russell Bryant <russell@digium.com>
* res/res_agi.c: Fix some documentation typos and spelling errors.
* res/res_agi.c: Fix a documentation spelling error.
2010-09-24 17:58 +0000 [r288821-288852] David Vossel <dvossel@digium.com>
* channels/chan_sip.c: Append Retry-After header on 500 error
response to Re-INVITE according to RFC3261 section 14.2. ABE-2301
* channels/chan_sip.c: Inspect Require header on BYE transaction
according to RFC3261 section 8.2.2.3. ABE-2293
2010-09-24 16:02 +0000 [r288748] Terry Wilson <twilson@digium.com>
* channels/chan_local.c, /: Merged revisions 288747 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r288747 | twilson | 2010-09-24 08:37:39 -0700
(Fri, 24 Sep 2010) | 12 lines Merged revisions 288746 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r288746 | twilson | 2010-09-24 08:26:09 -0700 (Fri, 24 Sep 2010)
| 5 lines Don't fail a masquerade if it is already being hung up
This avoids noise on some Local channel situations where we don't
use /n. Thanks to Alec Davis for the suggestion. ........
................
2010-09-24 13:54 +0000 [r288606-288713] Tilghman Lesher <tlesher@digium.com>
* /, funcs/func_strings.c: Merged revisions 288712 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
........ r288712 | tilghman | 2010-09-24 08:53:30 -0500 (Fri, 24
Sep 2010) | 5 lines Solaris won't printf a NULL. (closes issue
#18041) Reported by: asgaroth ........
* main/asterisk.exports.in: Export timersub for platforms which do
not have it
* include/asterisk/channel.h, cdr/cdr_pgsql.c, /, configure,
include/asterisk/autoconfig.h.in, include/asterisk/compat.h,
main/strcompat.c, configure.ac: Merged revisions 288637 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r288637 | tilghman | 2010-09-23 22:36:01 -0500
(Thu, 23 Sep 2010) | 9 lines Merged revisions 288636 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r288636 | tilghman | 2010-09-23 22:20:24 -0500 (Thu, 23
Sep 2010) | 2 lines Solaris compatibility fixes ........
................
* CHANGES: Add note about the checkhangup option of ${CHANNEL()}
2010-09-23 Leif Madsen <lmadsen@digium.com>
* Asterisk 1.8.0-rc2 Released.
2010-09-23 18:05 +0000 [r288507-288572] Terry Wilson <twilson@digium.com>
* main/manager.c: Make AMI honor enabled=no (closes issue #18040)
Reported by: twilson Review:
https://reviewboard.asterisk.org/r/938/
* channels/chan_local.c, /: Merged revisions 288500 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r288500 | twilson | 2010-09-22 16:10:09 -0700
(Wed, 22 Sep 2010) | 15 lines Merged revisions 288499 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r288499 | twilson | 2010-09-22 16:00:30 -0700 (Wed, 22 Sep 2010)
| 8 lines Don't let a Local channel get bridged to itself If a
local channel gets bridged to itself, it becomes orphaned with no
devices left to actually tell it to hang up. This patch modifies
local_fixup() to detect this case and deny it. Review:
https://reviewboard.asterisk.org/r/934 ........ ................
2010-09-22 Leif Madsen <lmadsen@digium.com>
* Asterisk 1.8.0-rc1 Released.
2010-09-22 17:49 +0000 [r288345-288418] David Vossel <dvossel@digium.com>
* /, channels/chan_sip.c: Merged revisions 288417 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r288417 | dvossel | 2010-09-22 12:49:05 -0500
(Wed, 22 Sep 2010) | 11 lines Merged revisions 288416 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r288416 | dvossel | 2010-09-22 12:48:15 -0500 (Wed, 22 Sep 2010)
| 5 lines RFC3261 section 12.2 explicitly says out of order
requests are responded with a 500 Server Internal Error response.
ABE-2458 ........ ................
* /, channels/chan_sip.c: Merged revisions 288344 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r288344 | dvossel | 2010-09-22 11:53:28 -0500
(Wed, 22 Sep 2010) | 9 lines Merged revisions 288343 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r288343 | dvossel | 2010-09-22 11:49:56 -0500 (Wed, 22
Sep 2010) | 2 lines During check_pendings, if the dialog is
terminated with a CANCEL, change the invitestate to INV_CANCEL
like in sip_hangup. ........ ................
2010-09-22 16:45 +0000 [r288341] Russell Bryant <russell@digium.com>
* main/asterisk.c, /: Merged revisions 288340 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r288340 | russell | 2010-09-22 11:44:13 -0500
(Wed, 22 Sep 2010) | 18 lines Merged revisions 288339 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r288339 | russell | 2010-09-22 11:39:16 -0500 (Wed, 22 Sep 2010)
| 11 lines Fix a 100% CPU consumption problem when setting
console=yes in asterisk.conf. The handling of -c and console=yes
should be the same, but they were not. When you specify -c, it
sets both a flag for console module and for asterisk not to
fork() off into the background. The handling of console=yes only
set console mode, so you would end up with a background process()
trying to run the Asterisk console and freaking out since it
didn't have anything to read input from. Thanks to beagles for
reporting and helping debug the problem! ........
................
2010-09-22 15:14 +0000 [r288268] Tilghman Lesher <tlesher@digium.com>
* UPGRADE.txt, cdr/cdr_pgsql.c, configs/cdr_pgsql.conf.sample, /:
Merged revisions 288267 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r288267 | tilghman | 2010-09-22 10:11:09 -0500
(Wed, 22 Sep 2010) | 23 lines Merged revisions 288265-288266 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r288265 | tilghman | 2010-09-22 09:48:04 -0500 (Wed, 22 Sep 2010)
| 9 lines Allow the encoding to be set, in case local charset
does not agree with database. (closes issue #16940) Reported by:
jamicque Patches: 20100827__issue16940.diff.txt uploaded by
tilghman (license 14) 20100921__issue16940__1.6.2.diff.txt
uploaded by tilghman (license 14) Tested by: jamicque ........
r288266 | tilghman | 2010-09-22 10:04:52 -0500 (Wed, 22 Sep 2010)
| 5 lines Document addition of encoding parameter. (issue #16940)
Reported by: jamicque ........ ................
2010-09-22 00:06 +0000 [r288194] Richard Mudgett <rmudgett@digium.com>
* channels/chan_iax2.c, /: Merged revisions 288193 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r288193 | rmudgett | 2010-09-21 19:03:37 -0500
(Tue, 21 Sep 2010) | 33 lines Merged revisions 288192 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r288192 | rmudgett | 2010-09-21 18:55:58 -0500 (Tue, 21 Sep 2010)
| 26 lines In chan_iax2.c:schedule_delivery() calls
ast_bridged_channel() on an unlocked channel. Near the beginning
of schedule_delivery(), ast_bridged_channel() is called on
iaxs[fr->callno]->owner. However, the channel is not locked,
which can result in ast_bridged_channel() crashing should
owner->tech change to a technology that doesn't implement
bridged_channel. I also fixed the other calls to
ast_bridged_channel() in chan_iax2.c since the owner lock was not
held there either. Converted the existing channel deadlock
avoidance to use iax2_lock_owner(). Using the new function
simplified some awkward code. In the process of fixing the
locking on ast_bridged_channel(), I also found a memory leak in
socket_process() for v1.6.2 and v1.8. The local struct variable
ies.vars is not freed on early/abnormal function exits. (closes
issue #17919) Reported by: rain Patches: issue17919_v1.4.patch
uploaded by rmudgett (license 664) issue17919_w_leak_v1.6.2.patch
uploaded by rmudgett (license 664) issue17919_w_leak_v1.8.patch
uploaded by rmudgett (license 664) Review:
https://reviewboard.asterisk.org/r/926/ ........ ................
2010-09-21 22:57 +0000 [r288159] Tilghman Lesher <tlesher@digium.com>
* /, channels/chan_sip.c: Merged revisions 288113 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r288113 | tilghman | 2010-09-21 16:59:46 -0500
(Tue, 21 Sep 2010) | 22 lines Merged revisions 288112 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r288112 | tilghman | 2010-09-21 16:58:13 -0500 (Tue, 21 Sep 2010)
| 15 lines Try both the encoded and unencoded subscription URI
for a match in hints. When a phone sends an encoded URI for a
subscription, the URI is not matched with the actual hint that is
in decoded format. For example, if we have an extension with a
hint that is named: "#5601" or "*5601", the subscription will
work fine if the phone subscribes with an already decoded URI,
but when it's decoded like "%255601" or "%2A5601", Asterisk is
unable to match it with the correct hint. (closes issue #17785)
Reported by: ramonpeek Patches: 20100831__issue17785.diff.txt
uploaded by tilghman (license 14) Tested by: ramonpeek ........
................
2010-09-21 22:26 +0000 [r288157] Paul Belanger <paul.belanger@polybeacon.com>
* channels/chan_iax2.c, /: Merged revisions 288147 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
........ r288147 | pabelanger | 2010-09-21 18:22:43 -0400 (Tue,
21 Sep 2010) | 9 lines Setup timer before set_config(). (closes
issue #18019) Reported by: Netview Patches: issue_0018019.patch
uploaded by pabelanger (license 224) Tested by: Netview ........
2010-09-21 21:03 +0000 [r288079-288082] Richard Mudgett <rmudgett@digium.com>
* doc/tex/partymanip.tex: Add note in party manipulation chapter on
interception macros.
* apps/app_queue.c, apps/app_dial.c: Simplify locking code for
REDIRECTING interception macro when forwarding a call. Simplified
the locking code by using a local copy of the redirecting party
information in app_dial.c:do_forward() and
app_queue.c:wait_for_answer() for launching the REDIRECTING
interception macro when a call is forwarded. Reduced the lock
time of the 'o->chan' and 'in' channels.
* main/channel.c: Protect channel access in CONNECTED_LINE and
REDIRECTING interception macro launch code.
2010-09-21 19:48 +0000 [r288007] Brett Bryant <bbryant@digium.com>
* main/channel.c, /: Merged revisions 288006 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r288006 | bbryant | 2010-09-21 15:46:20 -0400
(Tue, 21 Sep 2010) | 14 lines Merged revisions 288005 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r288005 | bbryant | 2010-09-21 15:43:46 -0400 (Tue, 21 Sep 2010)
| 8 lines Add a check to fix a rare segmentation fault you'd get
if ast_frdup couldn't allocate memory on the first frame being
queued in ast_queue_frame. (closes issue #17882) Reported by:
seanbright Tested by: seanbright ........ ................
2010-09-21 19:08 +0000 [r287935] Tilghman Lesher <tlesher@digium.com>
* main/asterisk.c, /: Merged revisions 287934 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r287934 | tilghman | 2010-09-21 14:07:53 -0500
(Tue, 21 Sep 2010) | 9 lines Merged revisions 287933 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r287933 | tilghman | 2010-09-21 14:07:07 -0500 (Tue, 21
Sep 2010) | 2 lines Less than zero is an error, not any non-zero
value. ........ ................
2010-09-21 19:02 +0000 [r287931] Terry Wilson <twilson@digium.com>
* main/channel.c: Revert change in favor of a more targeted fix
2010-09-21 18:32 +0000 [r287929] David Vossel <dvossel@digium.com>
* channels/chan_sip.c: Send a "415 Unsupported Media Type" after
failure to process sdp due to unknown Content-Encoding header.
ABE-2258
2010-09-21 15:53 +0000 [r287897] Richard Mudgett <rmudgett@digium.com>
* main/features.c: Cut-n-paste error in builtin_blindtransfer().
2010-09-21 15:43 +0000 [r287895] Russell Bryant <russell@digium.com>
* res/res_rtp_asterisk.c, main/dnsmgr.c, channels/chan_sip.c,
main/acl.c: Don't use ast_strdupa() from within the arguments to
a function. (closes issue #17902) Reported by: afried Patches:
issue_17902.rev1.txt uploaded by russell (license 2) Tested by:
russell Review: https://reviewboard.asterisk.org/r/927/
2010-09-21 15:24 +0000 [r287893] Tilghman Lesher <tlesher@digium.com>
* channels/chan_sip.c: Anonymous callerid needs a "sip:" uri
prefix. (closes issue #17981) Reported by: avalentin Patches:
sip-anonymous-aastra.patch uploaded by avalentin (license 1107)
(plus an additional fix by me) Tested by: avalentin
2010-09-21 13:41 +0000 [r287863] Russell Bryant <russell@digium.com>
* main/logger.c: Fix a regression in verbose logger processing.
2010-09-21 04:37 +0000 [r287833] Terry Wilson <twilson@digium.com>
* main/channel.c: Don't generate connected line buffer twice for
comparison
2010-09-21 00:00 +0000 [r287760] Brett Bryant <bbryant@digium.com>
* /, apps/app_meetme.c: Merged revisions 287759 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r287759 | bbryant | 2010-09-20 19:58:26 -0400
(Mon, 20 Sep 2010) | 23 lines Merged revisions 287758 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r287758 | bbryant | 2010-09-20 19:57:08 -0400 (Mon, 20 Sep 2010)
| 16 lines Fix misvalidation of meetme pins in conjunction with
the 'a' MeetMe flag. When using the 'a' MeetMe flag and having a
user and admin pin setup for your conference, using the user pin
would gain you admin priviledges. Also, when no user pin was set,
an admin pin was, the 'a' MeetMe flag wasn't used, and the user
tried to enter a conference then they were still prompted for a
pin and forced to hit #. (closes issue #17908) Reported by: kuj
Patches: pins_2.patch uploaded by kuj (license 1111) Tested by:
kuj Review: [full review board URL with trailing slash] ........
................
2010-09-20 23:51 +0000 [r287757] Terry Wilson <twilson@digium.com>
* main/channel.c: Avoid infinite loop with certain local channel
connected line updates Compare connected line data before sending
a connected line indication to avoid possible loops. Review:
https://reviewboard.asterisk.org/r/932/
2010-09-20 23:20 +0000 [r287701] Alec L Davis <sivad.a@paradise.net.nz>
* main/channel.c, /: Merged revisions 287685 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r287685 | alecdavis | 2010-09-21 11:16:45 +1200 (Tue, 21 Sep
2010) | 18 lines ast_channel_masquerade: Avoid recursive
masquerades. Check all 4 combinations of (original/clonechan) *
(masq/masqr). Initially original->masq and clonechan->masqr were
only checked. It's possible with multiple masq's planned - and
not yet executed, that the 'original' chan could already have
another masq'd into it - thus original->masqr would be set, that
masqr would lost. Likewise for the clonechan->masq. (closes issue
#16057;#17363) Reported by: amorsen;davidw,alecdavis Patches:
based on bug16057.diff4.txt uploaded by alecdavis (license 585)
Tested by: ramonpeek, davidw, alecdavis ........
2010-09-20 23:14 +0000 [r287683] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c: The inalarm flag was not set in sig_analog
struct if the port is initially in alarm. Fixed initial inalarm
value for sig_analog ports. Along with -r261007, this gets the
inalarm flag in sync with chan_dahdi for sig_analog ports.
(closes issue #16983)
2010-09-20 22:21 +0000 [r287661] Alec L Davis <sivad.a@paradise.net.nz>
* main/channel.c: ast_do_masquerade. Keep channels ao2_container
locked while unlink and linking channels. Previously, Masquerade
would unlock 'original' and 'clonechan' and allow another masq
thread to run. End result would be corrupted memory, and the
frequent report 'Bad Magic Number'. (closes issue #17801,#17710)
Reported by: notthematrix Patches: Based on bug17801.diff1.txt
uploaded by alecdavis (license 585) Tested by: alecdavis Review:
https://reviewboard.asterisk.org/r/928
2010-09-20 22:09 +0000 [r287645-287647] David Vossel <dvossel@digium.com>
* include/asterisk/channel.h, CHANGES, include/asterisk/framehook.h
(added), main/channel.c, main/framehook.c (added),
funcs/func_frame_trace.c (added): Addition of the FrameHook API
(AKA AwesomeHooks) So far all our tools for viewing and
manipulating media streams within Asterisk have been entirely
focused on audio. That made sense then, but is not scalable now.
The FrameHook API lets us tap into and manipulate _ANY_ type of
media or signaling passed on a channel present today or in the
future. This tool is a step in the direction of expanding
Asterisk's boundaries and will help generate some rather
interesting applications in the future. In addition to the
FrameHook API, a simple dialplan function exercising the api has
been included as well. This function is called FRAME_TRACE().
FRAME_TRACE() allows for the internal ast_frames read and written
to a channel to be output. Filters can be placed on this function
to debug only certain types of frames. This function could be
thought of as an internal way of doing ast_frame packet captures.
Review: https://reviewboard.asterisk.org/r/925/
* channels/chan_sip.c: Fixes issue with registrations not working
properly with pedantic=yes. (closes issue #18017) Reported by:
schmidts Patches: issues_18017_v1.diff uploaded by dvossel
(license 671) Tested by: schmidts
2010-09-20 21:29 +0000 [r287643] Jason Parker <jparker@digium.com>
* /, channels/chan_skinny.c: Merged revisions 287642 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
........ r287642 | qwell | 2010-09-20 16:28:32 -0500 (Mon, 20 Sep
2010) | 8 lines Don't crash when parking a non-bridged call.
(closes issue #17680) Reported by: jmhunter Patches:
chan_skinny-park-v1.txt uploaded by DEA (license 3) Tested by:
jmhunter, DEA ........
2010-09-20 21:19 +0000 [r287639] Brett Bryant <bbryant@digium.com>
* main/logger.c: Fixes an error with the logger that caused verbose
messages to be spammed to the screen if syslog was configured in
logger.conf (closes issue #17974) Reported by: lmadsen Review:
https://reviewboard.asterisk.org/r/915/
2010-09-20 15:57 +0000 [r287559] Matthew Nicholson <mnicholson@digium.com>
* main/pbx.c, /: Merged revisions 287558 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r287558 | mnicholson | 2010-09-20 10:56:21 -0500
(Mon, 20 Sep 2010) | 14 lines Use ast_str when processing hint
state changes Merged revisions 287555 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r287555 | mnicholson | 2010-09-20 10:48:14 -0500 (Mon, 20 Sep
2010) | 5 lines Use ast_dynamic_str when processing hint state
changes (related to issue #17928) Reported by: mdu113 ........
................
2010-09-19 16:09 +0000 [r287471] Olle Johansson <oej@edvina.net>
* main/manager.c, /: Merged revisions 287470 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r287470 | oej | 2010-09-19 18:06:10 +0200 (Sön,
19 Sep 2010) | 14 lines Merged revisions 287469 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r287469 | oej | 2010-09-19 17:56:50 +0200 (Sön, 19 Sep 2010) | 7
lines Make sure we always free variables properly in manager
originate. (closes issue #17891) reported, solved and tested by
oej Review: https://reviewboard.asterisk.org/r/869/ ........
................
2010-09-17 21:08 +0000 [r287388] Tilghman Lesher <tlesher@digium.com>
* apps/app_queue.c, /: Merged revisions 287387 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r287387 | tilghman | 2010-09-17 16:08:00 -0500
(Fri, 17 Sep 2010) | 14 lines Merged revisions 287386 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r287386 | tilghman | 2010-09-17 16:06:03 -0500 (Fri, 17 Sep 2010)
| 7 lines Blank columns should get set on reload, not ignored.
(closes issue #16893) Reported by: haakon Patches:
20100818__issue16893.diff.txt uploaded by tilghman (license 14)
........ ................
2010-09-17 13:37 +0000 [r287309] Matthew Nicholson <mnicholson@digium.com>
* main/pbx.c, /: Merged revisions 287308 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r287308 | mnicholson | 2010-09-17 08:36:07 -0500
(Fri, 17 Sep 2010) | 12 lines Merged revisions 287307 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r287307 | mnicholson | 2010-09-17 08:34:34 -0500 (Fri, 17 Sep
2010) | 5 lines Use ast_strdup() instead of ast_strdupa() while
processing in ast_hint_state_changed(). (related to issue #17928)
Reported by: mdu113 ........ ................
2010-09-17 08:44 +0000 [r287269-287271] Jan Kalab <pitlicek@gmail.com>
* res/res_calendar_ews.c: Events are visible after they were
removed from EWS calendar Because we must merge calendar even
when it's empty. (closes issue #17786)
* res/res_calendar_ews.c: Asterisk crashing because of double free
when EWS request fails The free is done later in code. I think
ast_free() should have built in checks for double free. (closes
issue #17782)
* res/res_calendar_caldav.c, res/res_calendar_ews.c,
res/res_calendar_exchange.c, res/res_calendar_icalendar.c:
Support for HTTP redirects in calendar's URL libneon does not
support HTTP redirects (3xx responses) by default. You must tell
it to follow them. Also, another little unsigned int fix. (closes
issue #17776) Review: https://reviewboard.asterisk.org/r/921/
2010-09-16 22:04 +0000 [r287195] Jason Parker <jparker@digium.com>
* contrib/init.d/rc.debian.asterisk: Don't fail when running the
Debian init script directly (as one would normally do). readlink
apparently returns 1 when the arg isn't a symlink, which caused
the script to exit. (closes issue #17910) Reported by: wurstsalat
2010-09-16 21:57 +0000 [r287193] Russell Bryant <russell@digium.com>
* UPGRADE.txt, apps/app_queue.c, configs/queues.conf.sample: Set
the default for "autofill" and "shared_lastcall" to "yes" in
queues.conf. Review: https://reviewboard.asterisk.org/r/922/
2010-09-16 20:07 +0000 [r287116-287120] Matthew Nicholson <mnicholson@digium.com>
* main/pbx.c, /: Merged revisions 287119 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r287119 | mnicholson | 2010-09-16 15:06:16 -0500
(Thu, 16 Sep 2010) | 15 lines Merged revisions 287118 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r287118 | mnicholson | 2010-09-16 15:04:46 -0500 (Thu, 16 Sep
2010) | 8 lines Don't limit hint processing in
ast_hint_state_changed() to AST_MAX_EXTENSION length strings.
(closes issue #17928) Reported by: mdu113 Patches:
20100831__issue17928.diff.txt uploaded by tilghman (license 14)
Tested by: mdu113 ........ ................
* main/cdr.c, /: Merged revisions 287115 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r287115 | mnicholson | 2010-09-16 14:53:41 -0500
(Thu, 16 Sep 2010) | 15 lines Merged revisions 287114 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r287114 | mnicholson | 2010-09-16 14:52:39 -0500 (Thu, 16 Sep
2010) | 8 lines Don't stop printing cdr variables if we encounter
one with a blank name or value. (closes issue #17900) Reported
by: under Patches: core-show-channel-cdr-fix1.diff uploaded by
mnicholson (license 96) Tested by: mnicholson ........
................
2010-09-15 22:17 +0000 [r287056] Terry Wilson <twilson@digium.com>
* res/res_srtp.c: Don't hang up a call on an SRTP unprotect failure
Also make it more obvious when there is an issue en/decrypting.
(closes issue #17563) Reported by: Alexcr Patches:
res_srtp.c.patch uploaded by sfritsch (license 1089) Tested by:
twilson
2010-09-15 20:58 +0000 [r287020] Jeff Peeler <jpeeler@digium.com>
* main/features.c: fix uninintialized variable
2010-09-15 20:53 +0000 [r287017] Richard Mudgett <rmudgett@digium.com>
* channels/misdn/isdn_msg_parser.c, channels/chan_misdn.c: Merged
revision 287014 from
https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
.......... r287014 | rmudgett | 2010-09-15 15:32:24 -0500 (Wed,
15 Sep 2010) | 58 lines The handling of call transfer signaling
for mISDN PTMP is not fully implemented. The handling of call
transfer signaling for mISDN PTMP is not fully implemented. The
signaling of number updates with ISDN/DSS1 ECT supplementary
services (ETS 300 369-1) comes along with a notification
indicator IE and redirection number IE for PTMP. The
implementation in the current Asterisk mISDN channel
unfortunately can handle these information elements only in a
NOTIFY message. These information elements are also signaled in a
FACILTY message with a RequestSubaddress facility, when the
subscriber is already in the active state (see 9.2.4 and 9.2.5 of
ETS 300 369-1). ********** abe_2526_ast.patch * Added support to
handle the notification indicator IE and redirection number IE
with the RequestSubaddress facility. * Made
misdn_update_connected_line() send a NOTIFY message if Asterisk
originated the call and it is not connected yet. * Made
misdn_update_connected_line() send a FACILITY message if the call
is already connected. This patch requires the presence of the
associated mISDN patches to compile. I had to enhance mISDN to
allow the notification indicator IE and the redirection number IE
to be used with a FACILITY message. Earlier versions of the
Digium enhanced mISDN are no longer going to work. **********
abe_2526_misdn.patch * Made an incoming FACILITY message allow
the presence of the notification indicator IE and the redirection
number IE. ********** abe_2526_misdnuser_v3.patch * Added support
to send and receive a FACILITY message with the notification
indicator IE and the redirection number IE. * Added the ability
to send a NOTIFY message in PTMP/NT mode to all responding
subcalls in Q.931 states 6, 7, 8, 9, and 25. ********** Patches:
abe_2526_ast.patch uploaded by rmudgett (license 664)
abe_2526_misdn.patch uploaded by rmudgett (license 664)
abe_2526_misdnuser_v3.patch uploaded by rmudgett (license 664)
Tested by: rmudgett and reporter JIRA SWP-2146 JIRA ABE-2526
..........
2010-09-15 20:32 +0000 [r286931-287015] Jeff Peeler <jpeeler@digium.com>
* apps/app_voicemail.c, /: Merged revisions 286998 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r286998 | jpeeler | 2010-09-15 15:28:02 -0500
(Wed, 15 Sep 2010) | 14 lines Merged revisions 286941 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r286941 | jpeeler | 2010-09-15 15:08:52 -0500 (Wed, 15 Sep 2010)
| 7 lines Ensure mailbox is not filled to capacity before doing
message forwarding. Specifically, before prompting to record a
prepended message the capacity is checked first. If the mailbox
is full the extension will be reprompted. ABE-2517 ........
................
* CHANGES, channels/chan_iax2.c, channels/sip/include/sip.h,
configs/features.conf.sample, channels/chan_mgcp.c,
include/asterisk/features.h, channels/chan_dahdi.c,
channels/sig_analog.c, channels/chan_sip.c, main/features.c: Add
parking extension for non-default parking lots. This is a new
feature that allows for parking to custom parking lots to be
accessed directly, rather than with channel variables or by
changing the default parking lot. The extension is set with the
parkext option just as the default parking lot is done. Also, the
manager action has been updated to optionally allow a specified
parking lot. (closes issue #14882) Reported by: vmikhnevych
Patches: patch_14882.txt uploaded by mnick (license 874) modified
by me Review: https://reviewboard.asterisk.org/r/884/
2010-09-15 18:29 +0000 [r286904-286905] Richard Mudgett <rmudgett@digium.com>
* channels/sig_analog.c: Simplify some code in sig_analog.
* channels/sig_analog.c: Unable to originate calls using E&M over
T1. When originating a call from Unit Under Test to Reference
Unit using E&M RBS signaling mode, I get the following warning
message: "Ring/Off-hook in strange state 3 on channel 1". Fixed
the sig_analog outgoing flag. It was never set when sig_analog
was extracted from chan_dahdi. JIRA SWP-2191 JIRA AST-408
2010-09-15 13:05 +0000 [r286868] Matthew Nicholson <mnicholson@digium.com>
* channels/chan_sip.c: Set tohost to the domain specified in the
configuration file instead of the IP address of the host we are
calling. This fixes a regression introduced in r274783. (closes
issue #17960) Reported by: adriavidal Patches:
sip-tohost-fix1.diff uploaded by mnicholson (license 96) Tested
by: mich, mnicholson, adriavidal (closes issue #17676) Reported
by: outcast Patches: sip-tohost-fix1.diff uploaded by mnicholson
(license 96) Tested by: mnicholson
2010-09-14 21:57 +0000 [r286834] David Vossel <dvossel@digium.com>
* channels/chan_sip.c: Sets subscribed type for outgoing MWI
subscriptions so correct Event header is used.
2010-09-14 19:28 +0000 [r286682-286758] Matthew Nicholson <mnicholson@digium.com>
* /, channels/chan_sip.c: Merged revisions 286757 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r286757 | mnicholson | 2010-09-14 14:27:28 -0500
(Tue, 14 Sep 2010) | 20 lines Merged revisions 286756 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r286756 | mnicholson | 2010-09-14 14:26:18 -0500 (Tue, 14 Sep
2010) | 13 lines Don't clear the username from a realtime
database when a registration expires. Non-realtime chan_sip does
not clear the username from memory when a registration expiries
so realtime probably shouldn't either. (closes issue #17551)
Reported by: ricardolandim Patches:
reg-expiry-username-1.4-fix1.diff uploaded by mnicholson (license
96) reg-expiry-username-1.6.2-fix1.diff uploaded by mnicholson
(license 96) reg-expiry-username-1.8-fix1.diff uploaded by
mnicholson (license 96) reg-expiry-username-trunk-fix1.diff
uploaded by mnicholson (license 96) Tested by: ricardolandim,
mnicholson ........ ................
* main/channel.c, /: Merged revisions 286681 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r286681 | mnicholson | 2010-09-14 13:02:24 -0500
(Tue, 14 Sep 2010) | 14 lines Merged revisions 286679 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r286679 | mnicholson | 2010-09-14 13:00:01 -0500 (Tue, 14 Sep
2010) | 7 lines Only drop duplicate answer frames if the channel
is bridged. Back in r3710 ast_read() was modified to drop answer
frames on channels that were in the UP state. This modification
prevented bridges that were up before the answer from being
broken and reestablished by an ANSWER control frame. That change
also prevents pickup of channels called from the ast_dial
framework from working properly. The ast_dial framework expects
to see an ANSWER frame after dialing and the pickup code queues
one but ast_read() drops it. This new change only drops ANSWER
frames when the channel is bridged, allowing the answer queued by
the pickup code to properly pass through ast_read() on to the
ast_dial framework. ABE-2473 (related to issue #2342) ........
................
2010-09-14 15:30 +0000 [r286647] Richard Mudgett <rmudgett@digium.com>
* doc/tex/channelvariables.tex, doc/tex/partymanip.tex: Corrected
documented CONNECTED_LINE and REDIRECTING party manipulation
macro names.
2010-09-14 06:55 +0000 [r286617] Jan Kalab <pitlicek@gmail.com>
* res/res_calendar_ews.c: Merging events for Exchange web service
doesn't work as expected, resulting in only one event in calendar
The solution is to use "global" counter of events, since we do
new requests for every event and calendar sync after every
request. So now we do sync only after last request. (closes issue
#17877) Review: https://reviewboard.asterisk.org/r/916/
2010-09-14 05:07 +0000 [r286528-286588] Tilghman Lesher <tlesher@digium.com>
* contrib/realtime/mysql/voicemail_data.sql (added), /,
contrib/realtime/mysql/voicemail_messages.sql (added): Merged
revisions 286587 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r286587 | tilghman | 2010-09-14 00:06:05 -0500 (Tue, 14 Sep 2010)
| 2 lines Add documentation on missing backend tables for
Voicemail ........
* /, main/features.c: Merged revisions 286557 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r286557 | tilghman | 2010-09-13 18:48:51 -0500 (Mon, 13 Sep 2010)
| 2 lines C precedence got me ........
* /, main/features.c: Merged revisions 286527 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r286527 | tilghman | 2010-09-13 18:03:26 -0500 (Mon, 13 Sep 2010)
| 2 lines Refactor conversion to ast_poll() to fix callparking
regression. ........
2010-09-13 19:40 +0000 [r286457] Jason Parker <jparker@digium.com>
* /, channels/chan_sip.c: Merged revisions 286456 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r286456 | qwell | 2010-09-13 14:38:35 -0500 (Mon, 13 Sep 2010) |
5 lines Remove "Internal IP" from sip show settings, as it's not
at all useful to display. (closes issue #17840) Reported by: oej
........
2010-09-13 15:52 +0000 [r286426] Richard Mudgett <rmudgett@digium.com>
* configs/chan_dahdi.conf.sample: Update chan_dahdi.conf.sample to
reflect new libpri T309 default value.
2010-09-11 17:09 +0000 [r286270] Olle Johansson <oej@edvina.net>
* /, main/file.c: Merged revisions 286268 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r286268 | oej | 2010-09-11 19:05:16 +0200 (Lör,
11 Sep 2010) | 11 lines Merged revisions 286267 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r286267 | oej | 2010-09-11 18:59:20 +0200 (Lör, 11 Sep 2010) | 4
lines Handle error response when we can't make file compatible
Review: https://reviewboard.asterisk.org/r/911/ ........
................
2010-09-10 22:04 +0000 [r286189] Terry Wilson <twilson@digium.com>
* include/asterisk/channel.h, include/asterisk/pbx.h,
include/asterisk/frame.h, channels/chan_local.c,
funcs/func_channel.c: Merged revisions 286115 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r286115 | twilson | 2010-09-10 15:35:25 -0500
(Fri, 10 Sep 2010) | 23 lines Merged revisions 286059 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r286059 | twilson | 2010-09-10 14:25:08 -0500 (Fri, 10 Sep 2010)
| 16 lines Inherit CHANNEL() writes to both sides of a Local
channel Having Local (/n) channels as queue members and setting
the language in the extension with Set(CHANNEL(language)=fr) sets
the language on the Local/...,2 channel. Hold time report
playbacks happen on the Local/...,1 channel and therefor do not
play in the specified language. This patch modifies
func_channel_write to call the setoption callback and pass the
CHANNEL() write info to the callback. chan_local uses this
information to look up the other side of the channel and apply
the same changes to it. (closes issue #17673) Reported by:
Guggemand Review: https://reviewboard.asterisk.org/r/903/
........ ................
2010-09-10 21:11 +0000 [r286120] Paul Belanger <paul.belanger@polybeacon.com>
* channels/chan_iax2.c, /: Merged revisions 286117 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r286117 | pabelanger | 2010-09-10 16:55:06 -0400
(Fri, 10 Sep 2010) | 11 lines Merged revisions 286114 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r286114 | pabelanger | 2010-09-10 16:35:08 -0400 (Fri, 10 Sep
2010) | 4 lines Load iax.conf before registering any
functions/applications/actions. Review:
https://reviewboard.asterisk.org/r/914/ ........ ................
2010-09-10 20:55 +0000 [r286118] Richard Mudgett <rmudgett@digium.com>
* channels/sig_pri.c, /: Merged revisions 286116 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r286116 | rmudgett | 2010-09-10 15:42:44 -0500
(Fri, 10 Sep 2010) | 18 lines Merged revisions 286113 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r286113 | rmudgett | 2010-09-10 15:33:16 -0500 (Fri, 10 Sep 2010)
| 11 lines An outgoing call may not get hung up if a pre-connect
incoming ISDN call is disconnected. If the ISDN link a
pre-connect incoming call is using fails or is reset, the
outgoing leg may not hang up or be delayed in hanging up.
(Causes: PRI_CAUSE_NETWORK_OUT_OF_ORDER,
PRI_CAUSE_DESTINATION_OUT_OF_ORDER, and
PRI_CAUSE_NORMAL_TEMPORARY_FAILURE.) Just hang up the call if the
incoming call leg hangs up before connecting for any reason. It
makes no sense to send a BUSY or CONGESTION control frame to the
outgoing call leg under these circumstances. ........
................
2010-09-10 20:31 +0000 [r286112] Russell Bryant <russell@digium.com>
* main/db.c: Rate limit calls to fsync() to 1 per second after
astdb updates. Astdb was determined to be one of the most
significant bottlenecks in SIP registration processing. This
patch improved the speed of an astdb load test by 50000% (yes,
Fifty-Thousand Percent). On this particular load test setup, this
doubled the number of SIP registrations the server could handle.
Review: https://reviewboard.asterisk.org/r/825/
2010-09-10 18:31 +0000 [r286025] Tilghman Lesher <tlesher@digium.com>
* /: Merged revisions 286024 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r286024 | tilghman | 2010-09-10 13:30:21 -0500
(Fri, 10 Sep 2010) | 9 lines Merged revisions 286023 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r286023 | tilghman | 2010-09-10 13:22:04 -0500 (Fri, 10
Sep 2010) | 2 lines Missing newline ........ ................
2010-09-10 13:13 +0000 [r285992] David Ruggles <thedavidfactor@gmail.com>
* doc/externalivr.txt, CHANGES: Added missing documentation for
ExternalIVR feature added in January 2010
2010-09-10 05:32 +0000 [r285931-285962] Tilghman Lesher <tlesher@digium.com>
* include/asterisk/select.h, /: Merged revisions 285961 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r285961 | tilghman | 2010-09-10 00:31:31 -0500 (Fri, 10 Sep 2010)
| 6 lines Another fix for Mac OS X. While trying to fix this the
"right" way, I wandered into dependency hell. Two hours later, I
backed out, and just removed the offending code. ast_inline_api
only goes one level deep and then it breaks. Ouch. ........
* tests/test_poll.c, include/asterisk/select.h, /, configure,
include/asterisk/autoconfig.h.in, configure.ac: Merged revisions
285930 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r285930 | tilghman | 2010-09-09 20:16:32 -0500
(Thu, 09 Sep 2010) | 14 lines Merged revisions 285889 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r285889 | tilghman | 2010-09-09 19:13:45 -0500 (Thu, 09 Sep 2010)
| 7 lines Fix Mac OS X build. This also fixes a rather grievous
calculation error for the offset of ast_fdset, which was masked
on Linux and FreeBSD, because these platforms check the first 256
FDs regardless of the bitmask setting (due to backwards
compatibility). ........ ................
2010-09-09 22:52 +0000 [r285819] Paul Belanger <paul.belanger@polybeacon.com>
* /, codecs/gsm/Makefile: Merged revisions 285818 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r285818 | pabelanger | 2010-09-09 18:49:19 -0400
(Thu, 09 Sep 2010) | 15 lines Merged revisions 285817 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r285817 | pabelanger | 2010-09-09 18:34:35 -0400 (Thu, 09 Sep
2010) | 8 lines GCC 4.2.x optimizations result in improper
behavior of GSM codec (closes issue #17688) Reported by:
pprindeville Patches: asterisk-trunk-bugid11243.patch uploaded by
pprindeville (license 347) Tested by: mkeuter, pprindeville
........ ................
2010-09-09 20:11 +0000 [r285745] Jason Parker <jparker@digium.com>
* main/channel.c, /: Merged revisions 285744 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r285744 | qwell | 2010-09-09 15:09:23 -0500
(Thu, 09 Sep 2010) | 16 lines Merged revisions 285742 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r285742 | qwell | 2010-09-09 15:06:31 -0500 (Thu, 09 Sep 2010) |
9 lines Transmit silence when reading DTMF in ast_readstring.
Otherwise, you could get issues with DTMF timeouts causing
hangups. (closes issue #17370) Reported by: makoto Patches:
channel-readstring-silence-generator.patch uploaded by makoto
(license 38) ........ ................
2010-09-09 18:51 +0000 [r285640-285711] Brett Bryant <bbryant@digium.com>
* main/pbx.c, /: Merged revisions 285710 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r285710 | bbryant | 2010-09-09 14:50:13 -0400 (Thu, 09 Sep 2010)
| 8 lines Fixes an issue with dialplan pattern matching where the
specificity for pattern ranges and pattern special characters was
inconsistent. (closes issue #16903) Reported by: Nick_Lewis
Patches: pbx.c-specificity.patch uploaded by Nick Lewis (license
657) Tested by: Nick_Lewis ........
* res/res_musiconhold.c, /: Merged revisions 285639 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r285639 | bbryant | 2010-09-09 13:22:25 -0400
(Thu, 09 Sep 2010) | 14 lines Merged revisions 285638 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r285638 | bbryant | 2010-09-09 13:20:17 -0400 (Thu, 09 Sep 2010)
| 7 lines Fixes an issue with MOH where it doesn't recover
cleanly when it can't play a file and would just stop, instead of
continuing to find the next playable file in the MOH class.
(closes issue #17807) Reported by: kshumard Review:
https://reviewboard.asterisk.org/r/910/ ........ ................
2010-09-08 22:14 +0000 [r285564-285568] David Vossel <dvossel@digium.com>
* /, channels/chan_sip.c: Merged revisions 285567 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r285567 | dvossel | 2010-09-08 17:11:28 -0500
(Wed, 08 Sep 2010) | 9 lines Merged revisions 285566 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r285566 | dvossel | 2010-09-08 17:07:31 -0500 (Wed, 08
Sep 2010) | 2 lines In retrans_pkt, do not unlock pvt until the
end of the function on a transmit failure. ........
................
* /, channels/chan_sip.c: Merged revisions 285563 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r285563 | dvossel | 2010-09-08 16:47:29 -0500 (Wed, 08 Sep 2010)
| 54 lines Fixes interoperability problems with session timer
behavior in Asterisk. CHANGES: 1. Never put "timer" in "Require"
header. This is not to our benefit and RFC 4028 section 7.1 even
warns against it. It is possible for one endpoint to perform
session-timer refreshes while the other endpoint does not support
them. If in this case the end point performing the refreshing
puts "timer" in the Require field during a refresh, the dialog
will likely get terminated by the other end. 2. Change the
behavior of 'session-timer=accept' in sip.conf (which is the
default behavior of Asterisk with no session timer configuration
specified) to only run session-timers as result of an incoming
INVITE request if the INVITE contains an "Session-Expires"
header... Asterisk is currently treating having the "timer"
option in the "Supported" header as a request for session timers
by the UAC. I do not agree with this. Session timers should only
be negotiated in "accept" mode when the incoming INVITE supplies
a "Session-Expires" header, otherwise RFC 4028 says we should
treat a request containing no "Session-Expires" header as a
session with no expiration. Below I have outlined some situations
and what Asterisk's behavior is. The table reflects the behavior
changes implemented by this patch. SITUATIONS: -Asterisk as UAS
1. Incoming INVITE: NO "Session-Expires" 2. Incoming INVITE: HAS
"Session-Expires" -Asterisk as UAC 3. Outgoing INVITE: NO
"Session-Expires". 200 Ok Response HAS "Session-Expires" header
4. Outgoing INVITE: NO "Session-Expires". 200 Ok Response NO
"Session-Expires" header 5. Outgoing INVITE: HAS
"Session-Expires". Active - Asterisk will have an active refresh
timer regardless if the other endpoint does. Inactive - Asterisk
does not have an active refresh timer regardless if the other
endpoint does. XXXXXXX - Not possible for mode.
______________________________________ |SITUATIONS |
'session-timer' MODES | |___________|________________________| |
| originate | accept | |-----------|------------|-----------| |1.
| Active | Inactive | |2. | Active | Active | |3. | XXXXXXXX |
Active | |4. | XXXXXXXX | Inactive | |5. | Active | XXXXXXXX |
-------------------------------------- (closes issue #17005)
Reported by: alexrecarey ........
2010-09-08 20:58 +0000 [r285533] Brett Bryant <bbryant@digium.com>
* /, apps/app_meetme.c: Merged revisions 285532 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r285532 | bbryant | 2010-09-08 16:56:12 -0400 (Wed, 08 Sep 2010)
| 8 lines Fixes a bug with MeetMe where after announcing the
amount of time left in a conference, if music on hold was
playing, it doesn't restart. (closes issue #17408) Reported by:
sysreq Patches: asterisk-issue-17408_fixed.patch uploaded by
sysreq (license 1009) Tested by: sysreq ........
2010-09-08 20:43 +0000 [r285527-285530] Jason Parker <jparker@digium.com>
* res/res_musiconhold.c, /, include/asterisk/astobj2.h: Merged
revisions 285529 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r285529 | qwell | 2010-09-08 15:42:44 -0500 (Wed, 08 Sep 2010) |
1 line Follow coding guidelines in moh rescan fix. Also fix the
documentation that got me in trouble. ........
* res/res_musiconhold.c, /: Merged revisions 285526 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
........ r285526 | qwell | 2010-09-08 15:31:43 -0500 (Wed, 08 Sep
2010) | 8 lines Fixes issue where moh files were no longer
rescanned during a reload. (closes issue #16744) Reported by: pj
Patches: 16744-reload.diff uploaded by qwell (license 4) Tested
by: qwell ........
2010-09-08 07:14 +0000 [r285484] Tilghman Lesher <tlesher@digium.com>
* funcs/func_channel.c: Documentation only
2010-09-07 22:22 +0000 [r285455] Jason Parker <jparker@digium.com>
* channels/chan_sip.c: Don't automatically add domains for wildcard
bindaddrs. (closes issue #17832) Reported by: oej Patches:
17832-wildcard.diff uploaded by qwell (license 4) Tested by:
qwell
2010-09-07 21:20 +0000 [r285373-285386] Tilghman Lesher <tlesher@digium.com>
* pbx/pbx_spool.c: Don't notify on attribute changes, and change
how the queuing mechanism works. Fixes call spools in 1.8.
(closes issue #17337) Reported by: loloski Patches:
20100827__issue17337.diff.txt uploaded by tilghman (license 14)
(closes issue #17924) Reported by: mkeuter Tested by: mkeuter
* funcs/func_channel.c: Add CHANNEL(checkhangup) to check whether a
channel is in the process of being hanged up. (closes issue
#17652) Reported by: kobaz Patches: func_channel.patch uploaded
by kobaz (license 834)
2010-09-07 21:08 +0000 [r285371] Richard Mudgett <rmudgett@digium.com>
* main/features.c: Fix cut-n-paste error.
2010-09-07 20:58 +0000 [r285369] Jason Parker <jparker@digium.com>
* channels/chan_sip.c: Add note to 'sip show settings' regarding
dual-stack support, and a :: bindaddress. (closes issue #17831)
Reported by: oej Patches: 17831-v6wildcardbind.diff uploaded by
qwell (license 4)
2010-09-07 20:56 +0000 [r285268-285367] Tilghman Lesher <tlesher@digium.com>
* pbx/pbx_config.c, /: Merged revisions 285366 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r285366 | tilghman | 2010-09-07 15:31:41 -0500
(Tue, 07 Sep 2010) | 16 lines Merged revisions 285365 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r285365 | tilghman | 2010-09-07 15:30:22 -0500 (Tue, 07 Sep 2010)
| 9 lines Catch invalid extensions at the parser, instead of
making the core deal with them. (closes issue #17794) Reported
by: PavelL Patches: 20100820__issue17794__1.6.2.diff.txt uploaded
by tilghman (license 14) 20100820__issue17794__1.4.diff.txt
uploaded by tilghman (license 14) Tested by: PavelL ........
................
* include/asterisk/compiler.h, addons/ooh323c/src/ooSocket.h: Fix
build on FreeBSD 8.0, take 2.
* main/poll.c, /: Merged revisions 285267 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r285267 | tilghman | 2010-09-07 14:07:17 -0500
(Tue, 07 Sep 2010) | 11 lines Merged revisions 285266 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r285266 | tilghman | 2010-09-07 14:04:50 -0500 (Tue, 07 Sep 2010)
| 4 lines Use poll, if indicated to do so, in the ast_poll2
implementation. This fixes the unit tests on FreeBSD 8.0.
........ ................
2010-09-07 17:54 +0000 [r285197] Brett Bryant <bbryant@digium.com>
* apps/app_voicemail.c, /: Merged revisions 285196 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r285196 | bbryant | 2010-09-07 13:49:07 -0400
(Tue, 07 Sep 2010) | 17 lines Merged revisions 285194 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r285194 | bbryant | 2010-09-07 13:45:41 -0400 (Tue, 07 Sep 2010)
| 10 lines Fixes voicemail.conf issues where mailboxes with
passwords that don't precede a comma would throw unnecessary
error messages. (closes issue #15726) Reported by: 298 Patches:
M15726.diff uploaded by junky (license 177) Tested by: junky
Review: [full review board URL with trailing slash] ........
................
2010-09-07 17:47 +0000 [r285195] Richard Mudgett <rmudgett@digium.com>
* channels/chan_misdn.c: Merged revisions 285193 via svnmerge from
https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
........ Merged revisions 285192 via svnmerge from
https://origsvn.digium.com/svn/asterisk/be/branches/C.3 ........
r285192 | rmudgett | 2010-09-07 11:58:57 -0500 (Tue, 07 Sep 2010)
| 8 lines COLP/CONP and chan_misdn missing update chan_misdn does
not update the caller id of the channel if a new connected number
or ECT-INFORM (w/ new peer number on call transfer) is received.
JIRA ABE-2502 JIRA SWP-2058 ........ ........
2010-09-06 20:10 +0000 [r285161-285162] Russell Bryant <russell@digium.com>
* configure: regenerate configure script.
* include/asterisk/autoconfig.h.in, configure.ac: Fix libsrtp -fPIC
check for when non-standard prefix is used. Thanks to loompek in
#asterisk for reporting the issue and testing this patch.
2010-09-06 06:56 +0000 [r285090] Tilghman Lesher <tlesher@digium.com>
* BSDmakefile (added), makeopts.in, /: Merged revisions 285089 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r285089 | tilghman | 2010-09-06 01:55:17 -0500
(Mon, 06 Sep 2010) | 9 lines Merged revisions 285088 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r285088 | tilghman | 2010-09-06 01:54:18 -0500 (Mon, 06
Sep 2010) | 2 lines Silly convenience script for BSD platforms.
........ ................
2010-09-04 18:08 +0000 [r285057] Russell Bryant <russell@digium.com>
* include/asterisk/cli.h: Add a C++ compatible version of
AST_CLI_DEFINE().
2010-09-03 23:19 +0000 [r285017] Terry Wilson <twilson@digium.com>
* channels/chan_sip.c: Call correct lock function as transferer is
a sip_pvt not a channel Both functions are #defined to ao2_lock,
but still...
2010-09-03 22:21 +0000 [r285006] David Vossel <dvossel@digium.com>
* configs/sip.conf.sample, channels/sip/include/sip.h,
channels/chan_sip.c: Disables auth_options_request option by
default. The auth_options_request option was created to do
authentication on OPTIONS request just like INVITES are done.
Since it has been noted that some endpoints use OPTIONS requests
as a way of qualifying a peer and that a 401 authentication
response could result in interoperability issues, this option has
been disabled by default.
2010-09-03 18:19 +0000 [r284967] Brett Bryant <bbryant@digium.com>
* channels/chan_iax2.c, /: Merged revisions 284958 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
........ r284958 | bbryant | 2010-09-03 14:15:49 -0400 (Fri, 03
Sep 2010) | 8 lines This is a patch provided for issue #17935 to
add the ActionID to the IAXregistry AMI response. (closes issue
#17935) Reported by: alexkuklin Patches: iaxshowreg uploaded by
alexkuklin (license 1115) Tested by: alexkuklin ........
2010-09-03 18:03 +0000 [r284950-284952] David Vossel <dvossel@digium.com>
* channels/chan_sip.c: During OPTIONS authentication, the authpeer
does not need to be returned for any reason.
* configs/sip.conf.sample, CHANGES, channels/sip/include/sip.h,
channels/chan_sip.c: authenticate OPTIONS requests just like we
would an INVITE OPTIONS requests should be treated the same as an
INVITE This includes authentication. This patch adds the ability
for incoming out of dialog OPTION requests to be authenticated
before providing a response indicating whether an extension is
available or not. The authentication routine works the exact same
way as it does for incoming INVITEs. This means that if a peer
has 'insecure=invite' in their peer definition, the same will be
true for the processing of the OPTIONS request. Review:
https://reviewboard.asterisk.org/r/881/
2010-09-03 16:28 +0000 [r284921] Terry Wilson <twilson@digium.com>
* apps/app_chanspy.c, /: Merged revisions 284897 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r284897 | twilson | 2010-09-03 11:20:45 -0500
(Fri, 03 Sep 2010) | 12 lines Merged revisions 284881 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r284881 | twilson | 2010-09-03 11:10:23 -0500 (Fri, 03 Sep 2010)
| 5 lines Properly detect when a sound file doesn't exist
ast_fileexists returns -1 for error and 0 for a non-existant
file. The existing code treated missing files as though they
existed. ........ ................
2010-09-03 13:07 +0000 [r284849-284852] Jan Kalab <pitlicek@gmail.com>
* res/res_calendar_ews.c: Calendar categories and priorities:
strdupa() fix
* res/res_calendar_ews.c: Fix for calendar categories and
priorities according to ISO C90
* res/res_calendar_caldav.c, include/asterisk/calendar.h,
res/res_calendar_ews.c, res/res_calendar.c,
res/res_calendar_icalendar.c: Support for calendar events
priorities and categories Review 880
2010-09-02 21:04 +0000 [r284781] Brett Bryant <bbryant@digium.com>
* main/manager.c, /: Merged revisions 284778 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r284778 | bbryant | 2010-09-02 16:54:33 -0400
(Thu, 02 Sep 2010) | 14 lines Merged revisions 284777 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r284777 | bbryant | 2010-09-02 16:25:03 -0400 (Thu, 02 Sep 2010)
| 7 lines Fixes a bug in manager.c where the default
configuration values weren't reset when the manager configuration
was reloaded. (closes issue #17917) Reported by: lmadsen Review:
https://reviewboard.asterisk.org/r/883/ ........ ................
2010-09-02 21:02 +0000 [r284779-284780] Richard Mudgett <rmudgett@digium.com>
* channels/sig_pri.c: Simplified pri_dchannel() poll timeout
duration code.
* channels/sig_pri.c, channels/sig_pri.h, channels/chan_dahdi.c:
Made output libpri event names if pri debugging is enabled when
sig_pri processes them. * Simplified CLI "pri debug xx span xx"
command code and removed redundant debugging enabled messages. *
Made CLI "pri debug xx span xx" command only close the debugging
log file if it was opened.
2010-09-02 16:56 +0000 [r284705] David Vossel <dvossel@digium.com>
* /, channels/chan_sip.c: Merged revisions 284704 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r284704 | dvossel | 2010-09-02 11:48:51 -0500
(Thu, 02 Sep 2010) | 13 lines Merged revisions 284703 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r284703 | dvossel | 2010-09-02 11:47:15 -0500 (Thu, 02 Sep 2010)
| 7 lines Removed relatedpeer code from sip_autodestruct Handling
of the relatedpeer structure associated with a sip_pvt should be
done during the final sip_destruction function, not in
sip_autodestruct. ........ ................
2010-09-02 16:43 +0000 [r284701] Jason Parker <jparker@digium.com>
* formats/format_wav.c: Add slin16 support for format_wav (new
wav16 file extension) (closes issue #15029) Reported by: andrew
Patches: wav16.patch uploaded by andrew (license 240) Tested by:
qwell, andrew
2010-09-02 16:34 +0000 [r284698] Richard Mudgett <rmudgett@digium.com>
* doc/tex/channelvariables.tex, doc/tex/partymanip.tex (added),
doc/tex/asterisk.tex: Added documentation for CONNECTEDLINE and
REDIRECTING functions. (closes issue #17808) Reported by: jtodd
Review: https://reviewboard.asterisk.org/r/875/
2010-09-02 16:27 +0000 [r284597-284696] Tilghman Lesher <tlesher@digium.com>
* addons/ooh323c/src/oochannels.c: Fixing build
* channels/chan_usbradio.c, /: Merged revisions 284665 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
........ r284665 | tilghman | 2010-09-02 11:07:19 -0500 (Thu, 02
Sep 2010) | 2 lines Fixing build. ........
* apps/app_queue.c, /: Merged revisions 284631 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r284631 | tilghman | 2010-09-02 00:30:16 -0500 (Thu, 02 Sep 2010)
| 7 lines Don't reset queue stats on a module reload. (closes
issue #17535) Reported by: raarts Patches:
20100819__issue17535.diff.txt uploaded by tilghman (license 14)
........
* channels/chan_iax2.c, apps/app_queue.c, apps/app_getcpeid.c,
apps/app_followme.c, main/loader.c, apps/app_speech_utils.c,
pbx/pbx_loopback.c, channels/chan_dahdi.c, funcs/func_aes.c,
include/asterisk/module.h, pbx/pbx_realtime.c, pbx/pbx_dundi.c,
apps/app_stack.c, channels/chan_mgcp.c, apps/app_voicemail.c,
apps/app_adsiprog.c, channels/chan_sip.c, channels/chan_agent.c:
When optional_api is non-optional, force dependent modules to be
loaded. (closes issue #17707) Reported by: ira Patches:
20100819__issue17707__asterisk1.8.diff.txt uploaded by tilghman
(license 14) Tested by: tilghman Review:
https://reviewboard.asterisk.org/r/876/
* include/asterisk/channel.h, res/res_jabber.c, res/res_pktccops.c,
main/poll.c, channels/chan_usbradio.c, include/asterisk/select.h
(added), channels/chan_phone.c, channels/chan_misdn.c, configure,
main/features.c, include/asterisk/poll-compat.h,
tests/test_poll.c (added), addons/ooh323c/src/oochannels.c,
main/asterisk.c, addons/ooh323c/src/ooSocket.h, main/stun.c,
res/res_ais.c, /, include/asterisk/autoconfig.h.in, configure.ac,
channels/console_video.c: Merged revisions 284593,284595 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r284593 | tilghman | 2010-09-01 17:59:50 -0500
(Wed, 01 Sep 2010) | 18 lines Merged revisions 284478 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r284478 | tilghman | 2010-09-01 13:49:11 -0500 (Wed, 01 Sep 2010)
| 11 lines Ensure that all areas that previously used select(2)
now use poll(2), with implementations that need poll(2)
implemented with select(2) safe against 1024-bit overflows. This
is a followup to the fix for the pthread timer in 1.6.2 and
beyond, fixing a potential crash bug in all supported releases.
(closes issue #17678) Reported by: russell Branch:
https://origsvn.digium.com/svn/asterisk/team/tilghman/ast_select
Review: https://reviewboard.asterisk.org/r/824/ ........
................ r284595 | tilghman | 2010-09-01 22:57:43 -0500
(Wed, 01 Sep 2010) | 2 lines Failed to rerun bootstrap.sh after
last commit ................
2010-09-01 21:47 +0000 [r284561] David Vossel <dvossel@digium.com>
* channels/chan_sip.c: During request to dialog matching, verify
init_ruri is present before comparing. During request to dialog
matching, we attempt a best effort routine for fork detection
which requires several elements to be in place. The dialog's
initial request uri is one of those elements. Since it is best
effort, if the init_ruri is not present for some reason we can
not proceed with that routine.
2010-09-01 Leif Madsen <lmadsen@digium.com>
* Asterisk 1.8.0-beta5 released.
2010-09-01 18:44 +0000 [r284477] Terry Wilson <twilson@digium.com>
* res/res_srtp.c, res/res_rtp_asterisk.c,
include/asterisk/res_srtp.h, main/rtp_engine.c,
channels/chan_sip.c: Fix SRTP for changing SSRC and multiple
a=crypto SDP lines Adding code to Asterisk that changed the SSRC
during bridges and masquerades broke SRTP functionality. Also
broken was handling the situation where an incoming INVITE had
more than one crypto offer. This patch caches the SRTP policies
the we use so that we can change the ssrc and inform libsrtp of
the new streams. It also uses the first acceptable a=crypto line
from the incoming INVITE. (closes issue #17563) Reported by:
Alexcr Patches: srtp.diff uploaded by twilson (license 396)
Tested by: twilson Review:
https://reviewboard.asterisk.org/r/878/
2010-09-01 18:16 +0000 [r284415-284473] Tilghman Lesher <tlesher@digium.com>
* res/res_config_pgsql.c, /: Merged revisions 284472 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
........ r284472 | tilghman | 2010-09-01 13:13:35 -0500 (Wed, 01
Sep 2010) | 5 lines Don't warn on floats and timestamps (closes
issue #17082) Reported by: coolmig ........
* /, channels/chan_sip.c: Merged revisions 284399 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r284399 | tilghman | 2010-08-31 15:18:32 -0500
(Tue, 31 Aug 2010) | 14 lines Merged revisions 284393 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r284393 | tilghman | 2010-08-31 15:13:21 -0500 (Tue, 31 Aug 2010)
| 7 lines Don't send a devstate change on poke_noanswer if the
state did not change. (closes issue #17741) Reported by: schmidts
Patches: chan_sip.c.patch uploaded by schmidts (license 1077)
........ ................
2010-08-31 19:00 +0000 [r284318] Leif Madsen <lmadsen@digium.com>
* configs/say.conf.sample, /: Merged revisions 284317 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r284317 | lmadsen | 2010-08-31 13:59:31 -0500
(Tue, 31 Aug 2010) | 15 lines Merged revisions 284316 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r284316 | lmadsen | 2010-08-31 13:57:59 -0500 (Tue, 31 Aug 2010)
| 7 lines Update say.conf.sample to match the rules in say.c
(closes issue #17835) Reported by: RoadKill Patches:
say.conf.sample.patch.rules uploaded by RoadKill (license 933)
Tested by: RoadKill ........ ................
2010-08-30 22:28 +0000 [r284281] Tilghman Lesher <tlesher@digium.com>
* /, apps/app_festival.c: Merged revisions 284280 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r284280 | tilghman | 2010-08-30 17:27:06 -0500 (Mon, 30 Aug 2010)
| 11 lines Fix 3 coding errors: 1) After we close FD, we should
not be trying to write to it. 2) Call _exit(0), not exit(0), to
avoid running shutdown routines in a child. 3) Use endian, not
processor, detection to ensure bytes are written in the correct
order. (closes issue #15706) Reported by: modelnine Patches:
asterisk-1.6.1.1-festival-debug.patch uploaded by modelnine
(license 865) Tested by: gmartinez ........
2010-08-29 07:05 +0000 [r284096-284158] Tilghman Lesher <tlesher@digium.com>
* configs/res_curl.conf.sample (added): Missed adding this file
* sounds: Also ignore the checksums
* configs/cel_odbc.conf.sample (added), cel/cel_adaptive_odbc.c
(removed), cel/cel_odbc.c (added),
configs/cel_adaptive_odbc.conf.sample (removed): Rename CEL
adaptive driver to plain driver, since there isn't another ODBC
driver (and the other CEL drivers have adaptive capabilities,
anyway).
2010-08-28 21:29 +0000 [r284065] Russell Bryant <russell@digium.com>
* main/manager.c: Be more flexible with whitespace on AMI action
headers. Previously, this code required exactly one space to be
after the ':' in headers for an AMI action. This now makes
whitespace optional, and allows whitespace that is there to vary
in amount. (closes issue #17862) Reported by: cmoye Patches:
manager.c.patch_trunk uploaded by cmoye (license 858)
manager.c.patch_1.8 uploaded by cmoye (license 858) Tested by:
cmoye
2010-08-27 22:37 +0000 [r284032] David Vossel <dvossel@digium.com>
* /, channels/chan_sip.c: Merged revisions 284002 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r284002 | dvossel | 2010-08-27 17:27:50 -0500
(Fri, 27 Aug 2010) | 14 lines Merged revisions 283960 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r283960 | dvossel | 2010-08-27 17:17:26 -0500 (Fri, 27 Aug 2010)
| 8 lines Parse all "Accept" headers for SIP SUBSCRIBE requests.
(closes issue #17758) Reported by: ibc Patches:
multiple_accept_headers_1.4.diff uploaded by dvossel (license
671) ........ ................
2010-08-27 21:33 +0000 [r283951] Russell Bryant <russell@digium.com>
* pbx/pbx_realtime.c: Print exten@context:priority in verbose
messages from pbx_realtime.
2010-08-27 20:31 +0000 [r283882] Jason Parker <jparker@digium.com>
* main/config.c, addons/res_config_mysql.c, res/res_config_odbc.c,
/: Merged revisions 283881 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r283881 | qwell | 2010-08-27 15:30:27 -0500
(Fri, 27 Aug 2010) | 15 lines Merged revisions 283880 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r283880 | qwell | 2010-08-27 15:29:11 -0500 (Fri, 27 Aug 2010) |
8 lines Fix issue with decoding ^-escaped characters in realtime.
(closes issue #17790) Reported by: denzs Patches:
17790-chunky.diff uploaded by qwell (license 4) Tested by: qwell,
denzs ........ ................
2010-08-26 23:47 +0000 [r283770] Tilghman Lesher <tlesher@digium.com>
* res/res_musiconhold.c: Convert MOH to use generic timers. (closes
issue #17726) Reported by: lmadsen Patches:
20100825__issue17726__2.diff.txt uploaded by tilghman (license
14) Tested by: tilghman
2010-08-26 15:26 +0000 [r283692] David Vossel <dvossel@digium.com>
* /, channels/chan_sip.c: Merged revisions 283691 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r283691 | dvossel | 2010-08-26 10:24:40 -0500
(Thu, 26 Aug 2010) | 25 lines Merged revisions 283690 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r283690 | dvossel | 2010-08-26 10:22:28 -0500 (Thu, 26 Aug 2010)
| 19 lines Fixed how Asterisk destroys a dialog on channel hangup
before invite receives a response. If an ast_channel with a SIP
tech pvt hangs up before the sip dialog gets a response to its
outgoing INVITE, Asterisk used to pretend_ack the INVITE. This is
not rfc compliant and results in confusion at the other endpoint.
sip_pretend_ack will ack and remove all the packets in the
retransmit queue. This means that the INVITE will stop
retransmitting, and that any response to that INVITE that comes
after the pretend_ack occurs will be ignored. Instead of faking
any sort of acknowledgement for an outgoing INVITE during an
internal hangup, we should let the protocol stack process the
INVITE transaction and terminate the dialog properly. This is
achieved by setting the PENDING_BYE flag. When this flag is used,
once the dialog proceeds to an escapable state the transaction
will either be canceled with a SIP_CANCEL or completed followed
immediately by a BYE. Attempting to do this any other way is
incorrect. If the endpoint is not responding to the INVITE
request, the INVITE must continue to be retransmitted until it
times out which will result in the dialog being destroyed.
........ ................
2010-08-26 13:26 +0000 [r283627-283659] Russell Bryant <russell@digium.com>
* res/res_odbc.c: Slight improvement to a debug message.
* keys/iaxtel.pub (removed), keys/freeworlddialup.pub (removed),
Makefile: Remove public keys that are no longer useful.
* configs/manager.conf.sample: Move httptimeout out from in between
port and bindaddr.
2010-08-25 22:57 +0000 [r283595] David Vossel <dvossel@digium.com>
* /, channels/chan_sip.c: Merged revisions 283594 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r283594 | dvossel | 2010-08-25 17:56:42 -0500 (Wed, 25 Aug 2010)
| 7 lines Add to and from tags to NOTIFY dialog-info xml body so
pickup can occur. When pedantic mode is used, the dialog-info xml
generated during a ringing event must contain the to and from tag
values. Otherwise if a pickup occurs using INVITE with replaces,
Astrisk will not be able to locate the subscription. ........
2010-08-25 16:12 +0000 [r283561] Tilghman Lesher <tlesher@digium.com>
* res/res_odbc.c: Initialize connect timeout on each time through
the loop. (closes issue #17911) Reported by: wurstsalat
2010-08-25 15:54 +0000 [r283559] David Vossel <dvossel@digium.com>
* channels/sip/include/sip.h, /, channels/chan_sip.c: Merged
revisions 283558 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r283558 | dvossel | 2010-08-25 10:52:54 -0500 (Wed, 25 Aug 2010)
| 10 lines Asterisk will not advertise session timers are
supported when 'session-timers=refuse' is used. Asterisk now
dynamically builds the "Supported" header depending on what is
enabled/disabled in sip.conf. Session timers used to always be
advertised as being supported even when they were disabled in the
configuration. This caused problems with some end points. (issue
#17005) ........
2010-08-25 14:55 +0000 [r283527] Russell Bryant <russell@digium.com>
* channels/chan_sip.c: Convert ast_log(LOG_DEBUG, ...) to
ast_debug(...)
2010-08-24 20:34 +0000 [r283493] David Vossel <dvossel@digium.com>
* UPGRADE.txt, configs/sip.conf.sample, channels/sip/include/sip.h:
Changes the default behavior for sip.conf's pedantic option from
"no" to "yes".
2010-08-24 18:56 +0000 [r283457] Leif Madsen <lmadsen@digium.com>
* res/res_rtp_asterisk.c, channels/chan_sip.c: Fix issue where TOS
is no longer set on RTP packets. Fix issue where the tos is no
longer being set on RTP packets through res_rtp_asterisk. (closes
issue #17890) Reported by: elguero Patches: qos_18.diff uploaded
by elguero (license 37) Review:
https://reviewboard.asterisk.org/r/868
2010-08-24 16:11 +0000 [r283382] David Vossel <dvossel@digium.com>
* /, channels/chan_sip.c: Merged revisions 283381 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r283381 | dvossel | 2010-08-24 11:07:37 -0500
(Tue, 24 Aug 2010) | 18 lines Merged revisions 283380 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r283380 | dvossel | 2010-08-24 11:01:51 -0500 (Tue, 24 Aug 2010)
| 11 lines This fix makes sure the ast_channel hangs up correctly
when the dialog's PENDING_BYE flag is set. When the pending bye
flag is used, it is possible that the dialog will terminate and
leave the sip_pvt->owner channel up. This is because we never
hangup the ast_channel after sending the SIP_BYE request. When we
receive the response for the SIP_BYE we set need_destroy which we
would expect to destroy the dialog on the next do_monitor loop,
but this is not the case. The dialog will only be destroyed once
the owner is hungup even with the need_destroy flag set. This
patch sets the softhangup flag on the ast_channel when a SIP_BYE
request is sent as a result of the pending bye flag. ........
................
2010-08-24 12:49 +0000 [r283350] Russell Bryant <russell@digium.com>
* funcs/func_odbc.c: Don't attempt to release a NULL ODBC handle.
2010-08-23 21:33 +0000 [r283319] Tilghman Lesher <tlesher@digium.com>
* cdr/cdr_adaptive_odbc.c, cdr/cdr_odbc.c, cel/cel_adaptive_odbc.c,
/: Merged revisions 283318 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r283318 | tilghman | 2010-08-23 16:32:14 -0500 (Mon, 23 Aug 2010)
| 2 lines CDR drivers depend upon res_odbc, not directly on the
ODBC libraries ........
2010-08-23 Leif Madsen <lmadsen@digium.com>
* Asterisk 1.8.0-beta4 Released.
2010-08-23 13:35 +0000 [r283177-283241] Russell Bryant <russell@digium.com>
* configs/cel.conf.sample: Add sample configuration for cel_radius.
* main/cel.c, include/asterisk/cel.h: Make the AST_CEL_AMA enum
match up with the AST_CDR_ ama flag values. Really, having 2
enums for this is silly and error prone, demonstrated by the
crash that I hit because there was an assumption in the code that
the values in each matched up. However, this is a quick fix to
get them to match up so it will work.
* main/cel.c: Don't blow up on an invalid AMA flag.
* configs/cel_custom.conf.sample: Tack on ${eventextra} to the
sample cel_custom.conf.
* configs/cel_custom.conf.sample: Cut down on excessive quotation.
2010-08-23 12:06 +0000 [r283175] Tilghman Lesher <tlesher@digium.com>
* res/res_stun_monitor.c: Don't fail to start if the config file is
missing.
2010-08-23 11:58 +0000 [r283173] Russell Bryant <russell@digium.com>
* configs/cel_custom.conf.sample: Expand cel_custom.conf.sample.
Include the usage of CSV_QUOTE() to ensure data has valid CSV
formatting. Also list the special CEL variables that are
available for use in the mapping.
2010-08-20 16:51 +0000 [r283050-283125] Richard Mudgett <rmudgett@digium.com>
* /: Recorded merge of revisions 283124 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r283124 | rmudgett | 2010-08-20 11:48:10 -0500
(Fri, 20 Aug 2010) | 16 lines Merged revisions 283123 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
................ r283123 | rmudgett | 2010-08-20 11:46:22 -0500
(Fri, 20 Aug 2010) | 9 lines Merged revision 278274 from
https://origsvn.digium.com/svn/asterisk/trunk .......... r278274
| rmudgett | 2010-07-20 17:38:13 -0500 (Tue, 20 Jul 2010) | 1
line Reference correct struct member for unlikely event
PRI_EVENT_CONFIG_ERR. .......... ................
................
* channels/sig_pri.c, /: Merged revisions 283049 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r283049 | rmudgett | 2010-08-20 10:31:03 -0500
(Fri, 20 Aug 2010) | 29 lines Merged revisions 283048 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r283048 | rmudgett | 2010-08-20 10:24:36 -0500 (Fri, 20 Aug 2010)
| 22 lines Q931 - Sending PROGRESS after sending ALERTING is a
protocol error The PRI layer in chan_dadhi will check if a
PROGRESS message has already been sent, and not allow sending
another (although that is technically allowed by the Q931 spec),
however it does not protect against sending an ALERTING and then
sending a PROGRESS message, which is a violation of the
specification. Most switches don't seem to care too deeply about
this, but some do, and will disconnect the call when receiving
this invalid sequence. Protocol specification reference:
T-REC-Q.931-199805-I page 223, "Figure A.5/Q.931 -- Overview
protocol control (network side) point-point (sheet 3 of 8)"
(closes issue #17874) Reported by: nic_bellamy Patches:
asterisk-1.4-r282537_no-progress-after-alerting.patch uploaded by
nic bellamy (license 299)
asterisk-1.6.2-r282537_no-progress-after-alerting.patch uploaded
by nic bellamy (license 299)
asterisk-trunk-r282537_no-progress-after-alerting.patch uploaded
by nic bellamy (license 299) ........ ................
2010-08-20 12:45 +0000 [r282979-283013] Russell Bryant <russell@digium.com>
* configs/cel_adaptive_odbc.conf.sample: Fix a typo in a column
name.
* apps/app_celgenuserevent.c: Add an argument missing from the
CELGenUserEvent documentation.
2010-08-19 21:07 +0000 [r282891-282895] David Vossel <dvossel@digium.com>
* /, channels/chan_sip.c: Merged revisions 282894 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r282894 | dvossel | 2010-08-19 16:05:54 -0500
(Thu, 19 Aug 2010) | 18 lines Merged revisions 282893 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r282893 | dvossel | 2010-08-19 16:03:24 -0500 (Thu, 19 Aug 2010)
| 11 lines tos_sip option was not being set correctly When
tos_sip is used, the tos of the sip socket is only set correctly
if the socket binding changes on a reload. If the binding stays
the same but the TOS changes, the new tos value would not take
into effect. This patch fixes that. (closes issue #17712)
Reported by: nickb ........ ................
* /, channels/chan_sip.c: Merged revisions 282890 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r282890 | dvossel | 2010-08-19 15:31:22 -0500 (Thu, 19 Aug 2010)
| 5 lines fixes sip peer memory leaks in the peer_by_ip table
(issue #17798) ........
2010-08-19 20:01 +0000 [r282860] Matthew Nicholson <mnicholson@digium.com>
* /, channels/chan_sip.c: Merged revisions 282859 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r282859 | mnicholson | 2010-08-19 14:44:00 -0500
(Thu, 19 Aug 2010) | 23 lines Merged revisions 277944 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r277944 | pabelanger | 2010-07-19 15:56:07 -0500 (Mon, 19 Jul
2010) | 16 lines Regression with T.38 negotiation Prior to
1.4.26.3 T.38 negotiation worked properly, in the case of the
reporter. (issue #16852) Reported by: cfc (closes issue #16705)
Reported by: mpiazzatnetbug Patches: issue16705_2.diff uploaded
by ebroad (license 878) Tested by: vrban, ebroad, c0rnoTa,
samdell3 Review: https://reviewboard.asterisk.org/r/754/ ........
................
2010-08-19 14:44 +0000 [r282826] Tilghman Lesher <tlesher@digium.com>
* main/netsock2.c: Only output debugging if the debug level is on.
2010-08-19 02:18 +0000 [r282740] Terry Wilson <twilson@digium.com>
* configs/sip.conf.sample, /: Merged revisions 282730 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r282730 | twilson | 2010-08-18 21:14:28 -0500
(Wed, 18 Aug 2010) | 9 lines Merged revisions 282729 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r282729 | twilson | 2010-08-18 21:12:55 -0500 (Wed, 18
Aug 2010) | 2 lines Add some documentation about codec
negotiation to sip.conf ........ ................
2010-08-18 15:28 +0000 [r282671-282672] Richard Mudgett <rmudgett@digium.com>
* channels/sig_pri.h: Use the correct type for aoce_delayhangup bit
field.
* channels/chan_dahdi.c: Use the correct operator when calculating
the PRI span devstate.
2010-08-18 13:10 +0000 [r282639] Matthew Nicholson <mnicholson@digium.com>
* channels/chan_sip.c: Properly handle 200 and unknown responses
conatined in NOTIFY requests received in response to REFER
requests. This patch fixes the way asterisk handles NOTIFY
requests received in response to REFER requests. These changes to
NOTIFY handler were first introduced in r217482. This new change
properly handles the 200 response by queueing an
AST_TRANSFER_SUCCESS control frame and also prevents that control
frame from being queued when provisional and unknown responses
are received. (issue #17486) Reported by: davidw Tested by:
mnicholson (issue #12713) Reported by: davidw Review:
https://reviewboard.asterisk.org/r/860/
2010-08-18 12:30 +0000 [r282638] Russell Bryant <russell@digium.com>
* channels/chan_multicast_rtp.c: Split _all_ arguments before
parsing them. This fixes multicast RTP paging using linksys mode.
2010-08-18 07:49 +0000 [r282608] Tilghman Lesher <tlesher@digium.com>
* channels/sig_pri.c, /: Merged revisions 282607 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r282607 | tilghman | 2010-08-18 02:43:14 -0500 (Wed, 18 Aug 2010)
| 9 lines Don't warn on callerid when completely text, instead of
numeric with localdialplan prefixes. (closes issue #16770)
Reported by: jamicque Patches: 20100413__issue16770.diff.txt
uploaded by tilghman (license 14) 20100811__issue16770.diff.txt
uploaded by tilghman (license 14) Tested by: jamicque ........
2010-08-17 21:36 +0000 [r282543-282577] David Vossel <dvossel@digium.com>
* /, channels/chan_sip.c: Merged revisions 282576 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r282576 | dvossel | 2010-08-17 16:35:17 -0500 (Tue, 17 Aug 2010)
| 9 lines fixes no default transport for temp peer creation in
chan_sip (closes issue #17829) Reported by: falves11 Patches:
issue_17829.rev1.txt uploaded by russell (license 2)
issue_17829.diff uploaded by dvossel (license 671) Tested by:
falves11 ........
* channels/chan_iax2.c: ACCEPT message should respond with the new
FORMAT2 ie (closes issue #17804) Reported by: tpanton
* include/asterisk/unaligned.h: fixes truncated uint64_t value in
put_unaligned_uint64_t() function (issue #17804)
2010-08-16 18:01 +0000 [r282470] Leif Madsen <lmadsen@digium.com>
* doc/tex/asterisk.tex, doc/tex/sounds.tex (added), /: Merged
revisions 282469 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r282469 | lmadsen | 2010-08-16 13:00:09 -0500 (Mon, 16 Aug 2010)
| 7 lines Add information about creating sounds files using the
sounds tools publically available so that others can create their
own sounds prompts using the same tools we use to generate sounds
releases. This allows people creating their own prompts to sound
consistent with the prompts available from the open source
project. SWP-595 ........
2010-08-16 17:53 +0000 [r282468] Terry Wilson <twilson@digium.com>
* main/channel.c, /: Merged revisions 282467 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r282467 | twilson | 2010-08-16 12:32:01 -0500
(Mon, 16 Aug 2010) | 23 lines Merged revisions 282430 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r282430 | twilson | 2010-08-16 12:06:37 -0500 (Mon, 16 Aug 2010)
| 16 lines Send a SRCCHANGE indication when we masquerade
Masquerading a channel means that the src of the audio is
potentially changing, so send a SRCCHANGE so that RTP-based media
streams can get a new SSRC generated to reflect the change.
Original patch by addix (along with lots of testing--thanks!).
(closes issue #17007) Reported by: addix Patches:
1001-reset-SSRC-original-channel.diff uploaded by addix (license
1006) srcchange.diff uploaded by twilson (license 396) Tested by:
addix, twilson Review: https://reviewboard.asterisk.org/r/862/
........ ................
2010-08-14 04:53 +0000 [r282366] Tilghman Lesher <tlesher@digium.com>
* channels/chan_iax2.c, include/asterisk/sched.h: Fix our FRACKing
issue with chan_iax2 a different way. Review:
https://reviewboard.asterisk.org/r/861/
2010-08-13 23:53 +0000 [r282334] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c: PRI CCSS may use a stale dial string for
the recall dial string. If an outgoing call negotiates a
different B channel than initially requested, the saved original
dial string was not transferred to the new B channel. CCSS uses
that dial string to generate the recall dial string.
2010-08-13 22:23 +0000 [r282236-282302] David Vossel <dvossel@digium.com>
* UPGRADE.txt, configs/sip.conf.sample, CHANGES,
channels/chan_sip.c: remove current STUN support from chan_sip.c
This patch removes the current broken/useless stun support from
chan_sip. (closes issue #17622) Reported by: philipp2 Review:
https://reviewboard.asterisk.org/r/855/
* CHANGES: res_stun_monitor and corresponding options CHANGES
documentation
* configs/res_stun_monitor.conf.sample (added),
configs/sip.conf.sample, channels/chan_iax2.c,
configs/iax.conf.sample, channels/chan_sip.c,
include/asterisk/event_defs.h, res/res_stun_monitor.c (added):
res_stun_monitor for monitoring network changes behind a NAT
device Review: https://reviewboard.asterisk.org/r/854
* /, channels/chan_sip.c: Merged revisions 282235 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r282235 | dvossel | 2010-08-13 13:54:53 -0500 (Fri, 13 Aug 2010)
| 16 lines only do magic pickup when notifycid is enabled A new
way of doing BLF pickup was introduced into 1.6.2. This feature
adds a call-id value into the XML of a SIP_NOTIFY message sent to
alert a subscriber that a device is ringing. This option should
only be enabled when the new 'notifycid' option is set... but
this was not the case. Instead the call-id value was included for
every RINGING Notify message, which caused a regression for
people who used other methods for call pickup. (closes issue
#17633) Reported by: urosh Patches: chan_sip.txt uploaded by
urosh (license ) blf_cid_issue.diff uploaded by dvossel (license
671) Tested by: dvossel, urosh, okrief, alecdavis ........
2010-08-13 16:02 +0000 [r282200-282201] Terry Wilson <twilson@digium.com>
* configure.ac: Whitespace fix :-/
* configure, configure.ac: Detect when libsrtp cannot be linked in
a shared library The libsrtp build system currently does not
produce a shared library or a static library compiled with -fPIC,
so on 64-bit systems it is possible that we will get a compile
error if libsrtp is installed and res_srtp is selected in
menuselect. This patch attempts to detect this situation and
provide the user with instructions to work around the problem.
2010-08-12 22:51 +0000 [r282131] Jason Parker <jparker@digium.com>
* pbx/pbx_config.c, /: Merged revisions 282130 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r282130 | qwell | 2010-08-12 17:50:54 -0500
(Thu, 12 Aug 2010) | 9 lines Merged revisions 282129 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r282129 | qwell | 2010-08-12 17:49:28 -0500 (Thu, 12 Aug
2010) | 1 line Register CLI commands before parsing config, in
case there is a config error. ........ ................
2010-08-12 22:06 +0000 [r282098] Richard Mudgett <rmudgett@digium.com>
* include/asterisk/ccss.h, main/ccss.c: Separate call completion
config parameter allocation and default initialization. If you
ever have a need to reset the call completion config parameters
to defaults, now you can. And no Virginia, C++ idioms do not
always work in C.
2010-08-12 20:41 +0000 [r282066] Russell Bryant <russell@digium.com>
* CHANGES, main/cli.c: Add a "core reload" CLI command. Review:
https://reviewboard.asterisk.org/r/859/
2010-08-12 20:15 +0000 [r282047] David Vossel <dvossel@digium.com>
* CHANGES, include/asterisk/translate.h, main/cli.c,
main/translate.c: improved translation paths for wideband codecs
The problem I'm addressing is that Asterisk's current method of
building the least cost translation paths between codecs does not
take into account sample rate. For instance, it was possible for
siren14 (a 32khz codec), to contain the a translation path to
siren7 (a 16khz audio codec) that goes through slin at 8khz. In
this case Asterisk takes a 32khz codec, down samples it to 8khz
and then up samples it to 16khz which is terrible regardless if
it is computationally less expensive. This patch now builds
translation paths that give priority to maintaining the best
possible sample rate before taking into consideration
computational cost. This patch also adds cli commands to expose
what translation paths are actually being used. Changes: 1.
Translation paths will never contain a step that changes the
sample rate unless absolutely necessary. 2. When choosing the
best codec to make two channels compatible. Shared codecs with
the highest sample rate are given priority. 3. A new cli command
to show all translation paths available for a specific codec
'core show translation paths [codec name]' has been added. 4.
'core show translation' which displays the translation matrix now
includes the new higher bit audio codecs in the table. 5. 'core
show channel [channel name]' now displays the translation paths
if translation is used. (closes issue #16841) Reported by:
dvossel Review: https://reviewboard.asterisk.org/r/842/
2010-08-12 18:03 +0000 [r281982-282015] Russell Bryant <russell@digium.com>
* main/pbx.c: Put back pointer value output for ast_debug(), such
that it is only removed for verbose output.
* main/pbx.c: Remove debugging output from verbose messages.
Pointer values to internal objects is not terribly useful to
users in the verbose messages about adding extensions and
contexts.
2010-08-12 03:03 +0000 [r281913] Jeff Peeler <jpeeler@digium.com>
* main/channel.c, /: Merged revisions 281912 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r281912 | jpeeler | 2010-08-11 22:01:38 -0500
(Wed, 11 Aug 2010) | 27 lines Merged revisions 281911 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r281911 | jpeeler | 2010-08-11 22:00:14 -0500 (Wed, 11 Aug 2010)
| 20 lines Ensure SSRC is changed when media source is changed to
resolve audio delay. This change causes the SSRC to change right
before the channels are bridged, which is what used to happen. It
seems that fixes were made to attempt limiting SSRC changes,
targeted mainly at sending DTMF. DTMF is not affecting the SSRC
with this change. There are two other control frames sent in
ast_channel_bridge that probably should also be changed to
AST_CONTROL_SRCCHANGE as well, but I'm going to leave this change
up to the discretion of resolving issue #17007. For reference -
old review implementing new control frame SRCCHANGE:
https://reviewboard.asterisk.org/r/540 (closes issue #17404)
Reported by: sdolloff Patches: bug17404.patch uploaded by jpeeler
(license 325) Tested by: sdolloff ........ ................
2010-08-11 21:12 +0000 [r281875] Leif Madsen <lmadsen@digium.com>
* configs/say.conf.sample, /: Merged revisions 281873 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r281873 | lmadsen | 2010-08-11 16:09:47 -0500
(Wed, 11 Aug 2010) | 14 lines Merged revisions 281819 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r281819 | lmadsen | 2010-08-11 13:28:10 -0500 (Wed, 11 Aug 2010)
| 6 lines Add Danish support to say.conf.sample (closes issue
#17836) Reported by: RoadKill Patches: say.conf.sample.patch.dk
uploaded by RoadKill (license 933) ........ ................
2010-08-11 21:11 +0000 [r281874] Matthew Nicholson <mnicholson@digium.com>
* channels/chan_sip.c: handle all possible responses to REFER
requests (closes issue #17486) Reported by: davidw Patches:
Issue17486-counterbid.diff.txt uploaded by davidw (license 780)
Tested by: davidw Review: https://reviewboard.asterisk.org/r/837/
2010-08-11 20:30 +0000 [r281870] Richard Mudgett <rmudgett@digium.com>
* channels/sig_analog.c, channels/sig_analog.h: Fix a call to
analog_set_pulsedial() not setting 0 or 1 only. * Also a couple
minor tweaks.
2010-08-11 17:54 +0000 [r281764] Leif Madsen <lmadsen@digium.com>
* configs/say.conf.sample, /: Merged revisions 281763 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r281763 | lmadsen | 2010-08-11 12:54:09 -0500
(Wed, 11 Aug 2010) | 14 lines Merged revisions 281762 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r281762 | lmadsen | 2010-08-11 12:51:40 -0500 (Wed, 11 Aug 2010)
| 6 lines Allow say.conf to handle large numbers ending with
multiple zeros. (closes issue #17833) Reported by: RoadKill
Patches: say.conf.sample.patch.largenumbers uploaded by RoadKill
(license 933) ........ ................
2010-08-11 17:27 +0000 [r281760] Matthew Nicholson <mnicholson@digium.com>
* channels/chan_sip.c: Avoid a deadlock in
add_header_max_forwards(). Related to r276951
2010-08-11 15:18 +0000 [r281723] Tilghman Lesher <tlesher@digium.com>
* /, apps/app_readexten.c: Merged revisions 281722 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
........ r281722 | tilghman | 2010-08-11 10:17:20 -0500 (Wed, 11
Aug 2010) | 7 lines Only set status TIMEOUT, if we have no
digits. (closes issue #15188) Reported by: jcovert Patches:
app_readexten.c.patch-1.6.2.8-rc1 uploaded by jcovert (license
551) ........
2010-08-11 13:30 +0000 [r281687] <simon.perreault@viagenie.ca>
* include/asterisk/netsock2.h, configs/sip.conf.sample,
channels/sip/config_parser.c, main/netsock2.c: Fix parsing of
IPv6 address literals in outboundproxy (closes issue #17757)
Reported by: oej Patches: 17757.diff uploaded by sperreault
(license 252) sip.conf.diff uploaded by sperreault (license 252)
Tested by: oej
2010-08-10 21:47 +0000 [r281568-281650] Russell Bryant <russell@digium.com>
* UPGRADE.txt, configs/sip.conf.sample, channels/sip/include/sip.h:
Change the default value for alwaysauthreject in sip.conf to
"yes". (closes issue #17756) Reported by: oej
* main/sched.c, /: Merged revisions 281574 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r281574 | russell | 2010-08-10 13:04:32 -0500 (Tue, 10 Aug 2010)
| 9 lines Don't move the time threshold for running scheduled
events on every iteration. Instead, only calculate the time
threshold each time ast_sched_runq() is called. (closes issue
#17742) Reported by: schmidts Patches: sched.c.patch uploaded by
schmidts (license 1077) ........
* apps/app_dial.c, /: Merged revisions 281567 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r281567 | russell | 2010-08-10 12:47:13 -0500
(Tue, 10 Aug 2010) | 15 lines Merged revisions 281566 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r281566 | russell | 2010-08-10 12:45:45 -0500 (Tue, 10 Aug 2010)
| 8 lines Reset visible indication after answer. (closes issue
#17641) Reported by: klaus3000 Patches:
ast1.6.2.9-app_dial-visible_indication.patch.txt uploaded by
klaus3000 (license 65) Tested by: schmidts ........
................
2010-08-10 Leif Madsen <lmadsen@digium.com>
* Asterisk 1.8.0-beta3 Released.
2010-08-10 17:48 +0000 [r281529-281568] Russell Bryant <russell@digium.com>
* apps/app_dial.c, /: Merged revisions 281567 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r281567 | russell | 2010-08-10 12:47:13 -0500
(Tue, 10 Aug 2010) | 15 lines Merged revisions 281566 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r281566 | russell | 2010-08-10 12:45:45 -0500 (Tue, 10 Aug 2010)
| 8 lines Reset visible indication after answer. (closes issue
#17641) Reported by: klaus3000 Patches:
ast1.6.2.9-app_dial-visible_indication.patch.txt uploaded by
klaus3000 (license 65) Tested by: schmidts ........
................
* channels/chan_sip.c: Ensure that the proper external address is
used for the RTP destination. (closes issue #17044) Reported by:
ebroad Tested by: ebroad Review:
https://reviewboard.asterisk.org/r/566/
* main/cli.c: Resolve a problem with channel name tab completion.
Hitting tab without typing any part of a channel name resulted in
no results. This now results in getting a full list of active
channels, just as it did in previous versions of Asterisk.
Review: https://reviewboard.asterisk.org/r/818/
2010-08-10 07:26 +0000 [r281497] TransNexus OSP Development <support@transnexus.com>
* apps/app_osplookup.c: Fixed the issue caused by EXTEN including
user parameters.
2010-08-09 23:04 +0000 [r281466] Jeff Peeler <jpeeler@digium.com>
* channels/chan_local.c: Add some more stuff to copy from 281429.
2010-08-09 20:47 +0000 [r281432] David Vossel <dvossel@digium.com>
* /, channels/chan_sip.c: Merged revisions 281430 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r281430 | dvossel | 2010-08-09 15:46:50 -0500 (Mon, 09 Aug 2010)
| 13 lines fixes SIP peers memory leak We zeroed out the peer's
addr before it was removed from the peers_by_ip container. This
made it impossible to be removed from the container as the addr
is the key used by the container to find the peer. (closes issue
#17774) Reported by: kkm Patches:
017774-sip-peer-leak-1.6.2.10.diff uploaded by kkm (license 888)
017774-sip-peer-leak-1.8.diff uploaded by kkm (license 888)
........
2010-08-09 20:43 +0000 [r281429] Jeff Peeler <jpeeler@digium.com>
* channels/chan_local.c, /: Merged revisions 281391 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r281391 | jpeeler | 2010-08-09 15:07:29 -0500
(Mon, 09 Aug 2010) | 20 lines Merged revisions 281390 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r281390 | jpeeler | 2010-08-09 15:04:30 -0500 (Mon, 09 Aug 2010)
| 13 lines Prevent loss of Caller ID information set on local
channel after masquerade. Caller ID set on the channel before a
masquerade occurs when using a local channel would cause the
information to be lost. The problem was that the information was
set on a channel destined to be hung up. The somewhat confusing
fix is to detect if any Caller ID has been set on the channel and
if so preswap the Caller ID data so that basically the masquerade
puts the data back. (closes issue #17138) Reported by: kobaz
Review: https://reviewboard.asterisk.org/r/847/ ........
................
2010-08-09 14:49 +0000 [r281358] Matthew Nicholson <mnicholson@digium.com>
* res/res_fax.c: Validate minrate, maxrate, and modem settings
before attempting a fax session. FAX-224
2010-08-09 14:31 +0000 [r281356] <simon.perreault@viagenie.ca>
* configs/sip.conf.sample: Added comment about IPv4-mapped IPv6
addresses and the output of netstat.
2010-08-09 12:51 +0000 [r281294-281325] Russell Bryant <russell@digium.com>
* configs/cdr.conf.sample: Add a couple of default values to the
documentation of cdr.conf.
* configs/cdr.conf.sample: Reorder some options in cdr.conf.sample.
Put all of the options that affect the contents of CDRs together,
instead of having the batch mode options in the middle of them.
2010-08-06 18:57 +0000 [r281085] Tilghman Lesher <tlesher@digium.com>
* main/utils.c: Fix alignment of stringfields on the SPARC
architecture (closes issue #17789) Reported by: Ian Mason
Patches: 20100806__issue17789__2.diff.txt uploaded by tilghman
(license 14) Tested by: Ian_Mason
2010-08-05 13:16 +0000 [r281052] Russell Bryant <russell@digium.com>
* main/cdr.c, /: Merged revisions 281051 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r281051 | russell | 2010-08-05 08:11:32 -0500 (Thu, 05 Aug 2010)
| 9 lines Cleanup default option value handling for cdr.conf
[general]. The default values would differ depending on whether
or not cdr.conf exists. That is no longer the case. Apply a
default value to the unanswered option. Define all default values
as named constants. ........
2010-08-05 07:46 +0000 [r280984] Tilghman Lesher <tlesher@digium.com>
* include/asterisk/pbx.h, main/pbx.c, /: Merged revisions 280983
via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r280983 | tilghman | 2010-08-05 02:40:47 -0500
(Thu, 05 Aug 2010) | 15 lines Merged revisions 280982 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r280982 | tilghman | 2010-08-05 02:28:33 -0500 (Thu, 05 Aug 2010)
| 8 lines Change context lock back to a mutex, because
functionality depends upon the lock being recursive. (closes
issue #17643) Reported by: zerohalo Patches:
20100726__issue17643.diff.txt uploaded by tilghman (license 14)
Tested by: zerohalo ........ ................
2010-08-04 15:11 +0000 [r280909] Matthew Nicholson <mnicholson@digium.com>
* res/res_fax.c: Initialize FAXOPT() status variables in sendfax
and receivefax instead of when the details structure is created.
2010-08-04 14:04 +0000 [r280809-280879] Tilghman Lesher <tlesher@digium.com>
* channels/chan_mgcp.c: Check cur value before attempting a deref.
(closes issue #17775) Reported by: svinson Patches:
20100804__issue17775.diff.txt uploaded by tilghman (license 14)
Tested by: svinson (closes issue #17743) Reported by: tgruenberg
Patches: 20100804__issue17775.diff.txt uploaded by tilghman
(license 14) Tested by: tgruenberg
* CHANGES, funcs/func_strings.c: Sneak FIELDNUM() into 1.8. Returns
a 1-based index into a list of a specified item. Matches up with
FIELDQTY() and CUT(). (closes issue #17713) Reported by: gareth
Patches: svn-279754.diff uploaded by gareth (license 208) Tested
by: gareth, tilghman Review:
https://reviewboard.asterisk.org/r/810/
2010-08-03 19:54 +0000 [r280777-280778] <simon.perreault@viagenie.ca>
* channels/chan_sip.c: Fixed IPv6-related SIP parsing bugs. (closes
issue #17663) Reported by: oej Patches: diff uploaded by
sperreault (license 252) diff2 uploaded by sperreault (license
252) get_domain.diff uploaded by sperreault (license 252)
* configs/sip.conf.sample: Better documentation related to IPv6.
(closes issue #17737) Reported by: oej Patches: doc.diff uploaded
by sperreault (license 252) Tested by: mmichelson
2010-08-03 18:48 +0000 [r280742] Russell Bryant <russell@digium.com>
* addons/Makefile, addons/mp3 (removed),
contrib/scripts/get_mp3_source.sh (added): Remove the MP3 decoder
source code and replace it with a small shell script. Review:
https://reviewboard.asterisk.org/r/836/
2010-08-03 18:42 +0000 [r280624-280740] Tilghman Lesher <tlesher@digium.com>
* doc/asterisk.sgml, /, doc/asterisk.8, doc/Makefile (added):
Merged revisions 280739 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r280739 | tilghman | 2010-08-03 13:39:28 -0500 (Tue, 03 Aug 2010)
| 2 lines Document -B and -W flags and regenerate manpage from
sgml ........
* apps/app_voicemail.c, /: Merged revisions 280671 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
........ r280671 | tilghman | 2010-08-02 16:26:11 -0500 (Mon, 02
Aug 2010) | 2 lines Allow the pipe, but also allow the comma
........
* main/Makefile: Make this a little more deterministic... we want
the latest value, not just a 1 somewhere.
* main/Makefile: Apparently, the values in makeopts are sometimes
1:1 and sometimes 1. Compensate for this.
2010-07-29 21:07 +0000 [r280557] Matthew Nicholson <mnicholson@digium.com>
* res/res_fax.c: Fix regression introduced in r1664. Give the fax
stack time to shutdown and populate the FAXOPT output variables.
FAX-222
2010-07-29 20:43 +0000 [r280552] David Vossel <dvossel@digium.com>
* /, channels/chan_sip.c: Merged revisions 280551 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r280551 | dvossel | 2010-07-29 15:42:29 -0500 (Thu, 29 Jul 2010)
| 11 lines fixes wrong SRV query for TLS connection (closes issue
#17612) Reported by: marcelloceschia Patches:
chan-sip_srvQuery.patch uploaded by marcelloceschia (license
1079) chan-sip_Trunk_srvQuery.patch uploaded by st (license 907)
chan-sip_asterisk18b1_srvQuery.patch uploaded by marcelloceschia
(license 1079) Tested by: marcelloceschia, st, pabelanger
........
2010-07-29 20:35 +0000 [r280549] Russell Bryant <russell@digium.com>
* configs/ccss.conf.sample: Add header to ccss.conf to appease oej.
(closes issue #17755) Reported by: oej
2010-07-29 19:47 +0000 [r280519] Sean Bright <sean@malleable.com>
* channels/sig_pri.c: Fix compilation error in chan_dahdi (strdupa
-> ast_strdupa). (closes issue #17751) Reported by: b11d Patches:
strdupa_oops.diff uploaded by malcolmd (license 924)
2010-07-29 19:13 +0000 [r280450] David Vossel <dvossel@digium.com>
* main/channel.c, /: Merged revisions 280449 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r280449 | dvossel | 2010-07-29 14:05:25 -0500
(Thu, 29 Jul 2010) | 18 lines Merged revisions 280448 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r280448 | dvossel | 2010-07-29 14:04:23 -0500 (Thu, 29 Jul 2010)
| 12 lines fixes issue with translator frame not getting freed A
translator frame even if it local storage so the translation path
can be freed. This issue prevented g729 licenses from being freed
up. (closes issue #17630) Reported by: manvirr Patches:
encoder_fix.diff uploaded by dvossel (license 671) Tested by:
manvirr, dvossel ........ ................
2010-07-29 18:37 +0000 [r280414-280446] Paul Belanger <paul.belanger@polybeacon.com>
* tests/test_utils.c: Remove res_crypto dependency.
* tests/test_utils.c: crypto_loaded_test depends on res_crypto,
else test will fail.
2010-07-29 16:25 +0000 [r280391] Russell Bryant <russell@digium.com>
* main/rtp_engine.c: Don't blow up if get_codec() was not provided
in the RTP glue.
2010-07-29 16:07 +0000 [r280346] Jean Galarneau <jgalarneau@digium.com>
* /, apps/app_meetme.c: Merged revisions 280345 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r280345 | jeang | 2010-07-29 11:01:35 -0500
(Thu, 29 Jul 2010) | 10 lines Merged revisions 280341 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r280341 | jeang | 2010-07-29 10:52:31 -0500 (Thu, 29 Jul 2010) |
2 lines Fix a dsp structure leak occuring when a local channel is
put into a meetme conference, then masquaraded away. ABE-2422
........ ................
2010-07-29 15:57 +0000 [r280307-280343] Matthew Nicholson <mnicholson@digium.com>
* channels/chan_usbradio.c: Use PRIx64 instead of PRId64 in format
string. related to r280302
* main/channel.c, channels/chan_local.c, /: Merged revisions 280306
via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r280306 | mnicholson | 2010-07-29 08:45:11 -0500 (Thu, 29 Jul
2010) | 2 lines Implement support for ast_channel_queryoption on
local channels. Currently only AST_OPTION_T38_STATE is supported.
ABE-2229 Review: https://reviewboard.asterisk.org/r/813/ ........
Additionally, pass AST_CONTROL_T38_PARAMETERS control frames
through generic bridges. This change appears to have been
unintentionally left out of rev 203699.
2010-07-29 00:45 +0000 [r280302] Paul Belanger <paul.belanger@polybeacon.com>
* channels/chan_usbradio.c: Use PRId64 with format_t
2010-07-28 20:49 +0000 [r280269] Jeff Peeler <jpeeler@digium.com>
* channels/sip/reqresp_parser.c: Give test category missing leading
slash
2010-07-28 20:12 +0000 [r280235] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c, /: Merged revisions 280229 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
........ r280229 | rmudgett | 2010-07-28 14:57:49 -0500 (Wed, 28
Jul 2010) | 2 lines Add missing enum value "unknown" to the SS7
called_nai and calling_nai config options. ........
2010-07-28 20:03 +0000 [r280233] Jason Parker <jparker@digium.com>
* sounds/Makefile, /: Merged revisions 280231 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r280231 | qwell | 2010-07-28 15:02:27 -0500 (Wed, 28 Jul 2010) |
6 lines Work around some silly behavior on BSD. A non-zero exit
from a subshell should make the build fail. (closes issue #17621)
........
2010-07-28 19:34 +0000 [r280225] Terry Wilson <twilson@digium.com>
* res/res_rtp_asterisk.c: Do rtp/rtcp debugging when it is turned
on w/o filtering
2010-07-28 18:24 +0000 [r280195] Jason Parker <jparker@digium.com>
* sounds/Makefile, /: Merged revisions 280193 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r280193 | qwell | 2010-07-28 13:05:54 -0500 (Wed, 28 Jul 2010) |
9 lines Remove unnecessary subshells. Attempt to make
checksumming work. Also improves readability. (issue #17621)
Reported by: bjm Review: https://reviewboard.asterisk.org/r/808/
........
2010-07-28 16:52 +0000 [r280161] Sean Bright <sean@malleable.com>
* apps/app_queue.c, /: Merged revisions 280160 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r280160 | seanbright | 2010-07-28 12:51:11 -0400 (Wed, 28 Jul
2010) | 8 lines Plug a reference leak in app_queue when adding
members dynamically. (closes issue #17738) Reported by:
bobwienholt Patches: issue17738.patch uploaded by bobwienholt
(license 950) Tested by: bobwienholt, seanbright ........
2010-07-28 13:52 +0000 [r280090] Leif Madsen <lmadsen@digium.com>
* contrib/scripts/live_ast, /: Merged revisions 280089 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r280089 | lmadsen | 2010-07-28 08:51:16 -0500
(Wed, 28 Jul 2010) | 9 lines Merged revisions 280088 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r280088 | lmadsen | 2010-07-28 08:50:38 -0500 (Wed, 28
Jul 2010) | 1 line Update help text to be less confusing.
........ ................
2010-07-28 13:01 +0000 [r280058] Russell Bryant <russell@digium.com>
* res/res_crypto.c: s/init keys/keys init/
2010-07-28 01:37 +0000 [r280023] Paul Belanger <paul.belanger@polybeacon.com>
* channels/chan_usbradio.c: Resolve compiler warning about
formatting (closes issue #17732) Reported by: pabelanger
2010-07-27 22:30 +0000 [r280019-280020] Sean Bright <sean@malleable.com>
* main/editline/el.h, main/term.c, main/cli.c,
main/editline/parse.c, main/editline/tokenizer.c,
main/editline/config.sub, main/editline/parse.h,
main/editline/tokenizer.h, configure, main/editline/histedit.h,
main/editline/sig.c, main/editline/PLATFORMS,
main/editline/sig.h, main/editline/key.c, main/editline/editrc.5,
main/editline/np/fgetln.c, main/editline/key.h,
main/editline/TEST/test.c, main/Makefile,
main/editline/configure, main/editline/Makefile.in, configure.ac,
main/editline/configure.in, main/editline/readline/readline.h,
main/editline/README, main/editline/editline.3,
main/editline/vi.c, main/editline/sys.h, main/editline/emacs.c,
main/asterisk.c, main/editline/install-sh, main/editline/term.c,
main/editline/config.guess, main/editline/read.c,
main/editline/term.h, main/editline/map.c,
main/editline/np/strlcpy.c, main/editline (added),
main/editline/config.h.in, main/editline/read.h,
main/editline/tty.c, main/editline/np/unvis.c,
main/editline/prompt.c, main/editline/map.h, main/editline/tty.h,
main/editline/chared.c, main/editline/prompt.h,
main/editline/np/strlcat.c, main/editline/chared.h,
main/editline/np, main/editline/TEST, main/editline/refresh.c,
main/editline/history.c, main/editline/readline,
include/asterisk/term.h, main/editline/refresh.h,
main/editline/search.c, main/editline/hist.c,
main/editline/search.h, main/editline/hist.h,
main/editline/np/vis.c, build_tools/menuselect-deps.in, main,
main/editline/readline.c, main/editline/np/vis.h,
main/editline/INSTALL, makeopts.in, main/editline/CHANGES,
main/editline/common.c, main/xmldoc.c, main/editline/makelist.in,
include/asterisk/autoconfig.h.in, main/editline/el.c: Revert
r280019 for now - This was poorly executed.
* include/asterisk/term.h, makeopts.in, main/asterisk.c,
main/xmldoc.c, main/cli.c, main/term.c, main/editline (removed),
build_tools/menuselect-deps.in, configure,
include/asterisk/autoconfig.h.in, main/Makefile, configure.ac,
main: Add ability to use system libedit and update bundled
libedit. The version of libedit that is bundled with asterisk is
old and has some bugs. This patch updates the bundled version of
libedit within asterisk, and also updates asterisk to use the
system libedit instead if one is available (and pkg-config is
available). This review integrates several patches from other
users specifically kkm and tzafrir. (closes issue #15929)
Reported by: kkm Patches: 015929-astcli-editrc-trunk.240324.diff
uploaded by kkm (license 888) (issue #16858) Reported by:
jw-asterisk (closes issue #17039) Reported by: tzafrir Patches:
0001-allow-using-system-copy-of-libedit.patch uploaded by tzafrir
(license 46) Review: https://reviewboard.asterisk.org/r/807/
2010-07-27 21:16 +0000 [r279953] Russell Bryant <russell@digium.com>
* res/ais, main/db1-ast/mpool, Makefile.rules, res/snmp, cdr,
formats, codecs/gsm/src, funcs, bridges, codecs/lpc10,
main/db1-ast/btree, configure, main/editline, codecs/g722, main,
main/db1-ast/recno, channels/sip, makeopts.in, pbx, res, res/ael,
channels, main/stdtime, main/editline/np, codecs, utils,
main/db1-ast/hash, cel, apps, configure.ac, main/db1-ast/db: Add
--enable-coverage option to configure script. This option enables
the proper compiler flags for tracking code coverage, which is
useful along side automated testing.
2010-07-27 20:57 +0000 [r279949] David Vossel <dvossel@digium.com>
* main/audiohook.c, main/channel.c, /,
include/asterisk/audiohook.h: Merged revisions 279946 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r279946 | dvossel | 2010-07-27 15:54:32 -0500
(Tue, 27 Jul 2010) | 24 lines Merged revisions 279945 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r279945 | dvossel | 2010-07-27 15:33:40 -0500 (Tue, 27 Jul 2010)
| 19 lines remove empty audiohook write list on channel If a
channel has an audiohook write list created on it, that list
stays on the channel until the channel is destroyed. There is no
reason to keep that list on the channel if it becomes empty. If
it is empty that just means we are doing needless translating for
every ast_read and ast_write. This patch removes the audiohook
list from the channel once it is detected to be empty on either a
read or write. If a audiohook is added back to the channel after
this list is destroyed, the list just gets recreated as if it
never existed to begin with. (closes issue #17630) Reported by:
manvirr Review: https://reviewboard.asterisk.org/r/799/ ........
................
2010-07-27 19:50 +0000 [r279916] Russell Bryant <russell@digium.com>
* channels/sig_pri.c, channels/chan_dahdi.c: Fix inband DTMF
detection on outgoing ISDN calls. This is a regression from the
sig_pri split from chan_dahdi. When a call is first initiated,
the inband DTMF detector is not enabled if it's an outgoing ISDN
call. However, it needs to be turned on once the media path
starts up. This handling was put back in the open_media()
callback of chan_dahdi. In sig_pri, open_media() calls were added
to a few places where it was needed, including handling of
PRI_EVENT_RINGING, PRI_EVENT_PROGRESS, and PRI_EVENT_PROCEEDING.
Thanks to rmudgett for helping me with the patch!
2010-07-27 18:54 +0000 [r279887] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Fix parsing error in sip_sipredirect(). The
code was written in a way that did a bad job of parsing the port
out of a URI. Specifically, it would do badly when dealing with
an IPv6 address. In this particular scenario, there was no value
from parsing the port out, so I just removed that logic. And
while I was messing around in the function, I changed some
variable names to be more descriptive. (closes issue #17661)
Reported by: oej Patches: 17661.diff uploaded by mmichelson
(license 60)
2010-07-27 16:40 +0000 [r279850] Jason Parker <jparker@digium.com>
* sounds/Makefile, /: Merged revisions 279849 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r279849 | qwell | 2010-07-27 11:39:16 -0500 (Tue, 27 Jul 2010) |
1 line Simply sounds/Makefile some more. ........
2010-07-27 16:09 +0000 [r279817] David Vossel <dvossel@digium.com>
* main/netsock2.c, channels/chan_sip.c: fix sip transaction match
with authentication, fix confusing log message when using
getaddrinfo
2010-07-27 16:06 +0000 [r279815] Russell Bryant <russell@digium.com>
* channels/chan_dahdi.c: Support "channels" in addition to
"channel" in chan_dahdi.conf. Review:
https://reviewboard.asterisk.org/r/804
2010-07-27 15:15 +0000 [r279785] Mark Michelson <mmichelson@digium.com>
* /, channels/chan_sip.c: Merged revisions 279784 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r279784 | mmichelson | 2010-07-27 10:13:24 -0500 (Tue, 27 Jul
2010) | 14 lines Fix bad behavior of dynamic_exclude_static
option in sip.conf. We were attempting to create a contactdeny
rule based on the peer's IP address before the peer's IP address
had been set. By moving the processing further down in the
function, we can ensure stuff works as we expect for it to.
(closes issue #17717) Reported by: mmichelson Patches:
17717.patch uploaded by mmichelson (license 60) Tested by:
DennisD ........
2010-07-27 02:57 +0000 [r279726-279755] Paul Belanger <paul.belanger@polybeacon.com>
* channels/chan_dahdi.c: If dringXcontext is null, fallback to
default context value. (closes issue #17693) Reported by:
iasgoscouk Patches: issue17693.patch uploaded by pabelanger
(license 224) Tested by: iasgoscouk Review:
https://reviewboard.asterisk.org/r/803/
* main/http.c: Use ast_sockaddr_setnull() when http is not enabled.
Otherwise, ast_tcptls_server_start() will still start http.
(closes issue #17708) Reported by: pabelanger Patches: http.patch
uploaded by pabelanger (license 224)
2010-07-26 Leif Madsen <lmadsen@digium.com>
* Asterisk 1.8.0-beta2 Released.
2010-07-26 23:29 +0000 [r279689] Paul Belanger <paul.belanger@polybeacon.com>
* UPGRADE.txt, CHANGES: Updated documentation for FAX logger level.
2010-07-26 23:03 +0000 [r279658] Jason Parker <jparker@digium.com>
* sounds/Makefile (added), /, sounds/Makefile.380 (removed),
configure, include/asterisk/autoconfig.h.in, sounds/Makefile.381
(removed), configure.ac: Merged revisions 279657 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
........ r279657 | qwell | 2010-07-26 17:59:52 -0500 (Mon, 26 Jul
2010) | 5 lines Really fix sounds Makefile (and make it
readableish). There was a rather large syntax error that should
have caused ALL versions of GNU make to fail. I don't know how it
worked. ........
2010-07-26 21:53 +0000 [r279636] Russell Bryant <russell@digium.com>
* main/channel.c: Ignore a control subclass of -1 in
ast_waitfordigit_full().
2010-07-26 21:20 +0000 [r279599-279619] Tilghman Lesher <tlesher@digium.com>
* /, configure, configure.ac: Merged revisions 279609 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
........ r279609 | tilghman | 2010-07-26 16:18:17 -0500 (Mon, 26
Jul 2010) | 2 lines Dunno why this worked on my machine, but it
works better this way. ........
* res/res_config_ldap.c, /: Merged revisions 279597 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
........ r279597 | ghenry | 2010-07-26 15:25:54 -0500 (Mon, 26
Jul 2010) | 13 lines Apply all patches in:
https://issues.asterisk.org/view.php?id=13573 (closes issue
#13573) Reported by: navkumar Patches:
res_config_ldap-category.diff uploaded by navkumar (license 580)
res_config_ldap.patch uploaded by bencer (license 961)
res_config_ldap uploaded by bencer (license 961) Tested by:
suretec ........
* /: Reverting property remove
2010-07-26 20:58 +0000 [r279598] Gavin Henry <ghenry@suretecsystems.com>
* /: Merged revisions 279597 via svnmerge from
https://origsvn.digium.com/svn/asterisk/1.6.2
-----------------------------------------------------------------------
r279597 | ghenry | 2010-07-26 15:25:53 -0500 (Mon, 26 Jul 2010) |
13 lines Apply all patches in:
https://issues.asterisk.org/view.php?id=13573 [^] (closes issue
0013573) Reported by: navkumar Patches:
res_config_ldap-category.diff uploaded by navkumar (license 580)
res_config_ldap.patch uploaded by bencer (license 961)
res_config_ldap uploaded by bencer (license 961) Tested by:
suretec
------------------------------------------------------------------------
2010-07-26 19:59 +0000 [r279568] David Vossel <dvossel@digium.com>
* channels/sip/include/sip.h,
channels/sip/include/reqresp_parser.h, channels/chan_sip.c,
channels/sip/reqresp_parser.c: transaction matching using top
most Via header This patch modifies the way chan_sip.c does
transaction to dialog matching. Asterisk now stores information
in the top most Via header of the initial incoming request and
compares that against other Requests that have the same call-id.
This results in Asterisk being able to detect a forked call in
which it has received multiple legs of the fork. I completely
stripped out the previous matching code and made the comparisons
a little more explicit and easier to understand. My comments in
the code should offer all the details involving this patch. This
patch also fixes a bug with the usage of the OBJ-MULTIPLE flag to
find multiple dialogs with the same call-id. Since the callback
function was returning (CMP_MATCH | CMP_STOP) only the first item
found was being returned. I fixed this by making a new callback
function for finding multiple dialogs that only returns
(CMP_MATCH) on a match allowing for multiple items to be
returned. Review: https://reviewboard.asterisk.org/r/776/
2010-07-26 19:51 +0000 [r279566] Paul Belanger <paul.belanger@polybeacon.com>
* UPGRADE.txt, CHANGES, configs/logger.conf.sample: Add
documentation for FAX logger level. (closes issue #17715)
Reported by: vrban Patches: 17715.patch uploaded by pabelanger
(license 224) Tested by: vrban
2010-07-26 19:18 +0000 [r279562] Tilghman Lesher <tlesher@digium.com>
* sounds/Makefile (removed), /, sounds/Makefile.380 (added),
configure, include/asterisk/autoconfig.h.in, sounds/Makefile.381
(added), configure.ac: Merged revisions 279561 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r279561 | tilghman | 2010-07-26 14:15:59 -0500 (Mon, 26 Jul 2010)
| 2 lines Use a special Makefile for noobs who still have GNU
Make 3.80. ........
2010-07-26 16:04 +0000 [r279504] Mark Michelson <mmichelson@digium.com>
* configure, include/asterisk/autoconfig.h.in, configure.ac,
channels/sip/reqresp_parser.c: Allow for systems without locale
support to be usable. A recent change to SIP URI comparison code
added a locale-specific string comparison to the mix, and certain
systems do not support such functions. This fix allows for those
systems to still use Asterisk 1.8 (closes issue #17697) Reported
by: pprindeville Patches: asterisk-trunk-bugid17697.patch
uploaded by pprindeville (license 347) Tested by: mmichelson
2010-07-26 15:43 +0000 [r279502] Sean Bright <sean@malleable.com>
* autoconf/ast_ext_lib.m4, /: Merged revisions 279501 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
........ r279501 | seanbright | 2010-07-26 11:41:13 -0400 (Mon,
26 Jul 2010) | 5 lines Expand the correct value within
AST_OPTION_ONLY. (closes issue #17703) Reported by: stuarth
........
2010-07-26 03:27 +0000 [r279472] Tilghman Lesher <tlesher@digium.com>
* formats/format_sln16.c, formats/format_wav_gsm.c,
formats/format_siren7.c, formats/format_ilbc.c,
formats/format_vox.c, formats/format_pcm.c,
formats/format_h263.c, formats/format_g723.c,
formats/format_h264.c, formats/format_g726.c,
formats/format_jpeg.c, formats/format_siren14.c,
formats/format_gsm.c, formats/format_g719.c,
formats/format_g729.c, formats/format_sln.c,
formats/format_wav.c, formats/format_ogg_vorbis.c: Formats need
to load before apps, because some apps call
ast_format_str_reduce() at load time.
2010-07-25 21:26 +0000 [r279442] Paul Belanger <paul.belanger@polybeacon.com>
* tests/test_func_file.c: Add trailing backslash to silence warning
message.
2010-07-25 18:21 +0000 [r279390-279410] Tilghman Lesher <tlesher@digium.com>
* cdr/cdr_odbc.c: Don't re-register CDR module on reload. (closes
issue #17304) Reported by: jnemeth Patches:
20100507__issue17304.diff.txt uploaded by tilghman (license 14)
Tested by: jnemeth
* main/logger.c: Don't assume qlog is open. (closes issue #17704)
Reported by: vrban Patches: issue17704.patch uploaded by
pabelanger (license 224) Tested by: vrban
2010-07-24 23:58 +0000 [r279348] Bradley Latus <brad.latus@gmail.com>
* doc/asterisk.8: Minor update to man page
2010-07-24 20:47 +0000 [r279273-279314] Paul Belanger <paul.belanger@polybeacon.com>
* Makefile: Remove duplicate -c flag when using $(INSTALL) (closes
issue #17695) Reported by: pabelanger Patches: Makefile.diff
uploaded by pabelanger (license 224)
* include/asterisk/netsock2.h: Check if ast_sockaddr is NULL then
return. (closes issue #17677) Reported by: outcast Patches:
issue0017677.patch uploaded by pabelanger (license 224) Tested
by: elguero
* main/manager.c: Default sin_family to AF_INET for TCP / TLS
Bindaddress. Otherwise, 'manager show settings' will generate
errors if manager is not enabled.
2010-07-23 22:20 +0000 [r279227] Richard Mudgett <rmudgett@digium.com>
* apps/app_queue.c, apps/app_dial.c, /: Merged revisions 279207 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r279207 | rmudgett | 2010-07-23 17:11:23 -0500
(Fri, 23 Jul 2010) | 14 lines Merged revisions 279206 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r279206 | rmudgett | 2010-07-23 16:56:44 -0500 (Fri, 23 Jul 2010)
| 7 lines SIP promiscuous redirect could fail to dial the
redirect. The ast_channel was created with one variable to
ast_request() but the call to ast_call() that initiates the
outgoing call was using a different variable. The two variables
are not equivalent if the call_forward string included a channel
technology specifier. e.g., SIP/200 ........ ................
2010-07-12 Leif Madsen <lmadsen@digium.com>
* Asterisk 1.8.0-beta1 Released.
2010-07-23 18:56 +0000 [r279113] Tilghman Lesher <tlesher@digium.com>
* res/res_odbc.c: Silly 64-bit compilers (who uses 64-bit anyway?)
2010-07-23 18:23 +0000 [r279056-279094] Russell Bryant <russell@digium.com>
* /: fix up properties on 1.8 branch
* / (added): Create a branch for Asterisk 1.8.
___ _ _ _ _ ___
/ _ \ ___| |_ ___ _ __(_)___| | __ / | ( _ )
| |_| / __| __/ _ \ '__| / __| |/ / | | / _ \
| _ \__ \ || __/ | | \__ \ < | || (_) |
|_| |_|___/\__\___|_| |_|___/_|\_\ |_(_)___/
2010-07-23 17:05 +0000 [r278982-278985] Tilghman Lesher <tlesher@digium.com>
* autoconf/ast_check_pwlib.m4, /, configure, configure.ac: Merged
revisions 278984 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r278984 | tilghman | 2010-07-23 12:04:15 -0500 (Fri, 23 Jul 2010)
| 5 lines Establish a maximum version for openh323 (i.e. not
opal), because chan_h323 will fail to load, even if it links.
(issue #17679) Reported by: am ........
* /, main/asterisk.c: Merged revisions 278981 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r278981 | tilghman | 2010-07-23 11:42:25 -0500 (Fri, 23 Jul 2010)
| 8 lines Avoid race with consolethread on shutdown (on parallel
processors). (closes issue #17080) Reported by: sybasesql
Patches: 20100721__issue17080.diff.txt uploaded by tilghman
(license 14) Tested by: sybasesql ........
2010-07-23 16:33 +0000 [r278980] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c, channels/sip/reqresp_parser.c,
channels/sip/include/reqresp_parser.h: SIP URI comparison fixes.
This initially was created to work around the issue of using a
string comparison instead of a binary comparison for IP
addresses. It evolved a bit when test cases were created and it
was discovered that comparison of URI parameters was not working
exactly as it should. sip_uri_cmp() and its helpers have been
moved to reqresp_parser.c and a new test has been added. (closes
issue #17662) Reported by: oej Review:
https://reviewboard.asterisk.org/r/792
2010-07-23 16:19 +0000 [r278957] Tilghman Lesher <tlesher@digium.com>
* include/asterisk/res_odbc.h, res/res_config_odbc.c,
configs/extconfig.conf.sample, CHANGES, main/config.c,
res/res_odbc.c, configs/res_odbc.conf.sample: Merge the realtime
failover branch
2010-07-23 16:07 +0000 [r278947] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
* doc/asterisk.8: Some left-over hyphen-minus fixes in the man page
2010-07-23 15:57 +0000 [r278944-278945] Russell Bryant <russell@digium.com>
* channels/chan_sip.c: ... just kidding. Enable SIP by default. :-)
* channels/chan_sip.c: Disable SIP support by default for Asterisk
1.8.
2010-07-23 15:52 +0000 [r278943] Mark Michelson <mmichelson@digium.com>
* addons/chan_ooh323.c: Well, who knew chan_ooh323 used udptl? I
sure didn't!
2010-07-23 15:41 +0000 [r278942] Richard Mudgett <rmudgett@digium.com>
* channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_pri.c:
Rename sig_pri_pri to sig_pri_span. More descriptive of concept.
2010-07-23 15:16 +0000 [r278908] Mark Michelson <mmichelson@digium.com>
* main/udptl.c, channels/chan_sip.c, include/asterisk/udptl.h,
channels/sip/include/sip.h: Allow IPv6 addresses for UDPTL
streams. Review: https://reviewboard.asterisk.org/r/795
2010-07-23 13:37 +0000 [r278875] Olle Johansson <oej@edvina.net>
* res/res_config_ldap.c: Minor corrections to the LDAP realtime
driver Review: https://reviewboard.asterisk.org/r/798/ Thanks
Mark for a quick review!
2010-07-23 13:26 +0000 [r278873] Paul Belanger <paul.belanger@polybeacon.com>
* Makefile, agi/Makefile, sounds/Makefile: Portability updates for
Makefiles. When possible, use $(INSTALL). This allows us to use
the functionality within install for setting directory / file
permissions, a requirement for unprivileged installation. Also
move any directory we plan to create within the installdirs
macro. Plus various other formatting issues. (issue #17436)
Reported by: pabelanger Patches: non-root.patch.v8 uploaded by
pabelanger (license 224) Tested by: pabelanger Review:
https://reviewboard.asterisk.org/r/654/
2010-07-23 11:01 +0000 [r278809-278841] Alec L Davis <sivad.a@paradise.net.nz>
* channels/chan_dahdi.c, channels/sig_analog.c: missed FXS kewl
start polarityswitch when finally on hook. (issue #17318)
* channels/chan_dahdi.c, configs/chan_dahdi.conf.sample,
channels/sig_analog.c, channels/sig_analog.h: Support FXS module
Polarity Reversal on remote party Answer and Hangup FXS lines
normally connect to a telephone. However, when FXS lines are
routed to an external PBX or Key System to act as "external" or
"CO" lines, it is extremely difficult, if not impossible for the
external PBX to know when the call has been disconnected without
receiving a polarity reversal on the line. Now using
answeronpolarityswitch and hanguponpolarityswitch keywords that
previously were used only for FXO ports, now applies like
functionality for an FXS port, but from the connected equipment's
point of view. (closes issue #17318) Reported by: armeniki
Patches: fxs_linepolarity.diff5.txt uploaded by alecdavis
(license 585) Tested by: alecdavis Review:
https://reviewboard.asterisk.org/r/797/
2010-07-22 21:16 +0000 [r278777] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c: DNID not cleared when channel hang up
(Affects PRI and SS7) The "dahdi show channels" CLI command still
reports the DNID of the previous call even if the call is already
hang up. The "dahdi show channels" command of older releases
clear the DNID once the channel is hang up. Regression from the
sig_analog/sig_pri extraction from chan_dahdi. (closes issue
#17623) Reported by: klaus3000 Patches: issue17623.patch uploaded
by rmudgett (license 664) Tested by: rmudgett
2010-07-22 19:45 +0000 [r278708] Jeff Peeler <jpeeler@digium.com>
* main/xmldoc.c: Add method for finding XML doc files for systems
that don't support GLOB_BRACE. In particular, Solaris and perhaps
others do not support the above mentioned GNU extension. In this
case the paths are simply expanded without the braces and the
calls to glob are made separately. Note: I could not explain
memory allocation failures that were being reported from within
libxml itself when making calls to glob without using
GLOB_NOCHECK. This is the only reason why that flag is being
used. (closes issue #15402) Reported by: snuffy Patches:
bug_xmlpatt-v3.diff uploaded by snuffy (license 35), modified by
me
2010-07-22 14:58 +0000 [r278620] Mark Michelson <mmichelson@digium.com>
* main/channel.c, /: Merged revisions 278618 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r278618 | mmichelson | 2010-07-22 09:55:04 -0500 (Thu, 22 Jul
2010) | 13 lines Allow PLC to function properly when channels use
SLIN for audio. If a channel involved in a bridge was using SLIN
audio, then translation paths were not guaranteed to be set up
properly since in all likelihood the number of translation steps
was only 1. This patch enforces the transcode_via_slin behavior
if transcode_via_slin or generic_plc is enabled and one of the
formats to make compatible is SLIN. AST-352 ........
2010-07-22 14:56 +0000 [r278619] David Vossel <dvossel@digium.com>
* channels/chan_sip.c: update sip subscription debug message to a
warning message If the Expire header of a SUBSCRIBE is less that
our expiremin, a log warning will be displayed.
2010-07-22 05:29 +0000 [r278579] Tilghman Lesher <tlesher@digium.com>
* include/asterisk/doxyref.h: Add the full current set of CDR
drivers
2010-07-21 19:16 +0000 [r278539] David Vossel <dvossel@digium.com>
* tests/test_func_file.c: make func_file unit test's category
consistent with other tests
2010-07-21 19:11 +0000 [r278538] Terry Wilson <twilson@digium.com>
* channels/iax2-parser.h, include/asterisk/crypto.h,
main/aescrypt.c (removed), include/asterisk/aes_internal.h
(removed), funcs/func_aes.c, res/res_crypto.c, main/aestab.c
(removed), main/aesopt.h (removed), include/asterisk/aes.h
(removed), main/aeskey.c (removed), pbx/pbx_dundi.c,
channels/chan_iax2.c, res/res_crypto.exports.in,
pbx/dundi-parser.h: Remove built-in AES code and use optional_api
instead Review: https://reviewboard.asterisk.org/r/793/
2010-07-21 18:52 +0000 [r278536] David Vossel <dvossel@digium.com>
* channels/chan_sip.c: send "423 Interval too small" Response to
Subscribe with Expires less that min allowed [RFC3265]3.1.6.1....
The notifier MAY also check that the duration in the "Expires"
header is not too small. If and only if the expiration interval
is greater than zero AND smaller than one hour AND less than a
notifier- configured minimum, the notifier MAY return a "423
Interval too small" error which contains a "Min-Expires" header
field. The "Min- Expires" header field is described in SIP [1].
2010-07-21 17:44 +0000 [r278501] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
* channels/chan_dahdi.c, channels/sig_analog.c: Fix invalid test
for rxisoffhook in FXO channels This fixes some cases of no
outgoing calls on FXO before an incoming call. Remove an
unnecessary testing of an "off-hook" bit from DAHDI for FXO
(KS/GS) channels.In some cases the bit would not be initialized
properly before the first inbound call and thus prevent an
outgoing call. If those tests are actually required by anybody,
they should define DAHDI_CHECK_HOOKSTATE in channels/sig_analog.c
. (closes issue #14577) Reported by: jkroon Patches:
asterisk_chan_dahdi_hookstate_fix_trunk_new.diff uploaded by
frawd (license 610) Tested by: frawd Review:
https://reviewboard.asterisk.org/r/699/
2010-07-21 16:15 +0000 [r278465] Russell Bryant <russell@digium.com>
* res/res_timing_pthread.c: Use poll() instead of select() in
res_timing_pthread to avoid stack corruption. This code did not
properly check FD_SETSIZE to ensure that it did not try to
select() on fds that were too large. Switching to poll() removes
the limitation on the maximum fd value. (closes issue #15915)
Reported by: keiron (closes issue #17187) Reported by: Eddie
Edwards (closes issue #16494) Reported by: Hubguru (closes issue
#15731) Reported by: flop (closes issue #12917) Reported by:
falves11 (closes issue #14920) Reported by: vrban (closes issue
#17199) Reported by: aleksey2000 (closes issue #15406) Reported
by: kowalma (closes issue #17438) Reported by: dcabot (closes
issue #17325) Reported by: glwgoes (closes issue #17118) Reported
by: erikje possibly other issues, too ...
2010-07-21 15:56 +0000 [r278463] Tilghman Lesher <tlesher@digium.com>
* apps/app_meetme.c: Ensure realtime conferences are treated the
same as static conferences when trying to find an empty one.
Also, parse the useropts properly, when retrieving from realtime,
and add them to the existing flags. (closes issue #17502)
Reported by: kenji Patches: 20100720__issue17502.diff.txt
uploaded by tilghman (license 14) Tested by: kenji
2010-07-21 15:54 +0000 [r278426-278462] Matthew Nicholson <mnicholson@digium.com>
* res/res_fax_spandsp.c: Properly show the current page being
transfered for 'fax show session'
* channels/chan_sip.c: Properly set the port number for UDPTL media
sessions.
* res/res_fax.c: Don't print failure status when the remote end
hangs up, it may not be an actual failure.
2010-07-21 13:02 +0000 [r278425] Russell Bryant <russell@digium.com>
* main/features.c, UPGRADE.txt, configs/features.conf.sample:
Update documentation for 'comebacktoorigin' in featuers.conf. The
documentation for this option did not match the code. Fix that
along with some minor cleanups to the code along the way.
Document a slight change in behavior (to something that was
previously undocumented) in UPGRADE.txt.
2010-07-21 06:45 +0000 [r278393] Tilghman Lesher <tlesher@digium.com>
* channels/chan_iax2.c: Change order so that it more closely
matches the related SIP command. (closes issue #17648) Reported
by: GMLudo Review: https://reviewboard.asterisk.org/r/789/
2010-07-21 03:53 +0000 [r278361] Jeff Peeler <jpeeler@digium.com>
* channels/chan_dahdi.c: include stat.h for everybody, needed for
device2chan
2010-07-20 23:23 +0000 [r278275-278307] Tilghman Lesher <tlesher@digium.com>
* res/res_config_pgsql.c, main/logger.c, CHANGES,
contrib/realtime/mysql/queue_log.sql (added),
configs/logger.conf.sample: Separate queue_log arguments into
separate fields, and allow the text file to be used, even when
realtime is used. (closes issue #17082) Reported by: coolmig
Patches: 20100720__issue17082.diff.txt uploaded by tilghman
(license 14) Tested by: coolmig
* /, apps/app_voicemail.c: Merged revisions 278261 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r278261 | tilghman | 2010-07-20 17:23:13 -0500 (Tue, 20
Jul 2010) | 7 lines Delete IMAP messages in reverse order, to
ensure reordering after each expunge does not cause deletion of
the wrong message. (closes issue #16350) Reported by: noahisaac
Patches: 20100623__issue16350.diff.txt uploaded by tilghman
(license 14) ........
2010-07-20 22:38 +0000 [r278274] Richard Mudgett <rmudgett@digium.com>
* channels/sig_pri.c: Reference correct struct member for unlikely
event PRI_EVENT_CONFIG_ERR.
2010-07-20 22:26 +0000 [r278272] Tilghman Lesher <tlesher@digium.com>
* main/autoservice.c, /, main/features.c,
include/asterisk/channel.h: Merged revisions 278167 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r278167 | tilghman | 2010-07-20 15:59:06 -0500 (Tue, 20
Jul 2010) | 4 lines Do not queue up DTMF frames while a call is
on hold. (Fixes ABE-2110) ........
2010-07-20 21:41 +0000 [r278234] David Vossel <dvossel@digium.com>
* channels/chan_sip.c: fixes sip CANCEL race condition If Asterisk
sends a 4xx error and the other side sends a CANCEl before
receiving the 4xx and responding with the ACK, Asterisk will
process the CANCEL and send a 487 Request Terminated as a new
final response to the INVITE. Since we are issuing a new final
response to the INVITE, the old one must be pretend_acked else it
will keep retransmitting.
2010-07-20 21:01 +0000 [r278168] Matthew Nicholson <mnicholson@digium.com>
* res/res_fax.c: This commit contains several changes to the way
output channel variables are handled. FAX output channel
variables will now match the values reported by FAXOPT() and
should be set in all failure and success cases. This commit also
contains a few modifications to the way FAXOPT() variables are
populated in a few spots and fixes for some reference count leaks
of the session details structure in some failure cases. Also
found and fixed more cases where FAXOPT(status) may not have
gotten set. FAX-214 FAX-203
2010-07-20 19:35 +0000 [r278132] Tilghman Lesher <tlesher@digium.com>
* cel/cel_pgsql.c, cdr/cdr_sqlite3_custom.c, channels/chan_local.c,
res/res_timing_dahdi.c, cdr/cdr_adaptive_odbc.c,
res/res_calendar_caldav.c, formats/format_sln16.c,
formats/format_wav_gsm.c, channels/chan_iax2.c, main/config.c,
main/loader.c, res/res_rtp_multicast.c, channels/chan_dahdi.c,
res/res_smdi.c, channels/chan_skinny.c,
include/asterisk/module.h, formats/format_pcm.c,
channels/chan_alsa.c, formats/format_h263.c, res/res_curl.c,
cdr/cdr_odbc.c, formats/format_jpeg.c, res/res_speech.c,
formats/format_gsm.c, cdr/cdr_manager.c, formats/format_g719.c,
res/res_calendar_exchange.c, cel/cel_tds.c, formats/format_wav.c,
channels/chan_bridge.c, channels/chan_agent.c,
formats/format_ogg_vorbis.c, res/res_monitor.c,
res/res_calendar_ews.c, res/res_config_curl.c,
channels/chan_misdn.c, funcs/func_curl.c,
res/res_timing_kqueue.c, formats/format_g726.c, main/asterisk.c,
res/res_odbc.c, cel/cel_adaptive_odbc.c, res/res_calendar.c,
cel/cel_radius.c, channels/chan_multicast_rtp.c,
apps/app_meetme.c, formats/format_sln.c, res/res_musiconhold.c,
channels/chan_gtalk.c, cdr/cdr_pgsql.c, cdr/cdr_radius.c,
res/res_jabber.c, res/res_config_sqlite.c,
formats/format_siren7.c, cdr/cdr_csv.c, formats/format_ilbc.c,
res/res_config_odbc.c, cel/cel_manager.c, cel/cel_custom.c,
cdr/cdr_sqlite.c, res/res_agi.c, res/res_timing_timerfd.c,
apps/app_confbridge.c, formats/format_h264.c,
res/res_config_ldap.c, addons/chan_mobile.c,
formats/format_siren14.c, cdr/cdr_custom.c, channels/chan_mgcp.c,
res/res_rtp_asterisk.c, res/res_config_pgsql.c,
res/res_calendar_icalendar.c, channels/chan_sip.c,
cdr/cdr_syslog.c, res/res_fax.c, res/res_crypto.c,
res/res_adsi.c, include/asterisk/config.h, pbx/pbx_lua.c,
channels/chan_console.c, apps/app_queue.c, cdr/cdr_tds.c,
res/res_srtp.c, channels/chan_jingle.c, formats/format_vox.c,
res/res_timing_pthread.c, channels/chan_h323.c,
cel/cel_sqlite3_custom.c, formats/format_g723.c,
funcs/func_devstate.c, formats/format_g729.c,
addons/res_config_mysql.c: Add load priority order, such that
preload becomes unnecessary in most cases
2010-07-20 18:11 +0000 [r278051-278096] Russell Bryant <russell@digium.com>
* contrib/scripts/install_prereq: Add a package to install_prereq.
* channels/chan_local.c: Only call ast_channel_cc_params_init() if
allocating a channel succeeds.
2010-07-20 16:50 +0000 [r278024] Tilghman Lesher <tlesher@digium.com>
* main/manager.c, /: Merged revisions 278023 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r278023 | tilghman | 2010-07-20 11:37:18 -0500 (Tue, 20 Jul 2010)
| 7 lines Off-by-one error (closes issue #16506) Reported by:
nik600 Patches: 20100629__issue16506.diff.txt uploaded by
tilghman (license 14) ........
2010-07-19 21:07 +0000 [r277945] Jean Galarneau <jgalarneau@digium.com>
* /, main/features.c: Merged revisions 277906 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r277906 | jeang | 2010-07-19 15:16:36 -0500 (Mon, 19 Jul 2010) |
7 lines Avoid trying to pickup a parked extension before the park
operation is completed. A crash could occur if the extension is
picked up while the parking extension is being announced. Testing
pu->notquiteyet while searching for a parked extension resolves
this crash. (ABE-2418) ........
2010-07-19 17:16 +0000 [r277872-277873] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c, configs/sip.conf.sample,
channels/sip/include/sip.h: Fix port setting of external address
in SIP. There are two changes here: 1. Since the externip setting
can now have a port attached to it, calling it "externip" is
misleading. The option is now documented and parsed as
"externaddr." This also extends to the "matchexterniplocally"
setting. It is now documented and parsed as
"matchexternaddrlocally." The old names for the options may still
be used, but they are no longer used in the sip.conf.sample file.
2. If no port is set for the externaddr, and UDP is the transport
to be used, then we will set the port of the externaddr to that
of the udpbindaddr. This was how things worked prior to the IPv6
merge, so this is a regression fix. (closes issue #17665)
Reported by: mmichelson Patches: 17665.diff#2 uploaded by
pprindeville (license 347) Tested by: pprindeville
* tests/test_acl.c: Remove the fe80:1234::1234 test case from
test_acl.c The ACL test was failing on Mac OS X because it would
convert the above invalid link-local address into fe80::1234
while reporting no error from getaddrinfo(). Linux does not do
this.
2010-07-19 14:39 +0000 [r277837] Jeff Peeler <jpeeler@digium.com>
* channels/chan_dahdi.c, channels/sig_analog.c,
channels/sig_analog.h: Fix regression with distinctive ring
detection. The issue here is that passing an array to a function
prohibits the ARRAY_LEN macro from returning the real size. To
avoid this the size is now defined and use of ARRAY_LEN is
avoided. (closes issue #15718) Reported by: alecdavis Patches:
bug15718.patch uploaded by jpeeler (license 325)
2010-07-19 14:17 +0000 [r277814] Mark Michelson <mmichelson@digium.com>
* include/asterisk/acl.h, main/netsock2.c, main/manager.c,
channels/chan_sip.c, channels/chan_skinny.c, tests/test_acl.c,
main/acl.c, include/asterisk/netsock2.h, configs/sip.conf.sample,
channels/chan_iax2.c: Make ACLs IPv6-capable. ACLs can now be
configured to match IPv6 networks. This is only relevant for ACLs
in chan_sip for now since other channel drivers do not support
IPv6 addressing. However, once those channel drivers are
outfitted to support IPv6 addressing, the ACLs will already be
ready for IPv6 support. https://reviewboard.asterisk.org/r/791
2010-07-17 17:42 +0000 [r277773-277775] Tilghman Lesher <tlesher@digium.com>
* /, autoconf/ast_func_fork.m4, configure,
include/asterisk/autoconfig.h.in: Merged revisions 277738 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r277738 | tilghman | 2010-07-17 11:59:11 -0500 (Sat, 17 Jul 2010)
| 5 lines Remove uclibc cross-compile triplet, as uclibc has a
working fork()... it's only uclinux that does not. (closes issue
#17616) Reported by: pprindeville ........
* res/res_config_pgsql.c, res/res_config_odbc.c, /,
include/asterisk/config.h, main/config.c,
addons/res_config_mysql.c: Merged revisions 277568 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r277568 | tilghman | 2010-07-16 16:54:29 -0500 (Fri, 16
Jul 2010) | 8 lines Since we split values at the semicolon, we
should store values with a semicolon as an encoded value. (closes
issue #17369) Reported by: gkservice Patches:
20100625__issue17369.diff.txt uploaded by tilghman (license 14)
Tested by: tilghman ........
2010-07-17 13:10 +0000 [r277703] Russell Bryant <russell@digium.com>
* Makefile, configure, include/asterisk/autoconfig.h.in,
configure.ac, makeopts.in: Allow xmllint to be used for XML docs
validation. xmllint seems to be more commonly available since it
comes with libxml2.
2010-07-17 00:03 +0000 [r277667] Bradley Latus <brad.latus@gmail.com>
* res/res_fax.c: Update res_fax.c to be a good xml citizen. (closes
issues #17667) Reported by: snuffy
2010-07-16 23:23 +0000 [r277657] Tim Ringenbach <tim.ringenbach@gmail.com>
* main/features.c: Merged revisions 277625 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r277625 | tringenbach | 2010-07-16 17:43:39 -0500 (Fri, 16 Jul
2010) | 9 lines Save and restore AST_FLAG_BRIDGE_HANGUP_DONT on
attended transfer. ast_bridge_call() clears
AST_FLAG_BRIDGE_HANGUP_DONT. But during an attended transfer,
ast_bridge_call() is called for a second bridge on the same
channel, and it clears that flag, which still needs to get set
for when the original ast_bridge_call() gets control back and
checks it. Review: https://reviewboard.asterisk.org/r/741
........
2010-07-16 21:24 +0000 [r277530] Matthew Nicholson <mnicholson@digium.com>
* /, channels/chan_sip.c: Merged revisions 277497 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r277497 | mnicholson | 2010-07-16 16:18:38 -0500 (Fri, 16 Jul
2010) | 4 lines Default to no udptl error correction so that
error correction will be disabled in the event that the remote
end indicates that they do not support the error correction mode
we requested. FAX-128 ........
2010-07-16 21:16 +0000 [r277488] Jeff Peeler <jpeeler@digium.com>
* apps/app_queue.c: Fix reporting estimated queue hold time. Just
say the number of seconds (after minutes) rather than doing some
incorrect calculation with respect to minutes. (closes issue
#17498) Reported by: corruptor Patches: holdesecs_bug.diff
uploaded by corruptor (license 253)
2010-07-16 20:35 +0000 [r277484] Tilghman Lesher <tlesher@digium.com>
* include/asterisk/sched.h, main/sched.c: Finally, a method that
really fixes the assertions in chan_iax2.c related to cancelling
lagid. No, replacing usleep(1) with sched_yield() did not have an
effect.
2010-07-16 20:27 +0000 [r277467] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c, /: Merged revisions 277419 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r277419 | rmudgett | 2010-07-16 15:18:54 -0500 (Fri, 16
Jul 2010) | 15 lines priexclusive in chan_dahdi.conf ignored when
reloading dahdi module During a reload, the priexclusive and
outsignalling parameters are not read in from the config file as
intended. Unfortunately, they get set to defaults as a result.
This patch makes sure that they do not get set to defaults during
a reload. (closes issue #17441) Reported by: mtryfoss Patches:
issue17441_v1.4.patch uploaded by rmudgett (license 664)
issue17441_v1.6.2.patch uploaded by rmudgett (license 664)
issue17441_trunk.patch uploaded by rmudgett (license 664) Tested
by: rmudgett ........
2010-07-16 20:25 +0000 [r277452] Tilghman Lesher <tlesher@digium.com>
* res/res_musiconhold.c, contrib/realtime/mysql/musiconhold.sql
(added): Add documentation for MOH realtime fields
2010-07-16 19:32 +0000 [r277409] Matthew Nicholson <mnicholson@digium.com>
* tests/test_devicestate.c: updated devicestate test for device
state changes
2010-07-16 19:22 +0000 [r277366] Jeff Peeler <jpeeler@digium.com>
* apps/app_queue.c: Add missing handling for ringing state for use
with queue empty options. (closes issue #17471) Reported by:
jazzy Patches: app_queue.c.diff uploaded by jazzy (license 1056)
2010-07-16 18:31 +0000 [r277331] Matthew Nicholson <mnicholson@digium.com>
* main/pbx.c, /: Merged revisions 277327 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r277327 | mnicholson | 2010-07-16 13:30:22 -0500 (Fri, 16 Jul
2010) | 8 lines Interpret device state AST_DEVICE_UNKNOWN as
extension state AST_EXTENSION_NOT_INUSE. (closes issue #16035)
Reported by: francesco_r Patches: pbx.c.patch uploaded by
viniciusfontes (license 978) Tested by: francesco_r, agx, lawbar
........
2010-07-16 18:14 +0000 [r277263] Tilghman Lesher <tlesher@digium.com>
* main/manager.c, /: Merged revisions 277261 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r277261 | tilghman | 2010-07-16 13:04:11 -0500 (Fri, 16 Jul 2010)
| 5 lines If variable gotten is not set, will segfault on
Solaris. (closes issue #17636) Reported by: bklang ........
2010-07-16 18:05 +0000 [r277250-277262] Matthew Nicholson <mnicholson@digium.com>
* main/channel.c: Print f->subclass.integer instead of f->subclass.
(fix build breakage introduced in r277250)
* main/channel.c, /: Merged revisions 277247 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r277247 | mnicholson | 2010-07-16 12:29:57 -0500 (Fri, 16 Jul
2010) | 4 lines For pass through DTMF tones, measure the actual
duration between the begin and end packets on the wire. If it is
detected to be less than AST_MIN_DTMF_DURATION, trigger dtmf
emulation. AST-362 ........
2010-07-16 17:13 +0000 [r277183] Paul Belanger <paul.belanger@polybeacon.com>
* /, apps/app_amd.c: Merged revisions 277182 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r277182 | pabelanger | 2010-07-16 13:10:36 -0400 (Fri, 16 Jul
2010) | 8 lines Total analysis time error with SIP and silence
suppression When using app_amd with SIP providers that have
silence suppression on, the iTotalTime count increases
exponentially. (closes issue #17656) Reported by: juls ........
2010-07-16 16:25 +0000 [r277175] Mark Michelson <mmichelson@digium.com>
* channels/sip/reqresp_parser.c: Fix up some weird indentation
problems in reqresp_parser.c
2010-07-16 15:20 +0000 [r277143] Sean Bright <sean@malleable.com>
* main/translate.c: Avoid crashing when installing a duplicate
translation path with a lower cost. (closes issue #17092)
Reported by: moy Patches: translate.rev254273.patch uploaded by
moy (license 222) Tested by: moy
2010-07-16 13:40 +0000 [r277103] Eliel C. Sardanons <eliels@gmail.com>
* CREDITS: Add Despegar.com (my main sponsor) to the CREDITS file.
2010-07-16 13:32 +0000 [r276950-277102] Olle Johansson <oej@edvina.net>
* main/dnsmgr.c, main/srv.c: Formatting changes
* channels/chan_sip.c: Formatting fixes
* configs/sip.conf.sample: Clarify syntax changes
* CREDITS: Adding a few more to the list of CREDITS
* channels/chan_sip.c: Formatting changes (guideline corrections)
Found a unused bag of curly brackets under my table. I always
wondered where they had gone. They where indeed needed in
chan_sip.c
* CREDITS: Adding a few more credits
* channels/chan_sip.c, doc/tex/channelvariables.tex,
configs/sip.conf.sample, CHANGES, channels/sip/include/sip.h: Add
ability to configure the Max-Forwards header in the dialplan, as
well as in sip.conf configuration for the channel and for
devices. The Max-Forwards header is used to prevent loops in a
SIP network. Each intermediary, like SIP proxys and SBCs,
decrement this counter and detects when it reaches zero, at which
point the SIP request is nicely killed in a SIP-friendly way.
Review: https://reviewboard.asterisk.org/r/778/ Thanks to dvossel
for the review and good advice.
* CHANGES, apps/app_queue.c: Add a dialplan function to check if a
queue exists: QUEUE_EXISTS Review:
https://reviewboard.asterisk.org/r/777/
2010-07-16 06:04 +0000 [r276910-276911] Tilghman Lesher <tlesher@digium.com>
* res/res_jabber.c: And yet one more
* res/res_jabber.c: "Item may be used uninitialized in this
function."
2010-07-16 05:42 +0000 [r276909] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Fix reversed logic of if statement. Found
based on message from Philip Prindeville on the Asterisk
Developers mailing list.
2010-07-16 05:38 +0000 [r276830-276908] Tilghman Lesher <tlesher@digium.com>
* configure, configure.ac: Detect the --dynamic-list flag a bit
better
* configure, main/Makefile, configure.ac, makeopts.in: Fix build on
FreeBSD
* tests/test_utils.c: Fix trunk build for Mac OS X 10.6
* contrib/realtime/mysql/iaxfriends.sql,
contrib/realtime/mysql/meetme.sql,
contrib/realtime/postgresql/realtime.sql,
contrib/realtime/mysql/sipfriends.sql: Allow ipaddress to contain
the maximum IPv6 address. Also, update meetme to the full list of
supported fields.
* configure, autoconf/ast_gcc_attribute.m4: Quote AC_SUBST within
m4_ifval, so it does not get prematurely expanded. (closes issue
#17654) Reported by: pprindeville Patches: issue17654.diff
uploaded by qwell (license 4) Tested by: qwell, pprindeville
2010-07-15 20:21 +0000 [r276788] Jeff Peeler <jpeeler@digium.com>
* channels/chan_sip.c: Correct not setting the bindport before
attempting to open the socket. Related to changes from 276571, I
was accidentally testing with a port set in my configuration
causing me to miss this. Also moved the TCP handling as well to
occur before build_peer is called.
2010-07-15 19:46 +0000 [r276731-276769] Tilghman Lesher <tlesher@digium.com>
* configure, include/asterisk/autoconfig.h.in,
include/asterisk/compat.h, configure.ac: Define LLONG_MAX on
systems that do not have it. (closes issue #17644) Reported by:
pprindeville
* configure, main/Makefile, autoconf/ast_gcc_attribute.m4,
configure.ac, makeopts.in: Fix linking asterisk on CentOS 5,
which is using gcc 4.1.1. Gcc 4.1.2 has the real fix. Review:
https://reviewboard.asterisk.org/r/790/
2010-07-15 13:51 +0000 [r276653] Jeff Peeler <jpeeler@digium.com>
* main/channel.c, /: Merged revisions 276652 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r276652 | jpeeler | 2010-07-15 08:48:58 -0500 (Thu, 15 Jul 2010)
| 2 lines In a perfect world, the frame source would never be
NULL. In the meantime, don't crash when it is. ........
2010-07-15 12:21 +0000 [r276616] Russell Bryant <russell@digium.com>
* contrib/scripts/install_prereq: Add lua5.1 to the handy dandy
list of packages.
2010-07-14 22:58 +0000 [r276571] Jeff Peeler <jpeeler@digium.com>
* channels/chan_sip.c: Fix MWI notification transmission problems
over SIP. MWI updates were not being sent if no messages were
found in the event cache. This was corrected since a phone may
need to clear its MWI status configured previously from another
mailbox. Upon module or sip reload, MWI updates could not be sent
due to the sipsock socket not being set early enough in
reload_config. The code handling the descriptor assignment and
such has simply been moved before the call to build_peer. Issuing
a sip reload cleared the IP address of the peer, but skipped
checking the database for registration information. The database
is now checked both for sip reload and actually reloading the
module. If a transmission occurs before the do_monitor thread has
started, do not attempt to send a signal to it. (closes issue
#17398) Reported by: ip-rob
2010-07-14 22:32 +0000 [r276570] Mark Michelson <mmichelson@digium.com>
* res/res_rtp_asterisk.c, main/dnsmgr.c, channels/chan_sip.c,
main/acl.c: Fix errors where incorrect address information was
printed. ast_sockaddr_stringiy_fmt (which is call by all
ast_sockaddr_stringify* functions) uses thread-local storage for
storing the string that it creates. In cases where
ast_sockaddr_stringify_fmt was being called twice within the same
statement, the result of one call would be overwritten by the
result of the other call. This usually was happening in
printf-like statements and was resulting in the same stringified
addressed being printed twice instead of two separate addresses.
I have fixed this by using ast_strdupa on the result of stringify
functions if they are used twice within the same statement. As
far as I could tell, there were no instances where a pointer to
the result of such a call were saved anywhere, so this is the
only situation I could see where this error could occur.
2010-07-14 21:29 +0000 [r276531] Richard Mudgett <rmudgett@digium.com>
* channels/chan_h323.c: Make compile again.
2010-07-14 21:11 +0000 [r276490-276493] Tilghman Lesher <tlesher@digium.com>
* main/loader.c: Oops, merge reverted this fix.
* include/asterisk/adsi.h, include/asterisk/agi.h,
include/asterisk/crypto.h, main/asterisk.dynamics, main/Makefile,
tests/test_utils.c, main/adsistub.c (removed), main/cryptostub.c
(removed), res/res_adsi.c, res/res_crypto.c,
res/res_crypto.exports.in (added), res/res_adsi.exports.in,
main/loader.c, include/asterisk/optional_api.h: Remove the old
stub files, preferring the optional_api method. (closes issue
#17475) Reported by: tilghman Review:
https://reviewboard.asterisk.org/r/695/
2010-07-14 20:15 +0000 [r276441] Kevin P. Fleming <kpfleming@digium.com>
* main/loader.c: Don't try to call an embedded module's
backup_globals() function until after confirming it exists.
2010-07-14 19:51 +0000 [r276439] David Vossel <dvossel@digium.com>
* channels/chan_sip.c: handle special case were "200 Ok" to pending
INVITE never receives ACK Unlike most responses, the 200 Ok to a
pending INVITE Request is acknowledged by an ACK Request. If the
ACK Request for this Response is not received the previous
behavior was to immediately destroy the dialog and hangup the
channel. Now in an effort to be more RFC compliant, instead of
immediately destroying the dialog during this special case,
termination is done with a BYE Request as the dialog is
technically confirmed when the 200 Ok is sent even if the ACK is
never received. The behavior of immediately hanging up the
channel remains. This only affects how dialog termination
proceeds for this one special case. RFC 3261 section 13.3.1.4 "If
the server retransmits the 2xx response for 64*T1 seconds without
receiving an ACK, the dialog is confirmed, but the session SHOULD
be terminated. This is accomplished with a BYE, as described in
Section 15."
2010-07-14 16:58 +0000 [r276393] Richard Mudgett <rmudgett@digium.com>
* channels/chan_vpb.cc, channels/chan_sip.c,
include/asterisk/channel.h, channels/sig_pri.c,
channels/chan_iax2.c, main/cel.c, channels/chan_oss.c,
main/channel.c, main/cdr.c, channels/chan_jingle.c,
channels/chan_usbradio.c, channels/chan_dahdi.c,
channels/chan_phone.c, channels/sig_analog.c,
channels/chan_misdn.c, channels/chan_skinny.c,
channels/chan_h323.c, res/snmp/agent.c, apps/app_amd.c,
funcs/func_callerid.c, channels/sig_ss7.c, channels/chan_mgcp.c:
Expand the caller ANI field to an ast_party_id Expand the ani
field in ast_party_caller and ast_party_connected_line to an
ast_party_id. This is an extension to the ast_callerid
restructuring patch in review:
https://reviewboard.asterisk.org/r/702/ Review:
https://reviewboard.asterisk.org/r/744/
2010-07-14 16:40 +0000 [r276392] David Vossel <dvossel@digium.com>
* channels/chan_sip.c: collapse debug code in retrans_pkt into
separate lines I've been working in this function a bunch lately,
and these huge debug strings are getting annoying.
2010-07-14 16:39 +0000 [r276391] Richard Mudgett <rmudgett@digium.com>
* res/snmp/agent.c: Make compile again.
2010-07-14 16:36 +0000 [r276389] Jeff Peeler <jpeeler@digium.com>
* channels/chan_sip.c: Do not skip sending MWI for a peer if an
address is defined. Really just a merge mistake from IPv6
2010-07-14 16:09 +0000 [r276349] Tim Ringenbach <tim.ringenbach@gmail.com>
* cel/cel_pgsql.c, doc/tex/celdriver.tex, doc/tex/cdrdriver.tex:
Fix documentation for pgsql cel and cdr, and slightly improve
pgsql_cel. Change the documented pgsql schema to use "timestamp"
instead of "time", as the latter is only a time without a date.
Added some missing columns for cel's pgsql schema, and corrected
spelling on some others. Updated cel's uniqueid size to be the
same as the cdr. Added id column to cel's pgsql schema and
updated code to allow unknown columns to get their default value
instead of forcing 0 or empty string. Added microseconds to the
timestamp cel logs to pgsql. Review:
https://reviewboard.asterisk.org/r/734
2010-07-14 15:48 +0000 [r276347] Richard Mudgett <rmudgett@digium.com>
* channels/chan_local.c, addons/chan_ooh323.c,
apps/app_alarmreceiver.c, channels/chan_iax2.c, main/cli.c,
channels/chan_dahdi.c, channels/sig_analog.c,
channels/chan_skinny.c, main/features.c, apps/app_dumpchan.c,
channels/sig_analog.h, apps/app_amd.c, channels/sig_ss7.c,
apps/app_dial.c, main/pbx.c, apps/app_privacy.c, apps/app_fax.c,
channels/chan_agent.c, apps/app_disa.c,
include/asterisk/channel.h, apps/app_talkdetect.c, main/cel.c,
funcs/func_redirecting.c (removed), channels/chan_misdn.c,
apps/app_macro.c, apps/app_zapateller.c, apps/app_voicemail.c,
channels/chan_unistim.c, tests/test_substitution.c,
channels/chan_vpb.cc, apps/app_meetme.c, main/ccss.c,
apps/app_readexten.c, channels/chan_gtalk.c, apps/app_followme.c,
include/asterisk/callerid.h, main/cdr.c, main/channel.c,
channels/chan_phone.c, main/dial.c, apps/app_setcallerid.c,
apps/app_osplookup.c, main/manager.c, apps/app_minivm.c,
res/res_agi.c, main/app.c, apps/app_rpt.c, channels/chan_mgcp.c,
apps/app_parkandannounce.c, apps/app_while.c,
funcs/func_dialplan.c, channels/chan_sip.c, UPGRADE.txt,
channels/chan_console.c, channels/sig_pri.c, apps/app_queue.c,
channels/chan_oss.c, channels/chan_usbradio.c,
channels/chan_jingle.c, funcs/func_blacklist.c,
apps/app_directed_pickup.c, main/file.c,
funcs/func_connectedline.c (removed), channels/chan_h323.c,
main/callerid.c, res/snmp/agent.c, apps/app_sms.c,
apps/app_stack.c, funcs/func_callerid.c: ast_callerid
restructuring The purpose of this patch is to eliminate struct
ast_callerid since it has turned into a miscellaneous collection
of various party information. Eliminate struct ast_callerid and
replace it with the following struct organization: struct
ast_party_name { char *str; int char_set; int presentation;
unsigned char valid; }; struct ast_party_number { char *str; int
plan; int presentation; unsigned char valid; }; struct
ast_party_subaddress { char *str; int type; unsigned char
odd_even_indicator; unsigned char valid; }; struct ast_party_id {
struct ast_party_name name; struct ast_party_number number;
struct ast_party_subaddress subaddress; char *tag; }; struct
ast_party_dialed { struct { char *str; int plan; } number; struct
ast_party_subaddress subaddress; int transit_network_select; };
struct ast_party_caller { struct ast_party_id id; char *ani; int
ani2; }; The new organization adds some new information as well.
* The party name and number now have their own presentation value
that can be manipulated independently. ISDN supplies the
presentation value for the name and number at different times
with the possibility that they could be different. * The party
name and number now have a valid flag. Before this change the
name or number string could be empty if the presentation were
restricted. Most channel drivers assume that the name or number
is then simply not available instead of indicating that the name
or number was restricted. * The party name now has a character
set value. SIP and Q.SIG have the ability to indicate what
character set a name string is using so it could be presented
properly. * The dialed party now has a numbering plan value that
could be useful to have available. The various channel drivers
will need to be updated to support the new core features as
needed. They have simply been converted to supply current
functionality at this time. The following items of note were
either corrected or enhanced: * The CONNECTEDLINE() and
REDIRECTING() dialplan functions were consolidated into
func_callerid.c to share party id handling code. * CALLERPRES()
is now deprecated because the name and number have their own
presentation values. * Fixed app_alarmreceiver.c
write_metadata(). The workstring[] could contain garbage. It also
can only contain the caller id number so using
ast_callerid_parse() on it is silly. There was also a typo in the
CALLERNAME if test. * Fixed app_rpt.c using ast_callerid_parse()
on the channel's caller id number string. ast_callerid_parse()
alters the given buffer which in this case is the channel's
caller id number string. Then using ast_shrink_phone_number()
could alter it even more. * Fixed caller ID name and number
memory leak in chan_usbradio.c. * Fixed uninitialized char arrays
cid_num[] and cid_name[] in sig_analog.c. * Protected access to a
caller channel with lock in chan_sip.c. * Clarified intent of
code in app_meetme.c sla_ring_station() and dial_trunk(). Also
made save all caller ID data instead of just the name and number
strings. * Simplified cdr.c set_one_cid(). It hand coded the
ast_callerid_merge() function. * Corrected some weirdness with
app_privacy.c's use of caller presentation. Review:
https://reviewboard.asterisk.org/r/702/
2010-07-14 11:51 +0000 [r276268] Leif Madsen <lmadsen@digium.com>
* /, configs/voicemail.conf.sample: Merged revisions 276267 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r276267 | lmadsen | 2010-07-14 06:49:01 -0500 (Wed, 14 Jul 2010)
| 1 line Update documentation for voicemail.conf externpass
option. ........
2010-07-13 22:18 +0000 [r276219] David Vossel <dvossel@digium.com>
* channels/chan_sip.c, channels/sip/include/sip.h: chan_sip: RFC
compliant retransmission timeout Retransmission of packets should
not be based on how many packets were sent, but instead on a
timeout period. Depending on whether or not the packet is for a
INVITE or NON-INVITE transaction, the number of packets sent
during the retransmission timeout period will be different, so
timing out based on the number of packets sent is not accurate.
This patch fixes this by removing the retransmit limit and only
stopping retransmission after a timeout period is reached. By
default this timeout period is 64*(Timer T1) for both INVITE and
non-INVITE transactions. For more information on sip timer values
refer to RFC3261 Appendix A. Review:
https://reviewboard.asterisk.org/r/749/
2010-07-13 21:42 +0000 [r276206] Terry Wilson <twilson@digium.com>
* channels/sip/include/dialog.h, channels/chan_sip.c: Revert early
destruction of RTP sessions Some code improperly assumes that the
sessions are still there, so revert the change until I can find
all of them and fix them.
2010-07-13 19:15 +0000 [r276124-276127] Russell Bryant <russell@digium.com>
* /: Recorded merge of revisions 276126 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r276126 | russell | 2010-07-13 14:14:54 -0500 (Tue, 13 Jul 2010)
| 2 lines Only reset a CDR that exists. ........
* /, main/features.c: Merged revisions 276123 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r276123 | russell | 2010-07-13 14:06:53 -0500 (Tue, 13 Jul 2010)
| 2 lines Use chan->cdr instead of chan_cdr (just like peer->cdr
instead of peer_cdr in the last commit). ........
2010-07-13 19:05 +0000 [r276114-276122] Tilghman Lesher <tlesher@digium.com>
* funcs/func_env.c: Oops, XML documentation fix.
* funcs/func_env.c: It really cannot fail in the places below, but
the stupid compiler doesn't know that.
* funcs/func_env.c: Weird compiler error on Bamboo.
* funcs/func_env.c, CHANGES, tests/test_func_file.c (added): FILE()
now supports line-mode and writing (altering) files. (closes
issue #16461) Reported by: skyman Patches:
20100622__issue16461.diff.txt uploaded by tilghman (license 14)
Tested by: tilghman Review:
https://reviewboard.asterisk.org/r/737/
2010-07-13 17:37 +0000 [r276074] Jeff Peeler <jpeeler@digium.com>
* /, apps/app_meetme.c: Merged revisions 275773 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r275773 | jpeeler | 2010-07-12 15:34:51 -0500 (Mon, 12 Jul 2010)
| 12 lines Make user removals and traversals thread safe in
meetme. Race conditions present in meetme involving the user list
where a lack of locking has the potential for a user to be
removed during a traversal or as in the case of the reporter
after checking if the list is empty could cause a crash. Fixing
this was done by convering the userlist to an ao2 container.
(closes issue #17390) Reported by: Vince Review:
https://reviewboard.asterisk.org/r/746/ ........
2010-07-13 17:11 +0000 [r275998] Terry Wilson <twilson@digium.com>
* channels/sip/include/dialog.h, channels/chan_sip.c: Destroy RTP
fds when we schedule final dialog destruction Since we are only
keeping the dialog around for retransmissions at this point and
there is no possibility that we are still handling RTP, go ahead
and destroy the RTP sessions. Keeping them alive for 32 past when
they are used is unnecessary and can lead to problems with having
too many open file descriptors, etc.
2010-07-13 16:53 +0000 [r275995] Russell Bryant <russell@digium.com>
* /, main/features.c: Merged revisions 275994 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r275994 | russell | 2010-07-13 11:51:18 -0500 (Tue, 13 Jul 2010)
| 14 lines Access peer->cdr directly instead of through a saved
off reference. At this point in the code, it is possible that
peer_cdr may be invalid. Specifically, in the blind transfer
code, CDRs are swapped between channels. So, peer_cdr is no
longer == peer->cdr. The scenario that exposed a crash in this
code was a blind transfer that hit the system call limit, causing
the transferee channel to get destroyed after the transfer
attempt failed. Even if it succeeds and this code doesn't crash,
this code was still trying to reset a CDR on a channel that was
now owned by a different thread, which is a BadThing(tm).
(ABE-2417) ........
2010-07-13 14:48 +0000 [r275910] Tilghman Lesher <tlesher@digium.com>
* contrib/scripts/realtime_pgsql.sql (removed),
contrib/scripts/iax-friends.sql (removed), /,
contrib/realtime/mysql/iaxfriends.sql, contrib/scripts/meetme.sql
(removed), contrib/realtime (added), contrib/realtime/postgresql,
contrib/realtime/postgresql/realtime.sql, contrib/realtime/mysql,
contrib/realtime/oracle, contrib/scripts/sip-friends.sql
(removed), contrib/realtime/mysql/sipfriends.sql,
contrib/realtime/mysql/voicemail.sql, contrib/scripts/vmdb.sql
(removed), contrib/realtime/mysql/meetme.sql,
contrib/realtime/sqlserver: Merged revisions 275909 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r275909 | tilghman | 2010-07-13 09:47:30 -0500 (Tue, 13
Jul 2010) | 2 lines Move SQL scripts into their own
database-specific directories. ........
2010-07-13 11:41 +0000 [r275863] Russell Bryant <russell@digium.com>
* configs/voicemail.conf.sample,
contrib/scripts/voicemailpwcheck.py (added): Add example script
for use with the externpasscheck voicemail.conf option. (closes
issue #17628) Reported by: lmadsen Tested by: russell, lmadsen
Review: https://reviewboard.asterisk.org/r/774/
2010-07-12 23:27 +0000 [r275816] Terry Wilson <twilson@digium.com>
* channels/chan_sip.c: Don't try to ref authpeer when it isn't set
2010-07-12 17:54 +0000 [r275725] Richard Mudgett <rmudgett@digium.com>
* main/channel.c: Add which ITU spec specifies the numbering plan.
2010-07-12 17:21 +0000 [r275682] Jeff Peeler <jpeeler@digium.com>
* main/channel.c, /: Merged revisions 275665 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r275665 | jpeeler | 2010-07-12 11:58:39 -0500 (Mon, 12 Jul 2010)
| 11 lines Change ast_write to not stop generator when called
from ast_prod. For SIP channels configured with the
progressinband option on, the ringback was being immediately
stopped. This problem was due to ast_prod being moved for a
deadlock fix in 259858. Prodding the channel after setting up the
generator triggered the check in ast_write to stop the generator.
The fix here should write the frame the same as was done before
the call to ast_prod was moved. (closes issue #17372) Reported
by: tech_admin ........
2010-07-12 15:37 +0000 [r275626] Leif Madsen <lmadsen@digium.com>
* cdr/cdr_pgsql.c: cdr_pgsql does not detect when a table is found.
This change adds an ERROR message to let you know when a failure
exists to get the columns from the pgsql database, which
typically means that the table does not exist. (closes issue
#17478) Reported by: kobaz Patches: cdr_pgsql.patch uploaded by
kobaz (license 834) Tested by: kobaz, russell, lmadsen
2010-07-12 14:55 +0000 [r275587] Mark Michelson <mmichelson@digium.com>
* main/netsock2.c: Allow netsock2.c to compile on systems that do
not define AI_NUMERICSERV. (closes issue #17617) Reported by:
pprindeville Patches: asterisk-trunk-bugid17617.patch uploaded by
pprindeville (license 347)
2010-07-12 04:16 +0000 [r275551] TransNexus OSP Development <support@transnexus.com>
* configs/osp.conf.sample, apps/app_osplookup.c: Added support for
indirect work mode.
2010-07-10 20:49 +0000 [r275509] Eliel C. Sardanons <eliels@gmail.com>
* apps/app_meetme.c: When creating a conference for a unit test, it
is not mandatory to open a dahdi pseudo channel, so if we fail
doing it, continue creating the conference.
2010-07-10 14:48 +0000 [r275424-275467] Russell Bryant <russell@digium.com>
* CHANGES: Make indentation consistent, move some queue features to
the queue section.
* CREDITS, channels/chan_unistim.c, configs/unistim.conf.sample,
CHANGES: Add support for devices with less than 3 lines on the
LCD. (closes issue #17600) Reported by: minaguib Patches:
ast_unistim_height_v2.patch uploaded by minaguib (license 1078)
Tested by: minaguib
* main/features.c, configs/features.conf.sample: Fix some issues
related to dynamic feature groups in features.conf. The bridge
handling code did not properly consider feature groups when
setting parameters that would affect whether or not a native
bridge would be attempted. If DYNAMIC_FEATURES only include a
feature group, a native bridge would occur that may prevent
features from working. Fix a bug in verbose output that would
show the key mapping as empty if it was using the default mapping
and not a custom mapping in the feature group. Add feature groups
to the output of "features show". Adjust the feature execution
logic to match that of the logic when executing a feature that
was not configured through a feature group. Update
features.conf.sample to show that an '=' is still required if
using the default key mapping from [applicationmap]. Finally,
clean up a little bit of formatting to better coform to coding
guidelines while in the area. (closes issue #17589) Reported by:
lmadsen Patches: issue_17589.rev4.txt uploaded by russell
(license 2) Tested by: russell, lmadsen
2010-07-09 20:58 +0000 [r275385] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Fix error in parsing SIP registry strings
from ASTdb. It was essentially an off-by-one error. The easiest
way to fix this was to use the handy-dandy
AST_NONSTANDARD_RAW_ARGS macro to parse the pieces of the
registration string out. Tested and it works wonderfully.
2010-07-09 20:01 +0000 [r275312] Tilghman Lesher <tlesher@digium.com>
* apps/app_meetme.c, channels/chan_iax2.c: Get more information
about the Bamboo test failures
2010-07-09 19:58 +0000 [r275309-275310] Russell Bryant <russell@digium.com>
* main/features.c: Add missing ao2_iterator_destroy().
* apps/app_voicemail.c: Fix compile error.
2010-07-09 19:46 +0000 [r275308] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Fix port parsing in check_via. If a Via
header contained an IPv6 address, we would not properly parse the
port. We would instead get the information after the first colon
in the address. (closes issue #17614) Reported by: oej Patches:
diff uploaded by sperreault (license 252)
2010-07-09 19:32 +0000 [r275307] Paul Belanger <paul.belanger@polybeacon.com>
* CHANGES, apps/app_voicemail.c: Include rdnis in msgXXXX.txt file.
(closes issue #17566) Reported by: outcast Patches:
voicemail-rdnis.patch uploaded by outcast (license 1071) Tested
by: outcast
2010-07-09 19:29 +0000 [r275294] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Fix an issue where the port for p->ourip was
being set to 0. This should fix all the CDR tests that were not
passing. When they would originate a call, all fields in the
INVITE that contained the source port would have the port set to
0. Most troubling of these was the Contact header. Tests are
passing locally now and should also pass on the bamboo build
agents.
2010-07-09 19:21 +0000 [r275249] Paul Belanger <paul.belanger@polybeacon.com>
* /, channels/chan_sip.c: Merged revisions 275241 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r275241 | pabelanger | 2010-07-09 15:20:00 -0400 (Fri, 09 Jul
2010) | 8 lines Fix logging message for stale nonce. (closes
issue #17582) Reported by: kenner Patches: chan_sip.c.diff
uploaded by kenner (license 1040) Tested by: lmadsen ........
2010-07-09 18:55 +0000 [r275227] Tilghman Lesher <tlesher@digium.com>
* apps/app_meetme.c, channels/chan_iax2.c: Weird, no output and
Bamboo still fails...
2010-07-09 18:24 +0000 [r275186] Matthew Nicholson <mnicholson@digium.com>
* /, main/loader.c: Merged revisions 275182 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r275182 | mnicholson | 2010-07-09 13:23:23 -0500 (Fri, 09 Jul
2010) | 2 lines give a better error message when attempting to
unload a module that is not loaded ........
2010-07-09 18:21 +0000 [r275172] Tilghman Lesher <tlesher@digium.com>
* apps/app_meetme.c, channels/chan_iax2.c: Add some diagnostic
feedback to our data tests
2010-07-09 18:11 +0000 [r275147] Russell Bryant <russell@digium.com>
* configs/features.conf.sample: Move parking lot sample config out
from the middle of dynamic features sample config.
2010-07-09 17:50 +0000 [r275144] Matthew Nicholson <mnicholson@digium.com>
* /, main/loader.c: Merged revisions 275143 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r275143 | mnicholson | 2010-07-09 12:50:05 -0500 (Fri, 09 Jul
2010) | 2 lines don't unload modules that returned
AST_MODULE_LOAD_DECLINE when they were loaded ........
2010-07-09 17:00 +0000 [r275105] Tilghman Lesher <tlesher@digium.com>
* main/netsock2.c, tests/test_substitution.c, tests/test_heap.c,
apps/app_meetme.c, tests/test_gosub.c, funcs/func_strings.c,
tests/test_event.c, channels/sip/reqresp_parser.c,
channels/chan_iax2.c, tests/test_stringfields.c,
tests/test_time.c, tests/test_devicestate.c, tests/test_utils.c,
main/features.c, res/res_agi.c, include/asterisk/netsock2.h,
tests/test_astobj2.c, channels/chan_sip.c,
tests/test_ast_format_str_reduce.c, tests/test_app.c,
funcs/func_math.c, include/asterisk/channel.h,
tests/test_sched.c, tests/test_pbx.c, tests/test_strings.c,
main/data.c, tests/test_skel.c, tests/test_acl.c,
channels/sip/dialplan_functions.c, tests/test_aoc.c, main/test.c,
channels/sip/config_parser.c, res/res_timing_kqueue.c,
apps/app_voicemail.c: Kill some startup warnings and errors and
make some messages more helpful in tracking down the source.
2010-07-09 16:39 +0000 [r275104] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Return logic of sip_debug_test_addr() to its
original functionality.
2010-07-09 16:05 +0000 [r275028] Matthew Nicholson <mnicholson@digium.com>
* apps/app_dial.c, /: Merged revisions 275027 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r275027 | mnicholson | 2010-07-09 11:04:21 -0500 (Fri, 09 Jul
2010) | 8 lines Clear the AST_CDR_FLAG_DIALED flag for channels
going into the pbx via the G option in app_dial (closes issue
#17592) Reported by: jamicque Patches: G-flag-cdr-fix1.diff
uploaded by mnicholson (license 96) Tested by: jamicque,
mnicholson ........
2010-07-09 15:35 +0000 [r275022] Russell Bryant <russell@digium.com>
* include/asterisk/test.h, /, main/test.c: Merged revisions 275021
via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r275021 | russell | 2010-07-09 10:33:08 -0500 (Fri, 09 Jul 2010)
| 4 lines Document that a leading and trailing slash is expected
for test categories. Also, emit a warning if a test is registered
without one of these. ........
2010-07-09 14:27 +0000 [r274984] Mark Michelson <mmichelson@digium.com>
* channels/sip/reqresp_parser.c: Fix sip_uri_parse test comparison.
Part of the change with the IPv6 changes is to treat a host:port
as a single 'domain' entity. This test was not updated to have
the correct expectation after calling parse_uri().
2010-07-09 13:30 +0000 [r274909-274947] <simon.perreault@viagenie.ca>
* channels/chan_sip.c: Copy the address into the peer structure
after we set the default port
* main/netsock2.c: Sadly we can't dereference a pointer cast and
use it as an lvalue without getting this warning (at least with
gcc 4.4.4): netsock2.c:492: warning: dereferencing pointer
‘({anonymous})’ does break strict-aliasing rules So we're back to
using memcpy()...
2010-07-09 12:48 +0000 [r274907] Russell Bryant <russell@digium.com>
* include/asterisk/indications.h: Extend length limit on country
name in indications.conf.
2010-07-09 11:06 +0000 [r274866] Olle Johansson <oej@edvina.net>
* configs/cdr.conf.sample, cdr/cdr_csv.c: Make it possible to
disable individual cdr files per accountcode in cdr_csv Review:
https://reviewboard.asterisk.org/r/678/
2010-07-08 23:46 +0000 [r274827-274828] Richard Mudgett <rmudgett@digium.com>
* channels/chan_jingle.c, channels/chan_h323.c,
channels/chan_gtalk.c: Fix calls of ast_sockaddr_from_sin() from
IPv6 integration.
* addons/chan_ooh323.c: Fix compile of chan_ooh323.c from IPv6
integration.
2010-07-08 22:16 +0000 [r274783-274786] Mark Michelson <mmichelson@digium.com>
* /: And the automerge property.
* /: Delete properties I merged during v6-new merge.
* channels/chan_unistim.c, include/asterisk/acl.h, main/netsock2.c
(added), channels/sip/include/dialog.h,
channels/chan_multicast_rtp.c, addons/chan_ooh323.c,
main/rtp_engine.c, /, channels/sip/reqresp_parser.c,
include/asterisk/tcptls.h, channels/chan_gtalk.c,
channels/chan_iax2.c, main/config.c, res/res_rtp_multicast.c,
main/manager.c, channels/chan_skinny.c,
channels/sip/include/globals.h, main/http.c, main/app.c,
include/asterisk/netsock2.h (added), apps/app_externalivr.c,
configs/sip.conf.sample, include/asterisk/rtp_engine.h,
channels/sip/include/sip.h, channels/chan_mgcp.c,
channels/sip/include/reqresp_parser.h, res/res_rtp_asterisk.c,
main/dnsmgr.c, channels/chan_sip.c, include/asterisk/config.h,
main/acl.c, CHANGES, channels/chan_jingle.c, main/tcptls.c,
channels/sip/dialplan_functions.c, channels/chan_h323.c,
include/asterisk/dnsmgr.h: Add IPv6 to Asterisk. This adds a
generic API for accommodating IPv6 and IPv4 addresses within
Asterisk. While many files have been updated to make use of the
API, chan_sip and the RTP code are the files which actually
support IPv6 addresses at the time of this commit. The way has
been paved for easier upgrading for other files in the near
future, though. Big thanks go to Simon Perrault, Marc Blanchet,
and Jean-Philippe Dionne for their hard work on this. (closes
issue #17565) Reported by: russell Patches:
asteriskv6-test-report.pdf uploaded by russell (license 2)
Review: https://reviewboard.asterisk.org/r/743
2010-07-08 22:05 +0000 [r274773-274782] Richard Mudgett <rmudgett@digium.com>
* main/channel.c: Generate a correct AstData string for
ast_callerid.cid_ton
* main/channel.c: Fix trunk compile.
2010-07-08 14:48 +0000 [r274727] Eliel C. Sardanons <eliels@gmail.com>
* main/pbx.c, channels/chan_sip.c, apps/app_meetme.c,
include/asterisk/indications.h, channels/chan_agent.c,
include/asterisk/channel.h, include/asterisk/cdr.h,
include/asterisk/data.h, channels/chan_iax2.c, apps/app_queue.c,
main/indications.c, main/channel.c, main/cdr.c,
channels/chan_dahdi.c, main/data.c, res/res_odbc.c,
apps/app_voicemail.c: Implement AstData API data providers as
part of the GSOC 2010 project, midterm evaluation. Review:
https://reviewboard.asterisk.org/r/757/
2010-07-07 20:09 +0000 [r274686] David Vossel <dvossel@digium.com>
* channels/chan_sip.c: Fixes some ref count issues introduced by
r274539
2010-07-07 18:32 +0000 [r274595-274639] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c: Add missing conditional around chan_dahdi
mfcr2_skip_category config parameter.
* channels/chan_dahdi.c, /: Merged revisions 274579 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r274579 | rmudgett | 2010-07-07 13:12:41 -0500 (Wed, 07
Jul 2010) | 1 line Close the DAHDI FD on error when processing
chan_dahdi toneduration config parameter. ........
2010-07-07 16:40 +0000 [r274540] Matthew Nicholson <mnicholson@digium.com>
* res/res_fax.c: Set proper FAXOPT(status), FAXOPT(statusstr), and
FAXOPT(error) values where possible. Previously some failure
cases did not result in proper FAXOPT values. FAX-203
2010-07-07 16:21 +0000 [r274539] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Use the relatedpeer field of a sip_pvt
during INVITE processing. Review:
https://reviewboard.asterisk.org/r/629
2010-07-07 07:07 +0000 [r274492] TransNexus OSP Development <support@transnexus.com>
* configs/osp.conf.sample, doc/osp.txt: Changed OSP TCP port from
1080 to 5045.
2010-07-07 06:32 +0000 [r274418-274491] Tilghman Lesher <tlesher@digium.com>
* CHANGES, apps/app_voicemail.c: Also run the externnotify script
when the pollmailboxes thread notices a change.
* /, configs/say.conf.sample: Merged revisions 274417 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r274417 | tilghman | 2010-07-07 01:13:54 -0500 (Wed, 07
Jul 2010) | 8 lines Correct how 100, 200, 300, etc. is said. Also
add the crazy British numbers. (closes issue #16102) Reported by:
Delvar Patches: say.conf.fix.patch uploaded by Delvar (license
908) (plus a few additional fixes and simplifications by me)
........
2010-07-06 22:23 +0000 [r274316] Jeff Peeler <jpeeler@digium.com>
* /, configs/sip.conf.sample: Merged revisions 274283 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r274283 | jpeeler | 2010-07-06 17:15:21 -0500 (Tue, 06
Jul 2010) | 7 lines Correct sip.conf.sample comments for
prematuremedia option. (closes issue #17513) Reported by: festr
Patches: patch uploaded by festr (license 443) ........
2010-07-06 22:15 +0000 [r274284] Terry Wilson <twilson@digium.com>
* /, channels/chan_sip.c, UPGRADE.txt: Merged revisions 274280 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r274280 | twilson | 2010-07-06 17:08:20 -0500 (Tue, 06 Jul 2010)
| 9 lines Add option to not do a call forward on 482 Loop
Detected Asterisk has always set up a forwarded call when
receiving a 482 Loop Detected. This prevents handling the call
failure by just continuing on in the dialplan. Since this would
be a change in behavior, the new option to disable this behavior
is forwardloopdetected which defaults to 'yes'. Review:
https://reviewboard.asterisk.org/r/764/ ........ (no option for
trunk, just changing the behavior)
2010-07-06 22:09 +0000 [r274281] Tilghman Lesher <tlesher@digium.com>
* channels/chan_dahdi.c: Status shows all non-CRC4 lines as
"yellow", even if "yellow" was not in the bitfield.
2010-07-06 19:53 +0000 [r274243] Matthew Nicholson <mnicholson@digium.com>
* res/res_fax.c: Properly detect and report invalid maxrate and
maxrate values in the FAXOPT dialplan function. Also make
fax_rate_str_to_int() return an unsigned int and return 0 instead
of -1 in the event of an error. FAX-202
2010-07-06 14:31 +0000 [r274164] Mark Michelson <mmichelson@digium.com>
* res/res_rtp_asterisk.c, /: Merged revisions 274157 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r274157 | mmichelson | 2010-07-06 09:29:23 -0500 (Tue,
06 Jul 2010) | 16 lines Fix problem with RFC 2833 DTMF not being
accepted. A recent check was added to ensure that we did not
erroneously detect duplicate DTMF when we received packets out of
order. The problem was that the check did not account for the
fact that the seqno of an RTP stream will roll over back to 0
after hitting 65535. Now, we have a secondary check that will
ensure that the seqno rolling over will not cause us to stop
accepting DTMF. (closes issue #17571) Reported by: mdeneen
Patches: rtp_seqno_rollover.patch uploaded by mmichelson (license
60) Tested by: richardf, maxochoa, JJCinAZ ........
2010-07-06 06:01 +0000 [r274053] Tilghman Lesher <tlesher@digium.com>
* main/pbx.c: Uh, yeah.
2010-07-05 13:53 +0000 [r273886] Paul Belanger <paul.belanger@polybeacon.com>
* /, main/config.c: Merged revisions 273884 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r273884 | pabelanger | 2010-07-05 09:51:29 -0400 (Mon, 05 Jul
2010) | 8 lines Remove extra line breaks from 'core show config
mappings' (closes issue #17583) Reported by: pabelanger Patches:
issue17583.patch uploaded by pabelanger (license 224) Tested by:
lmadsen ........
2010-07-03 02:36 +0000 [r273714-273830] Tilghman Lesher <tlesher@digium.com>
* channels/chan_local.c, /, channels/chan_agent.c,
channels/chan_h323.c, include/asterisk/lock.h: Merged revisions
273793 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r273793 | tilghman | 2010-07-02 16:36:39 -0500 (Fri, 02 Jul 2010)
| 9 lines Have the DEADLOCK_AVOIDANCE macro warn when an unlock
fails, to help catch potentially large software bugs. (closes
issue #17407) Reported by: pdf Patches:
20100527__issue17407.diff.txt uploaded by tilghman (license 14)
Review: https://reviewboard.asterisk.org/r/751/ ........
* main/autoservice.c, /: Merged revisions 273717 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r273717 | tilghman | 2010-07-02 12:09:47 -0500 (Fri, 02 Jul 2010)
| 8 lines Autoservice loop optimization causes a busy loop, when
channels are serviced while in hangup. (closes issue #17564)
Reported by: ramonpeek Patches: 20100630__issue17564.diff.txt
uploaded by tilghman (license 14) Tested by: ramonpeek ........
* apps/app_queue.c: The switch fallthrough could create some
errorneous situations, so best to force directly to the default
case.
2010-07-02 15:57 +0000 [r273641] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
* channels/chan_dahdi.c, channels/chan_misdn.c,
channels/chan_sip.c, main/say.c, main/fixedjitterbuf.c,
res/res_agi.c, channels/chan_h323.c, main/utils.c,
channels/chan_iax2.c, addons/chan_mobile.c, apps/app_rpt.c,
channels/chan_mgcp.c, main/xmldoc.c, apps/app_voicemail.c,
apps/app_while.c: Fix various typos reported by Lintian (Also fix
the typos in the comments)
2010-07-01 22:16 +0000 [r273566] Russell Bryant <russell@digium.com>
* /, main/datastore.c: Merged revisions 273565 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r273565 | russell | 2010-07-01 17:09:19 -0500 (Thu, 01 Jul 2010)
| 7 lines Don't return a partially initialized datastore. If
memory allocation fails in ast_strdup(), don't return a partially
initialized datastore. Bad things may happen. (related to
ABE-2415) ........
2010-07-01 20:28 +0000 [r273522] Jeff Peeler <jpeeler@digium.com>
* /, apps/app_meetme.c: Merged revisions 273474 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r273474 | jpeeler | 2010-07-01 15:19:16 -0500 (Thu, 01 Jul 2010)
| 14 lines Allow admin user to join conference without using
admin mode and no user pin. Configuring the conference in
meetme.conf like the following: conf => 2345,,6666 did not prompt
for pin when used without admin mode. This meant that the
conference could not be joined as an admin even if the user knew
the correct pin. The original bug report was submitted claiming
that the blank user pin should deny entry into the conference. I
think a better way to handle this would be with a feature
enhancement that used the following syntax: conf => 2345,X,6666 -
where X denotes no acceptable pin allowed (closes issue #15704)
Reported by: modelnine ........
2010-07-01 19:34 +0000 [r273464] Matthew Nicholson <mnicholson@digium.com>
* res/res_fax.c: Properly handle failures of fax->start_session()
FAX-177
2010-07-01 16:40 +0000 [r273427] David Vossel <dvossel@digium.com>
* channels/chan_sip.c, channels/sip/include/sip.h: correct handling
of get_destination return values A failure when calling the
get_destination can mean multiple things. If the extension is not
found, a 404 error is appropriate, but if the URI scheme is
incorrect, a 404 is not approperiate. This patch adds the
get_destination_result enum to differentiate between these and
other failure types. The only logical difference in this patch is
that we now send a "416 Unsupported URI scheme" response instead
of a "404" when the scheme is not recognized. This indicates to
the initiator of the INVITE to retry the request with a correct
URI.
2010-07-01 15:12 +0000 [r273355] Jeff Peeler <jpeeler@digium.com>
* /, apps/app_meetme.c: Merged revisions 273354 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r273354 | jpeeler | 2010-07-01 10:05:43 -0500 (Thu, 01 Jul 2010)
| 12 lines Ensure channel placed in meetme in ringing state is
properly hung up. An outgoing channel placed in meetme while
still ringing which was then hung up would not exit meetme and
the channel was not properly destroyed. Specifically checking for
this scenario by looking at the appropriate control frames
resolves the issue. (closes issue #15871) Reported by: Ivan
Patches: meetme_congestion_trunk_v2.patch uploaded by Ivan
(license 229) ........
2010-07-01 14:37 +0000 [r273270-273352] Matthew Nicholson <mnicholson@digium.com>
* main/manager.c: Fixed whitespace problems
* main/manager.c: Altered my comment about TCP_NODELAY
* addons/chan_mobile.c: Don't free written frames in chan_mobile's
mbl_write() function. (closes issue #16430) Reported by: azbest
Tested by: azbest
* main/manager.c: Set TCP_NODELAY on manager TCP sockets to prevent
delays on outgoing packets. This regression was introduced in
r48338. AST-359
2010-06-30 17:28 +0000 [r273233] Paul Belanger <paul.belanger@polybeacon.com>
* res/res_rtp_asterisk.c: Fix rt(c)p set debug ip taking wrong
argument Also clean up some coding errors. (closes issue #17469)
Reported by: wdoekes Patches: astsvn-rtp-set-debug-ip.patch
uploaded by wdoekes (license 717) Tested by: wdoekes, pabelanger
2010-06-30 17:17 +0000 [r273197-273198] Richard Mudgett <rmudgett@digium.com>
* include/asterisk/config.h: Remove unnecessary if test in
CV_DSTR()
* include/asterisk/config.h: Misc doxygen cleanup in config.h
2010-06-30 01:07 +0000 [r273054-273144] Tilghman Lesher <tlesher@digium.com>
* main/manager.c: Permission checking for the system application is
backwards. (closes issue #17550) Reported by: kenner Patches:
manager.c.diff uploaded by kenner (license 1040) Tested by:
kenner
* main/config.c: Don't attempt to proceed if our internal parser
indicates an invalid file. (closes issue #17560) Reported by:
Nick_Lewis
* /, channels/chan_sip.c: Merged revisions 273060 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r273060 | tilghman | 2010-06-29 18:15:28 -0500 (Tue, 29 Jun 2010)
| 10 lines Allow the "useragent" value to be restored into memory
from the realtime backend. This value is purely informational. It
does not alter configuration at all. (closes issue #16029)
Reported by: Guggemand Patches: realtime-useragent.patch uploaded
by Guggemand (license 897) Tested by: Guggemand ........
* /: Recorded merge of revisions 273057 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r273057 | tilghman | 2010-06-29 17:58:58 -0500 (Tue, 29 Jun 2010)
| 4 lines _Really_ skip the channel... don't just retry for
another 200 cycles. (Closes issue SWP-1652, ABE-2240) ........
* configure, include/asterisk/autoconfig.h.in, configure.ac:
Exclude libical for insufficient versions.
* main/pbx.c: Send DialPlanComplete as a response, not as a
separate event. Otherwise, it goes to all manager sessions and
may exclude the current session, if the Events mask excludes it.
(closes issue #17504) Reported by: rrb3942 Patches:
showdialplan_patch.diff uploaded by rrb3942 (license 1003) Tested
by: rrb3942
2010-06-29 20:44 +0000 [r272981] David Vossel <dvossel@digium.com>
* channels/chan_sip.c: send a 400 Bad Request on malformed sip
request RFC 2361 section 24.4.1 send a 400 Bad Request if the
request can not be understood due to malformed syntax. Currently
we simply ignore a packet with a missing callid, to, from, or via
header. Instead of ignoring we now send the 400 Bad request.
2010-06-28 21:50 +0000 [r272923-272926] Tilghman Lesher <tlesher@digium.com>
* /, main/asterisk.c: Merged revisions 272925 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r272925 | tilghman | 2010-06-28 16:50:02 -0500 (Mon, 28 Jun 2010)
| 8 lines Don't change ownership/group/permissions on run
directory, if it already exists. (closes issue #17076) Reported
by: stuarth Patches: 20100324__issue17076.diff.txt uploaded by
tilghman (license 14) Tested by: stuarth ........
* /, main/config.c: Merged revisions 272921-272922 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r272921 | tilghman | 2010-06-28 16:29:27 -0500 (Mon, 28
Jun 2010) | 8 lines Change the way that we read include files, to
accommodate for changes in GCC 4.4. (closes issue #17472)
Reported by: seandarcy Patches: config2.patch uploaded by nivan
(license 1066) Tested by: nivan ........ r272922 | tilghman |
2010-06-28 16:38:49 -0500 (Mon, 28 Jun 2010) | 2 lines Also trim
trailing blanks on #includes ........
2010-06-28 18:38 +0000 [r272880] David Vossel <dvossel@digium.com>
* channels/chan_sip.c, channels/sip/reqresp_parser.c,
channels/sip/include/sip.h,
channels/sip/include/reqresp_parser.h: rfc compliant sip option
parsing + new unit test RFC 3261 section 8.2.2.3 states that if
any unsupported options are found in the Require header field, a
"420 (Bad Extension)" response should be sent with an Unsupported
header field containing only the unsupported options. This is not
currently being done correctly. Right now, if Asterisk detects
any unsupported sip options in a Require header the entire list
of options are returned in the Unsupported header even if some of
those options are in fact supported. This patch fixes that by
building an unsupported options character buffer when parsing the
options that can be sent with the 420 response. A unit test
verifying this functionality has been created. Some code
refactoring was required. Review:
https://reviewboard.asterisk.org/r/680/
2010-06-28 17:33 +0000 [r272805] Mark Michelson <mmichelson@digium.com>
* /, channels/chan_sip.c: Merged revisions 272804 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r272804 | mmichelson | 2010-06-28 12:31:40 -0500 (Mon, 28 Jun
2010) | 5 lines Decode URI in contact header of 302 response.
ABE-2352 ........
2010-06-28 15:33 +0000 [r272684] Russell Bryant <russell@digium.com>
* doc/tex/chan-mobile.tex (added), doc/tex/celdriver.tex,
doc/tex/chan_mobile.tex (removed), doc/tex/cdrdriver.tex,
doc/tex/asterisk.tex, doc/tex/cel-doc.tex: Use the underscore
package so that underscores do not need to be escaped.
2010-06-28 14:55 +0000 [r272652] David Vossel <dvossel@digium.com>
* channels/chan_sip.c: code guidelines cleanup for retrans_pkt()
function I am doing work in this function. I noticed a large
number of coding guidline fixes that needed to be made. Rather
than have those changes distract from my functional changes I
decided to separate these into a separate patch.
2010-06-25 20:18 +0000 [r272568] Tilghman Lesher <tlesher@digium.com>
* /, doc/voicemail_odbc_postgresql.txt: Merged revisions 272562 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r272562 | tilghman | 2010-06-25 15:17:37 -0500 (Fri, 25 Jun 2010)
| 5 lines Make the structure of the table specified before match
the queries and results. (closes issue #17557) Reported by: cmaj
........
2010-06-25 19:42 +0000 [r272558] Matthew Nicholson <mnicholson@digium.com>
* res/res_fax.c, include/asterisk/res_fax.h: Implemement support
for handling multiple documents when sending.
2010-06-25 19:39 +0000 [r272557] David Vossel <dvossel@digium.com>
* channels/chan_sip.c: chan_sip: more accurate retransmissions
RFC3261 states that Timer A should start at 500ms (T1) by
default. In chan_sip this value initially started at 1000ms and I
changed it to 500ms recently. After doing that I noticed in my
packet captures that it still occasionally retransmitted starting
at 1000ms instead of 500ms like I told it to. This occurs because
the scheduler runs in the do_monitor thread. If a new
retransmission is added while the do_monitor thread is sleeping
then it may not detect that retransmission for nearly 1000ms. To
fix this I just poke the do_monitor thread to wake up when a new
packet is sent reliably requiring retransmits. The thread then
detects the new scheduler entry and adjusts its sleep time to
account for it. Review: https://reviewboard.asterisk.org/r/747
2010-06-25 19:17 +0000 [r272533] Tilghman Lesher <tlesher@digium.com>
* sounds/Makefile: Symlink sounds files, to save disk space, when
multiple tarballs/checkouts are on the same system.
2010-06-24 22:11 +0000 [r272447] Richard Mudgett <rmudgett@digium.com>
* /, channels/sig_pri.c: Merged revisions 272446 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r272446 | rmudgett | 2010-06-24 16:58:49 -0500 (Thu, 24 Jun 2010)
| 10 lines ss_thread calls pri_grab without lock during overlap
dial Recent changes to chan_dahdi with relation to overlap
dialing call pri_grab without first obtaining a lock. (closes
issue #17414) Reported by: pdf Patches: bug17414.patch uploaded
by jpeeler (license 325) ........
2010-06-23 23:09 +0000 [r272370] Russell Bryant <russell@digium.com>
* channels/chan_iax2.c: Resolve some errors produced during module
unload of chan_iax2. The external test suite stops Asterisk using
the "core stop gracefully" command. The logs from the tests show
that there are a number of problems with Asterisk trying to
cleanly shut down. This patch addresses the following type of
error that comes from chan_iax2: [Jun 22 16:58:11] ERROR[29884]:
lock.c:129 __ast_pthread_mutex_destroy: chan_iax2.c line 11371
(iax2_process_thread_cleanup): Error destroying mutex
&thread->lock: Device or resource busy For an example in the
context of a build, see:
http://bamboo.asterisk.org/browse/AST-TRUNK-739/log The primary
purpose of this patch is to change the thread pool shutdown
procedure to be more explicit to ensure that the thread exits
from a point where it is not holding a lock. While testing that,
I encountered various crashes due to the order of operations in
unload_module() being problematic. I reordered some things there,
as well. Review: https://reviewboard.asterisk.org/r/736/
2010-06-23 22:36 +0000 [r272368] Matthew Nicholson <mnicholson@digium.com>
* /, apps/app_queue.c: Merged revisions 272367 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 This version
of the patch only adds AgentComplete for attended transfers. It
was already present for blind transfers. ........ r272367 |
mnicholson | 2010-06-23 17:33:51 -0500 (Wed, 23 Jun 2010) | 8
lines Send AgentComplete manager events in the event of blind and
attended transfers. (closes issue #16819) Reported by: elbriga
Patches: app_queue.diff uploaded by elbriga (license 482)
........
2010-06-23 21:53 +0000 [r272260-272332] Tilghman Lesher <tlesher@digium.com>
* res/res_musiconhold.c: If there is realtime configuration, it
does not get re-read on reload unless the config file also
changes. (closes issue #16982) Reported by: dmitri Patches:
res_musiconhold.patch uploaded by dmitri (license 1001) Tested
by: atis
* res/ael/ael.tab.c, res/ael/ael.y, res/ael/ael_lex.c,
res/ael/ael.flex: Ensure a NULL file while debugging cannot crash
AEL. (closes issue #17215) Reported by: vazir Patches:
20100518__issue17215.diff.txt uploaded by tilghman (license 14)
Tested by: tilghman
2010-06-23 21:06 +0000 [r272257-272259] Paul Belanger <paul.belanger@polybeacon.com>
* apps/app_meetme.c: Fix previous merge. ast_test_flag !=
ast_test_flag64
* /, apps/app_meetme.c: Merged revisions 272255 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r272255 | pabelanger | 2010-06-23 16:57:01 -0400 (Wed, 23 Jun
2010) | 12 lines First caller into a dynamic conference now enter
pin once. If MeetMe is configured to use dynamic conference
numbers, then the first caller (which creates the conference) had
to enter the PIN number twice. (closes issue #15878) Reported by:
shawkris Patches: issue15878.patch uploaded by pabelanger
(license 224) Tested by: pabelanger ........
2010-06-23 20:59 +0000 [r272254-272256] Terry Wilson <twilson@digium.com>
* configure, include/asterisk/autoconfig.h.in: Update configure
when changing autconf m4 files...
* autoconf/ast_ext_tool_check.m4: Honor the --with-${library}=path
for AST_EXT_TOOL_CHECK (closes issue #16991) Reported by:
pprindeville Patches: with_netsnmp.patch.txt uploaded by twilson
(license 396) Tested by: twilson Review:
https://reviewboard.asterisk.org/r/739/
2010-06-23 20:35 +0000 [r272243-272252] Paul Belanger <paul.belanger@polybeacon.com>
* main/manager.c: Correct manager variable 'EventList' case.
(closes issue #17520) Reported by: kobaz Patches: manager.patch
uploaded by kobaz (license 834) Tested by: lmadsen
* configs/say.conf.sample: Add localization support for Spanish
(closes issue #17548) Reported by: cjacobsen Patches:
say.conf.sample.diff uploaded by cjacobsen (license 1029)
2010-06-23 19:59 +0000 [r272218] Tim Ringenbach <tim.ringenbach@gmail.com>
* channels/chan_local.c: Add new AMI command LocalOptimizeAway.
This command lets you request a "/n" local channel optimize
itself out of the way anyway. Review:
https://reviewboard.asterisk.org/r/732/
2010-06-23 18:45 +0000 [r272148-272150] Tilghman Lesher <tlesher@digium.com>
* channels/chan_mgcp.c: D'oh! Defaultenabled FTL.
* /: Recorded merge of revisions 272147 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r272147 | tilghman | 2010-06-23 13:40:28 -0500 (Wed, 23 Jun 2010)
| 5 lines Backport part of revision 136715 to fix callerid in
voicemail text files (IMAP only). (closes issue #16945) Reported
by: mneuhauser ........
2010-06-23 18:39 +0000 [r272146] Terry Wilson <twilson@digium.com>
* apps/app_meetme.c: Don't start the sla thread unless we realy
need it
2010-06-23 18:25 +0000 [r272145] Tilghman Lesher <tlesher@digium.com>
* channels/chan_mgcp.c: Load all lines from realtime, not just the
first one. (closes issue #17144) Reported by: nahuelgreco
Patches: 20100513__issue17144__trunk.diff.txt uploaded by
tilghman (license 14) Tested by: tilghman
2010-06-23 17:21 +0000 [r272109] Terry Wilson <twilson@digium.com>
* apps/app_meetme.c: Make sure reload updates SLA config Even if
there are no stations or trunks defined, we need to start the sla
thread to make sure we get the reload event. Also, when doing a
reload we need to remove the existing trunks and stations or they
end up hanging around. (closes issue #16818) Reported by: mbonin
Patches: sla_reload.patch uploaded by twilson (license 396)
Tested by: twilson
2010-06-23 17:08 +0000 [r272090] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Add extra protection for reinvite glare
scenario. Testing proved that if Asterisk sent a connected line
reinvite, and the endpoint to which the reinvite were being sent
sent a reinvite, Asterisk would not properly respond with a 491
response. The reason is that on connected line reinvites, we set
the dialog's invitestate to INV_CALLING to prevent Asterisk from
sending a rapid flurry of connected line reinvites. For other
reinvites we do not do this. Because of the current invitestate,
when Asterisk received the reinvite, we interpreted this as a
spiraled INVITE, and thus did not behave properly. The fix for
this is to not enter the loop detection or spiral logic in
handle_request_invite if the channel state is currently up. This
way, no mid-call reinvites will be misinterpreted, no matter what
the nature of the reinvite may have been.
2010-06-22 23:20 +0000 [r272052] Russell Bryant <russell@digium.com>
* channels/chan_dahdi.c: Don't try to lock/unlock an uninitialized
lock on a dahdi_pri. This small changes prevents
destroy_all_channels() from accessing a lock on an unused
dahdi_pri struct, resolving a ton of ERRORs that get spewed out
when shutting Asterisk down gracefully.
2010-06-22 22:11 +0000 [r271905-272014] David Vossel <dvossel@digium.com>
* pbx/pbx_config.c: fixes issue with 'dialplan remove extension
blah' segfaulting with tab completion (closes issue #17440)
Reported by: kobaz
* channels/chan_sip.c: ignore CANCEL request after having already
received final response to INVITE RFC 3261 section 9 states that
a CANCEL has no effect on a request to a UAS that has already
given a final response. This patch checks to make sure there is a
pending invite before allowing a CANCEL request to be processed,
otherwise it responds to the CANCEL with a "481 Call/Transaction
Does Not Exist". Review: https://reviewboard.asterisk.org/r/697/
* main/manager.c: minor fixes for white/black event filters This
fixes a ref count leak in event filters and checks for a filter
container allocation failure during session creation.
2010-06-22 17:35 +0000 [r271903] Matthew Nicholson <mnicholson@digium.com>
* /, channels/chan_sip.c: Merged revisions 271902 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r271902 | mnicholson | 2010-06-22 12:31:57 -0500 (Tue, 22 Jun
2010) | 8 lines Decrease the module ref count in sip_hangup when
SIP_DEFER_BYE_ON_TRANSFER is set. This is necessary to keep the
ref count correct. (closes issue #16815) Reported by: rain
Patches: chan_sip-unref-fix.diff uploaded by rain (license 327)
(modified) Tested by: rain ........
2010-06-22 16:29 +0000 [r271868] Jeff Peeler <jpeeler@digium.com>
* main/manager.c, configs/manager.conf.sample, CHANGES: Add regular
expression filtering for manager events. This patch as documented
in the sample config allows one to optionally apply white, black,
or both types of filtering to manager events. The new
'eventfilter' option is set per user. (closes issue #14861)
Reported by: fnordian Patches: eventfilter3.patch uploaded by
fnordian (license 110), modified by me Review:
https://reviewboard.asterisk.org/r/673/
2010-06-22 16:28 +0000 [r271833-271867] Russell Bryant <russell@digium.com>
* res/ais/clm.c, res/ais/evt.c: Resolve some errors that occur on a
graceful shutdown. Don't Finalize() if Initialize() did not
succeed. This resulted in an error about trying to Finalize() an
invalid handle. Also trim some trailing whitespace while in the
area.
* res/res_fax.c: Change the method of retrieving the Asterisk
version string. Using this method makes it so res_fax doesn't
have to be rebuilt on every svn update.
2010-06-22 15:46 +0000 [r271831] David Vossel <dvossel@digium.com>
* main/features.c: fixes attended transfer behavior when both
transferee and transferer hung up If both the transferer and
transferee of a attended transfer hangup before the new channel
picks up, the new channel should be hung up as well as it has no
endpoint to talk to. This mirrors the expected behavior used in
1.4. (closes issue #17444) Reported by: corruptor
2010-06-22 15:08 +0000 [r271690-271764] Matthew Nicholson <mnicholson@digium.com>
* CHANGES: Updated the CHANGES file documenting the addition of a
configurable port in the dundi config file.
* configs/dundi.conf.sample, /, pbx/pbx_dundi.c: Merged revisions
271761 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r271761 | mnicholson | 2010-06-22 09:49:36 -0500 (Tue, 22 Jun
2010) | 9 lines Allow users to specify a port for dundi peers.
(closes issue #17056) Reported by: klaus3000 Patches:
dundi-peerport-patch-trunk.txt uploaded by klaus3000 (license 65)
Tested by: klaus3000 ........
* /, channels/chan_sip.c, include/asterisk/strings.h,
channels/sip/include/sip.h: Merged revisions 271689 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r271689 | mnicholson | 2010-06-22 07:52:27 -0500 (Tue,
22 Jun 2010) | 8 lines Modify chan_sip's packet generation api to
automatically calculate the Content-Length. This is done by
storing packet content in a buffer until it is actually time to
send the packet, at which time the size of the packet is
calculated. This change was made to ensure that the
Content-Length is always correct. (closes issue #17326) Reported
by: kenner Tested by: mnicholson, kenner Review:
https://reviewboard.asterisk.org/r/693/ ........ This change also
adds an ast_str_copy_string() function (similar to
ast_copy_string), that copies one ast_str into another, properly
handling embedded nulls.
2010-06-21 22:41 +0000 [r271657] Tilghman Lesher <tlesher@digium.com>
* build_tools/menuselect-deps.in, configure, configure.ac,
res/res_timing_kqueue.c: Conflict kqueue on OS X, since it
doesn't work there yet, anyway.
2010-06-21 21:58 +0000 [r271625] David Vossel <dvossel@digium.com>
* codecs/codec_speex.c, codecs/ex_speex.h,
contrib/editors/asterisk.vim: add speex 16khz sample frame so
codec cost can be calculated (closes issue #17534) Reported by:
fabled Patches: speex-wb-sample.diff uploaded by fabled (license
448)
2010-06-21 20:46 +0000 [r271554] Jeff Peeler <jpeeler@digium.com>
* res/ael/pval.c, /: Merged revisions 271552 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r271552 | jpeeler | 2010-06-21 15:37:47 -0500 (Mon, 21 Jun 2010)
| 7 lines Do not use sizeof to calculate size of a heap allocated
character array. Change left out from 271399. (closes issue
#16053) Reported by: diLLec ........
2010-06-21 20:46 +0000 [r271551-271553] David Vossel <dvossel@digium.com>
* channels/chan_sip.c, channels/sip/reqresp_parser.c: fixes crash
when From header URI is missing "sip:" (closes issue #17437)
Reported by: klaus3000 Patches: sip_crash uploaded by dvossel
(license 671) Tested by: klaus3000
* res/res_rtp_asterisk.c: fixes logic error introduced by slin16
sip support
2010-06-21 05:10 +0000 [r271520] Tilghman Lesher <tlesher@digium.com>
* apps/app_saycounted.c (added), CHANGES: Add new application for
declining counting words in multiple languages. (closes issue
#16869) Reported by: chappell Patches: app_say_counted-20100317.c
uploaded by chappell (license 8) Tested by: chappell
2010-06-18 21:32 +0000 [r271483] Jeff Peeler <jpeeler@digium.com>
* res/ael/pval.c, /, include/asterisk/pval.h, pbx/pbx_ael.c: Merged
revisions 271399 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r271399 | jpeeler | 2010-06-18 14:28:24 -0500 (Fri, 18 Jun 2010)
| 11 lines Fix crash when parsing some heavily nested statements
in AEL on reload. Due to the recursion used when compiling AEL in
gen_prios, all the stack space was being consumed when parsing
some AEL that contained nesting 13 levels deep. Changing a few
large buffers to be heap allocated fixed the crash, although I
did not test how many more levels can now be safely used. (closes
issue #16053) Reported by: diLLec Tested by: jpeeler ........
2010-06-18 18:59 +0000 [r271341] David Vossel <dvossel@digium.com>
* main/file.c: file.c was truncating audio file formats to the
lower 32bits.
2010-06-18 18:36 +0000 [r271336] Jeff Peeler <jpeeler@digium.com>
* /: Recorded merge of revisions 271335 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r271335 | jpeeler | 2010-06-18 13:33:17 -0500 (Fri, 18 Jun 2010)
| 13 lines Eliminate deadlock potential in dahdi_fixup(). (This
is a backport of 269307, committed to trunk by rmudgett.) Calling
dahdi_indicate() when the channel private lock is already held
can cause a deadlock if the PRI lock is needed because
dahdi_indicate() will also get the channel private lock. The
pri_grab() function assumes that the channel private lock is held
once to avoid deadlock. (closes issue #17261) Reported by: aragon
........
2010-06-17 21:23 +0000 [r271231-271300] David Vossel <dvossel@digium.com>
* channels/sip/reqresp_parser.c: fixes some coding guideline issue
* channels/sip/include/dialog.h, channels/chan_sip.c,
channels/sip/include/sip.h: retransmit response to BYE requests
until timer J expires According to RFC 3261 section 17.2.2, which
describes non-INVITE server transaction, when a dialog enters the
Completed state it must destroy the dialog after Timer J (T1*64)
fires. For a BYE transaction Asterisk terminates the dialog
immediately during sip_hangup() when it should be waiting T1*64
ms. This results in some odd behavior. For instance if Asterisk
receives a BYE and transmits a 200ok in response, if the endpoint
never receives the 200ok it will retransmit the BYE to which
Asterisk responds with a "481 Call leg/transaction does not
exist" because the dialog is already gone. To resolve this I made
a function called sip_scheddestroy_final(). This differs slightly
from sip_schedestroy() in that it enables a flag that will
prevent the destruction from ever being rescheduled or canceled
afterwards. It also prevents the pvt's needdestroy flag from
being set which triggers the destruction of the dialog within the
do_monitor thread(). By using this function we are guaranteed
destruction will not occur until the scheduled time. This allows
Asterisk to respond to any possible retransmits for a dialog
after we process the initial BYE request for T1*64 ms. Other
changes: I removed two instances where sip_cancel_destroy is used
right before calling sip_scheddestroy. sip_scheddestroy always
calls sip_cancel_destroy before scheduling the new destruction so
it is completely unnecessary. Review:
https://reviewboard.asterisk.org/r/694/
* res/res_rtp_asterisk.c, main/rtp_engine.c, CHANGES: adds support
for slin16 in sip (closes issue #16153) Reported by: kfister
Patches: 16153-1.6.2.0-rc5.patch uploaded by kfister (license
912) slin16.sip.patch.1 uploaded by malcolmd (license 924) Tested
by: kfister, malcolmd
* main/channel.c, res/res_rtp_asterisk.c, main/frame.c,
main/rtp_engine.c, codecs/codec_speex.c, CHANGES,
include/asterisk/frame.h: adds speex 16khz audio support (closes
issue #17501) Reported by: fabled Patches:
asterisk-trunk-speex-wideband-v2.patch uploaded by fabled
(license 448) Tested by: malcolmd, fabled, dvossel
2010-06-17 15:34 +0000 [r271192] Jeff Peeler <jpeeler@digium.com>
* channels/sig_analog.c: Change expected operation from error to
debug message
2010-06-17 00:30 +0000 [r271089] Paul Belanger <paul.belanger@polybeacon.com>
* apps/app_meetme.c: option w[(secs)] incorrectly capitalized in
xmldoc (closes issue #17516) Reported by: karlfife
2010-06-16 22:37 +0000 [r271056] David Vossel <dvossel@digium.com>
* channels/sip/reqresp_parser.c: addition of more parse_uri test
cases
2010-06-16 21:17 +0000 [r270987] Paul Belanger <paul.belanger@polybeacon.com>
* /, configs/extensions.conf.sample: Merged revisions 270979 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r270979 | pabelanger | 2010-06-16 17:10:05 -0400 (Wed, 16 Jun
2010) | 4 lines Fixed typo in macro-page Reported to
#asterisk-dev by a student of jsmith. ........
2010-06-16 21:12 +0000 [r270981-270983] Jason Parker <jparker@digium.com>
* channels/chan_agent.c: Fix the actual place that was pointed out,
for previous commit.
* /, channels/chan_agent.c: Merged revisions 270980 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r270980 | qwell | 2010-06-16 16:10:09 -0500 (Wed, 16 Jun
2010) | 4 lines Need to lock the agent chan before access its
internal bits. Pointed out by russellb on asterisk-dev mailing
list. ........
2010-06-16 20:34 +0000 [r270974] Matthew Nicholson <mnicholson@digium.com>
* main/dnsmgr.c, main/acl.c: Set sin_family to AF_INET when doing
lookups, also reset sin_port the first time the ip address
changes. (closes issue #17496) Reported by: ManChicken (closes
issue #15827) Reported by: DennisD Patches: dnsmgr_15827.patch
uploaded by chappell (license 8) Tested by: DennisD, gentlec,
damage, wimpy
2010-06-16 19:03 +0000 [r270940] David Vossel <dvossel@digium.com>
* main/channel.c, res/res_rtp_asterisk.c, main/frame.c,
main/rtp_engine.c, channels/chan_sip.c, CHANGES,
channels/chan_iax2.c, include/asterisk/frame.h,
formats/format_g719.c (added): addition of G.719 pass-through
support (closes issue #16293) Reported by: malcolmd Patches:
g719.passthrough.patch.7 uploaded by malcolmd (license 924)
format_g719.c uploaded by malcolmd (license 924)
2010-06-16 18:43 +0000 [r270936] Paul Belanger <paul.belanger@polybeacon.com>
* res/res_agi.c, CHANGES: MSG_OOB flag on HANGUP packet removed.
Per Tilghman's request on IRC (#asterisk-bugs). (closes issue
#17506) Reported by: brycebaril Tested by: pabelanger, tilghman
2010-06-16 17:36 +0000 [r270867] David Vossel <dvossel@digium.com>
* /, channels/chan_iax2.c: Merged revisions 270866 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r270866 | dvossel | 2010-06-16 12:35:29 -0500 (Wed, 16
Jun 2010) | 22 lines fixes chan_iax2 race condition There is code
in chan_iax2.c that attempts to guarantee that only a single
active thread will handle a call number at a time. This code
works once the thread is added to an active_list of threads, but
we are not currently guaranteed that a newly activated thread
will enter the active_list immediately because it is left up to
the thread to add itself after frames have been queued to it.
This means that if two frames come in for the same call number at
the same time, it is possible for them to grab two separate
threads because the first thread did not add itself to the
active_list fast enough. This causes some pretty complex
problems. This patch resolves this race condition by immediately
adding an activated thread to the active_list within the network
thread and only depending on the thread to remove itself once it
is done processing the frames queued to it. By doing this we are
guaranteed that if another frame for the same call number comes
in at the same time, that this thread will immediately be found
in the active_list of threads. Review:
https://reviewboard.asterisk.org/r/720/ ........
2010-06-16 16:45 +0000 [r270836] Jeff Peeler <jpeeler@digium.com>
* channels/sig_analog.c: Fix no call waiting caller ID Clearing the
callwaitcas flag in analog_call was causing the incoming D digit
to be ignored which triggers sending the caller ID.
2010-06-16 15:05 +0000 [r270801] Paul Belanger <paul.belanger@polybeacon.com>
* doc/tex/channelvariables.tex: Update formatting for
channelvariables.tex (closes issue #17511) Reported by: klaus3000
Patches: channelvariables.tex-patch.txt uploaded by klaus3000
(license 65) Tested by: pabelanger
2010-06-15 22:48 +0000 [r270726] Russell Bryant <russell@digium.com>
* channels/sig_analog.c: Don't blow up if an ast_channel doesn't
get allocated.
2010-06-15 21:42 +0000 [r270658-270692] Terry Wilson <twilson@digium.com>
* main/http.c: Don't continue sending the file when there has been
an error If there is a problem with a firmware file, Polycom
phones will close the connection. We were continuing to send the
file anyway. There should be no reason to continue sending a file
if there is an error writing it. (closes issue #16682) Reported
by: lmadsen
* res/res_phoneprov.c: Don't send files twice and remove extra \r\n
from header After the manager http auth changes, we forgot to
remove the manual sending of the file. Also, ast_http_send adds
two \r\n to the header that is passed to it, so a trailing \r\n
is removed from the Content-type header. It might be better to
change ast_http_send, but I don't like changing the behavior of
an API function. (closes issue #17239) Reported by: cjacobsen
Patches: patch2.diff uploaded by cjacobsen (license 1029) Tested
by: lathama, cjacobsen
* channels/chan_sip.c: Make contactdeny apply to src ip when
nat=yes chan_sip's "contactdeny" feature screens the "to be
registered contact". In case of nat=yes it should not use the
address information from the Contact header (which is not used at
all for routing), but the source IP address of the request. Thus,
if nat=yes and a client sends a request from a denied IP address
(e.g. by spoofing the src-IP address) it can bypass the
screening. This commit makes contactdeny apply to the src ip when
nat=yes instead. (closes issue #17276) Reported by: klaus3000
Patches: patch-asterisk-trunk-contactdeny.txt uploaded by
klaus3000 (license 65) Tested by: klaus3000
2010-06-15 18:26 +0000 [r270519-270584] Tilghman Lesher <tlesher@digium.com>
* main/pbx.c, /: Merged revisions 270583 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r270583 | tilghman | 2010-06-15 13:25:12 -0500 (Tue, 15 Jun 2010)
| 5 lines Variables have always been case-sensitive, so we should
not be removing case-insensitive matches. Bug reported via the
-dev list. See
http://lists.digium.com/pipermail/asterisk-dev/2010-June/044510.html
........
* res/res_jabber.c: Argh, mixed declarations and code.
* configs/jabber.conf.sample, include/asterisk/jabber.h,
doc/distributed_devstate-XMPP.txt (added), CHANGES,
res/res_jabber.c: Add distributed devicestate via the XMPP
protocol. (closes issue #15757) Reported by: Marquis Patches:
distributed_devstate-XMPP.txt uploaded by lmadsen (license 10)
Tested by: Marquis, lmadsen, marcelloceschia Review:
https://reviewboard.asterisk.org/r/351/
2010-06-15 12:51 +0000 [r270443] Leif Madsen <lmadsen@digium.com>
* /, configs/voicemail.conf.sample: Merged revisions 270442 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r270442 | lmadsen | 2010-06-15 07:47:03 -0500 (Tue, 15 Jun 2010)
| 1 line Move information about zonemessages into the
[zonemessages] section. ........
2010-06-14 21:33 +0000 [r270332] Paul Belanger <paul.belanger@polybeacon.com>
* /, res/res_musiconhold.c: Merged revisions 270331 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r270331 | pabelanger | 2010-06-14 17:31:59 -0400 (Mon,
14 Jun 2010) | 14 lines Properly play first file in sort list.
When using sort=alpha we would always skip the first file in the
list first time through. We now check for that properly. (closes
issue #17470) Reported by: pabelanger Patches: sort.aplha.patch
uploaded by pabelanger (license 224) Tested by: lmadsen Review:
https://reviewboard.asterisk.org/r/703/ ........
2010-06-14 20:51 +0000 [r270298] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c, channels/sig_ss7.h, channels/sig_ss7.c:
Extract sig_ss7_init_linkset() to sig_ss7. Also found a place
where sig_pri_init_pri() was inlined and called it instead.
2010-06-14 19:41 +0000 [r270260] Jason Parker <jparker@digium.com>
* channels/chan_agent.c: Add option to get untruncated channel name
from AGENT function. The "channel" option would chop the channel
name at the last '-', which made it useless for something like a
channel transfer from the dialplan. The "fullchannel" option will
return the channel name as-is. ABE-2218
2010-06-14 15:55 +0000 [r270219] Richard Mudgett <rmudgett@digium.com>
* channels/sig_pri.h, channels/chan_dahdi.c,
configs/chan_dahdi.conf.sample, channels/sig_pri.c: Add digit
manipulation tag support to chan_dahdi/sig_pri like chan_misdn.
Add the append_msn_to_cid_tag option to chan_dahdi like
chan_misdn. Review: https://reviewboard.asterisk.org/r/696/
2010-06-13 09:16 +0000 [r270184] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
* autoconf/ast_check_pwlib.m4, configure: bashism in configure
script Theoretically the ./configure script is a pure
bourne-shell script. Practically it may be run by bash if /bin/sh
is not good enough. But we should not count on it. See bug report
for the gory details. (closes issue #17485) Patches:
0001-remove-bashism-from-ast_check_pwlib.m4.patch uploaded by
tzafrir (license 46)
2010-06-13 01:53 +0000 [r270042-270151] Paul Belanger <paul.belanger@polybeacon.com>
* configure, include/asterisk/autoconfig.h.in, configure.ac:
Reverting patch and reopening issue #16155, as patch breaks
FreeBSD / OSX builds.
* /, doc/HOWTO_collect_debug_information.txt: Merged revisions
270078 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r270078 | pabelanger | 2010-06-12 14:54:20 -0400 (Sat, 12 Jun
2010) | 2 lines Fix typo in example ........
* configure, include/asterisk/autoconfig.h.in, configure.ac: Use
pkg-config to find gmime libraries This way the libraries can be
found even if they are in non-standard locations. (closes issue
#16155) Reported by: jcollie Patches:
0008-change-configure.ac-to-look-for-pkg-config-gmime-2.0.patch
uploaded by jcollie (license 412) Tested by: jsmith, tilghman,
pabelanger
2010-06-11 18:31 +0000 [r269936-269976] Tilghman Lesher <tlesher@digium.com>
* main/frame.c, /: Merged revisions 269960 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r269960 | tilghman | 2010-06-11 13:23:05 -0500 (Fri, 11 Jun 2010)
| 8 lines For SpeeX, 0 bits remaining is valid and does not need
an emitted warning. (closes issue #15762) Reported by: nblasgen
Patches: issue15672.patch uploaded by pabelanger (license 224)
Tested by: nblasgen ........
* CHANGES, main/db.c: Add DBGetComplete event after a
DBGetResponse. (closes issue #16965) Reported by: rrb3942
Patches: DBGetComplete.patch uploaded by rrb3942 (license 1003)
* main/logger.c: Remove lines from the output related to the
backtrace itself.
2010-06-10 20:30 +0000 [r269889] Paul Belanger <paul.belanger@polybeacon.com>
* Makefile, makeopts.in: Remove ASTBINDIR variable (closes issue
#17031) Reported by: pabelanger Patches:
Makefile.ASTBINDIR.v2.patch uploaded by pabelanger (license 224)
Tested by: pabelanger, tilghman
2010-06-10 19:34 +0000 [r269749-269822] Mark Michelson <mmichelson@digium.com>
* main/channel.c, /: Merged revisions 269821 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r269821 | mmichelson | 2010-06-10 14:30:12 -0500 (Thu, 10 Jun
2010) | 19 lines Fix potential crash when writing raw SLIN audio
on a PLC-enabled channel. The issue here was that the frame
created when adjusting for PLC had no offset to its audio data.
If this frame were translated to another format prior to being
sent out an RTP socket, all went well because the translation
code would put an appropriate offset into the frame. However, if
the SLIN audio were not translated before being sent out the RTP
socket, bad things would happen. Specifically, the
ast_rtp_raw_write makes the assumption that the frame has at
least enough of an offset that it can accommodate an RTP header.
This was not the case. As such, data was being written prior to
the allocation, likely corrupting the data the memory allocator
had written. Thus when the time came to free the data, all hell
broke loose. ....Well, Asterisk crashed at least. The fix was
just what one would expect. Offset the data in the frame by a
reasonable amount. The method I used is a bit odd since the data
in the frame is 16 bit integers and not bytes. I left a big ol'
comment about it. This can be improved on if someone is
interested. I was more interested in getting the crash resolved.
........
* doc/tex/plc.tex (added), doc/tex/asterisk.tex: Add documentation
explaining PLC in Asterisk. Review:
https://reviewboard.asterisk.org/r/688/
2010-06-10 13:17 +0000 [r269711] Russell Bryant <russell@digium.com>
* tests/test_heap.c: Fix an off by one error that caused a unit
test to occasionally crash.
2010-06-10 12:28 +0000 [r269707] Kevin P. Fleming <kpfleming@digium.com>
* main/logger.c: Ensure that 'logger show channels' works properly
when wildcards are used in logger.conf.
2010-06-10 08:15 +0000 [r269636] Tilghman Lesher <tlesher@digium.com>
* /, main/logger.c, utils/extconf.c, main/asterisk.c: Merged
revisions 269635 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r269635 | tilghman | 2010-06-10 02:52:34 -0500 (Thu, 10 Jun 2010)
| 9 lines Ensure restartable system calls can restart (BSD signal
semantics). This eliminates the annoying <beep> on the console.
(closes issue #17477) Reported by: jvandal Patches:
20100610__issue17477.diff.txt uploaded by tilghman (license 14)
........
2010-06-10 00:32 +0000 [r269417-269602] Russell Bryant <russell@digium.com>
* channels/chan_dahdi.c: Attempt to fix a FreeBSD build error by
including sys/stat.h.
http://bamboo.asterisk.org/download/AST-TRUNKFREEBSD/build_logs/AST-TRUNKFREEBSD-187.log
* main/lock.c: Attempt to fix FreeBSD build problem.
* /, channels/chan_oss.c: Merged revisions 269495 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r269495 | russell | 2010-06-09 17:18:37 -0500 (Wed, 09 Jun 2010)
| 2 lines Don't stop Asterisk if chan_oss fails to register
'Console' (due to another channel driver already claiming it).
........
* include/asterisk/event.h, main/event.c: Resolve an invalid memory
read on an event. Valgrind pointed out that attempting to get an
IE value from an event that has no IEs produces an invalid memory
read past the end of the event. Thanks to mmichelson for pointing
the problem out to me and then testing the fix.
2010-06-09 17:32 +0000 [r269346] Paul Belanger <paul.belanger@polybeacon.com>
* contrib/init.d/rc.debian.asterisk, /, main/term.c: Merged
revisions 269334 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r269334 | pabelanger | 2010-06-09 13:24:53 -0400 (Wed, 09 Jun
2010) | 12 lines Fix Debian init script to not use -c. When using
the init script as-is currently, it could cause issues on Debian
such as high CPU usage. This fix has worked for several people so
I'm implementing the change. We now handle color displays
properly. (closes issue #16784) Reported by: pabelanger Patches:
20100530__issue16784__2.diff.txt uploaded by tilghman (license
14) Tested by: pabelanger, tilghman ........
2010-06-09 17:06 +0000 [r269307-269308] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c, channels/sig_ss7.h, channels/sig_ss7.c:
Add missing API function to sig_ss7: sig_ss7_fixup().
* channels/chan_dahdi.c: Eliminate deadlock potential in
dahdi_fixup(). Calling dahdi_indicate() within dahdi_fixup()
while the owner pointers are in a potentially inconsistent state
is a potentially bad thing in principle. However, calling
dahdi_indicate() when the channel private lock is already held
can cause a deadlock if the PRI lock is needed because
dahdi_indicate() will also get the channel private lock. The
pri_grab() function assumes that the channel private lock is held
once to avoid deadlock.
2010-06-09 15:09 +0000 [r269271] David Vossel <dvossel@digium.com>
* res/res_musiconhold.c: fixes crash in moh when cachertclasses
flag is used The result for moh_register was not verified to
guarantee the mohclass as added to the container. (closes issue
#16993) Reported by: dmitri Patches:
res_musiconhold_rtclass2.patch uploaded by dmitri (license 1001)
moh_crash2.diff uploaded by dvossel (license 671) Tested by:
dmitri
2010-06-09 13:17 +0000 [r269238] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
* channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, CHANGES:
dial by name in chan_dahdi * chan_dahdi supports dialing
configuring and dialing by device file name.
DAHDI/span-name!local!1 will use /dev/dahdi/span-name/local/1 .
Likewise it may appear in chan_dahdi.conf as 'channel =>
span-name!local!1'. * A new options for chan_dahdi.conf:
'ignore_failed_channels'. Boolean. False by default. If set,
chan_dahdi will ignore failed 'channel' entries. Handy for the
above name-based syntax as it does not depend on initialization
order. * have my_pri_make_cc_dialstring() only manupulate
dial-strings of group (gGrR) dialing, which make it lsightly more
complicated. https://reviewboard.asterisk.org/r/535/
2010-06-09 10:55 +0000 [r269187-269205] Russell Bryant <russell@digium.com>
* contrib/scripts/install_prereq: Add libjack-dev to
install_prereq.
* contrib/scripts/install_prereq: Add libpopt-dev, libical-dev, and
libspandsp-dev to install_prereq.
* contrib/scripts/install_prereq: Add libnewt-dev to
install-prereq.
* contrib/scripts/install_prereq: Add libopenais-dev to
install_prereq.
* contrib/scripts/install_prereq: Add an "install-unpackaged"
command to install_prereq for installing unpackaged dependencies
(such as NBS and libresample).
* contrib/scripts/install_prereq: Add libcurl to install_prereq.
* contrib/scripts/install_prereq: Add freetds-dev to
install_prereq.
* contrib/scripts/install_prereq: Add libradiusclient-ng-dev to
install_prereq.
* contrib/scripts/install_prereq: Add libbluetooth-dev to
install_prereq.
* contrib/scripts/install_prereq: Add libmysqlclient-dev to
install_prereq.
* contrib/scripts/install_prereq: Add libgtk2.0-dev to the packages
list for install_prereq.
2010-06-08 23:48 +0000 [r269153] Bradley Latus <brad.latus@gmail.com>
* configs/cdr_custom.conf.sample, configs/cdr_tds.conf.sample,
cdr/cdr_sqlite.c, configs/cdr_sqlite3_custom.conf.sample,
funcs/func_cdr.c, configs/cdr_syslog.conf.sample, UPGRADE.txt,
cdr/cdr_adaptive_odbc.c, addons/cdr_mysql.c, cdr/cdr_pgsql.c,
CHANGES, cdr/cdr_odbc.c, cdr/cdr_tds.c,
configs/cdr_odbc.conf.sample: Add High Resolution Times to CDRs
for Asterisk People expressed an interest in having access to the
exact length of calls to a finer degree than seconds. See the
CHANGES and UPGRADE.txt for usage also updated the sample configs
to note the change. Patch by snuffy. (closes issue #16559)
Reported by: cianmaher Tested by: cianmaher, snuffy Review:
https://reviewboard.asterisk.org/r/461/
2010-06-08 22:45 +0000 [r269119] Tilghman Lesher <tlesher@digium.com>
* configure, include/asterisk/autoconfig.h.in, configure.ac,
include/asterisk/localtime.h: Fix build on Mac OS X (and maybe
FreeBSD, too)
2010-06-08 18:50 +0000 [r269083] Matthew Nicholson <mnicholson@digium.com>
* apps/app_fax.c: Don't pass null to manager_event() (closes issue
#17087) Reported by: bklang Patches: app-fax-null-sprintf1.diff
uploaded by mnicholson (license 96) Tested by: bklang
2010-06-08 15:41 +0000 [r269008] Russell Bryant <russell@digium.com>
* Makefile.rules: Ensure CONFIG_FLAGS makes it into the build rules
when doing out of tree builds. (closes issue #16685) Reported by:
pprindeville
2010-06-08 15:39 +0000 [r269007] Sean Bright <sean@malleable.com>
* /, cdr/cdr_tds.c: Merged revisions 269006 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r269006 | seanbright | 2010-06-08 11:28:49 -0400 (Tue, 08 Jun
2010) | 11 lines Reduce startup time for cdr_tds with large CDR
tables. Since we are just checking for table existence, add a
WHERE clause that will return no rows but will raise an error if
the table doesn't exist. (closes issue #17380) Reported by:
kkwong Patches: issue17380-01.patch uploaded by seanbright
(license 71) Tested by: kkwong ........
2010-06-08 15:23 +0000 [r268969-268988] Leif Madsen <lmadsen@digium.com>
* configs/sip.conf.sample: Update note in sip.conf.sample. Update
note in sip.conf.sample about externip and externhost with STUN.
(closes issue #16323) Reported by: klaus3000 Patches:
sip.conf.sample-patch.txt uploaded by klaus3000 (license 65)
* apps/app_meetme.c, main/ccss.c, include/asterisk/data.h,
res/res_jabber.c, res/res_config_sqlite.c,
include/asterisk/callerid.h, channels/chan_dahdi.c,
include/asterisk/bridging_technology.h,
include/asterisk/doxyref.h, include/asterisk/event.h,
include/asterisk/astmm.h, main/ast_expr2f.c, main/features.c,
include/asterisk/timing.h, include/asterisk/rtp_engine.h,
include/asterisk/ccss.h, include/asterisk/threadstorage.h,
include/asterisk/xml.h, main/pbx.c, channels/chan_sip.c,
include/asterisk/astobj2.h, include/asterisk/channel.h,
include/asterisk/calendar.h, include/asterisk/manager.h,
include/asterisk/features.h, include/asterisk/logger.h,
include/asterisk/http.h, channels/sig_pri.h,
include/asterisk/app.h, main/audiohook.c, include/asterisk/pbx.h,
include/asterisk/dnsmgr.h, include/asterisk/smdi.h,
apps/app_voicemail.c: Fix some doxygen warnings. (closes issue
#17336) Reported by: snuffy Patches: doxygen-fixes1.diff uploaded
by snuffy (license 35) Tested by: russell
2010-06-08 06:57 +0000 [r268896-268933] Tilghman Lesher <tlesher@digium.com>
* res/res_config_sqlite.c: Release list lock before returning on
error.
* utils/extconf.c: Fix trunk build on Mac OS X.
2010-06-08 05:29 +0000 [r268894] Terry Wilson <twilson@digium.com>
* channels/sip/sdp_crypto.c (added), res/res_rtp_asterisk.c,
main/global_datastores.c, main/rtp_engine.c,
include/asterisk/res_srtp.h (added), channels/sip/srtp.c (added),
channels/chan_sip.c, include/asterisk/autoconfig.h.in,
res/res_srtp.exports.in (added), configure.ac, CHANGES,
channels/chan_iax2.c, res/res_srtp.c (added), main/channel.c,
build_tools/menuselect-deps.in, main/asterisk.exports.in,
configure, funcs/func_channel.c,
channels/sip/dialplan_functions.c,
channels/sip/include/sdp_crypto.h (added),
doc/tex/secure-calls.tex (added),
include/asterisk/global_datastores.h, channels/sip/include/srtp.h
(added), makeopts.in, include/asterisk/rtp_engine.h,
include/asterisk/frame.h, doc/tex/asterisk.tex,
channels/sip/include/sip.h: Add SRTP support for Asterisk After 5
years in mantis and over a year on reviewboard, SRTP support is
finally being comitted. This includes generic CHANNEL dialplan
functions that work for getting the status of whether a call has
secure media or signaling as defined by the underlying channel
technology and for setting whether or not a new channel being
bridged to a calling channel should have secure signaling or
media. See doc/tex/secure-calls.tex for examples. Original patch
by mikma, updated for trunk and revised by me. (closes issue
#5413) Reported by: mikma Tested by: twilson, notthematrix,
hemanshurpatel Review: https://reviewboard.asterisk.org/r/191/
2010-06-08 00:45 +0000 [r268857] Richard Mudgett <rmudgett@digium.com>
* channels/sip/dialplan_functions.c: Make SIP tests compile again.
2010-06-07 22:56 +0000 [r268817-268818] Tilghman Lesher <tlesher@digium.com>
* channels/chan_sip.c: Use the mailbox destructor function,
instead.
* channels/chan_sip.c, channels/sip/include/sip.h: Mailbox list
would previously grow at each reload, containing duplicates.
Also, optimize the allocation of mailboxes to avoid additional
memory structures. (closes issue #16320) Reported by: Marquis
Patches: 20100525__issue16320.diff.txt uploaded by tilghman
(license 14)
2010-06-07 20:04 +0000 [r268774] Richard Mudgett <rmudgett@digium.com>
* channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_ss7.h
(added), channels/Makefile, channels/sig_pri.c,
channels/sig_ss7.c (added): Extract sig_ss7 out of chan_dahdi.
Extract the SS7 specific code out of chan_dahdi like what was
done to ISDN/PRI and analog signaling. The new SS7 structures
were modeled on sig_pri. The changes to sig_pri are an
enhancement and a bug fix made possible because SS7 was
extracted. 1) The sig_pri TRANSFERCAPABILITY channel variable
should have been set unconditionally in
sig_pri_new_ast_channel(). 2) SS7/PRI transfer capability
interaction in dahdi_new() fixed because of SS7 extraction. 3)
Module ref count error in dahdi_new() if startpbx failed to start
the PBX for some reason. Review:
https://reviewboard.asterisk.org/r/661/
2010-06-07 19:52 +0000 [r268773] Tilghman Lesher <tlesher@digium.com>
* main/rtp_engine.c, channels/chan_sip.c,
channels/sip/dialplan_functions.c, include/asterisk/rtp_engine.h:
Seems strange (and the code backs up) that if the max and min of
a statistic is expressed as a double, the last value would not
also need to be a double. (closes issue #15807) Reported by:
klaus3000
2010-06-07 19:06 +0000 [r268734] Richard Mudgett <rmudgett@digium.com>
* channels/sig_pri.c: Moved AOC request code out of the middle of
code parsing the dialed number.
2010-06-07 18:59 +0000 [r268731] Tilghman Lesher <tlesher@digium.com>
* main/manager.c: Event well was going dry. (issue #17234)
2010-06-07 17:34 +0000 [r268690] Paul Belanger <paul.belanger@polybeacon.com>
* main/dsp.c: Set threshold for silence detection defaults to 256
(closes issue #15685) Reported by: david_s5 Patches:
dsp-silence-threshold-init.diff uploaded by dant (license 670)
issue15685.patch.v5 uploaded by pabelanger (license 224) Tested
by: danti Review: https://reviewboard.asterisk.org/r/670/
2010-06-07 17:14 +0000 [r268653] Tilghman Lesher <tlesher@digium.com>
* res/res_smdi.c: Avoid unloading res_smdi twice. (closes issue
#17237) Reported by: pabelanger
2010-06-07 15:51 +0000 [r268578] Richard Mudgett <rmudgett@digium.com>
* main/file.c: Suppress warning in waitstream_core(). Suppress the
warning about unexpected control subclass frames for
AST_CONTROL_CONNECTED_LINE, AST_CONTROL_REDIRECTING, and
AST_CONTROL_AOC in file.c:waitstream_core().
2010-06-06 05:29 +0000 [r268454-268534] Tilghman Lesher <tlesher@digium.com>
* contrib/init.d/rc.redhat.asterisk: Take advantage of variable
substitution already in the Makefile to specify the correct
location for files in init.d. (closes issue #16979) Reported by:
jw-asterisk (issue #15691) Reported by: itamarjp
* channels/chan_iax2.c: Finally track down and eliminate the
"FRACK! warnings from chan_iax2".
* main/dsp.c: Fix crash in DTMF detection. What I did not
originally see in my previous commit was that even though the
next digit could be detected before the previous was considered
ended, the detection of the next digit effectively ends the
detection of the previous. Therefore, the length moves in
lockstep with the digit, and no separate counter is needed for
the length alone. (closes issue #17371) Reported by: alecdavis
(closes issue #17474) Reported by: kenner
* main/manager.c: Verify event is not NULL before attempting to
lower its usecount. (closes issue #17234) Reported by: mav3rick
2010-06-05 05:23 +0000 [r268395-268417] Kevin P. Fleming <kpfleming@digium.com>
* CHANGES: Typo fix.
* CHANGES: Grammatical error fix.
2010-06-05 02:51 +0000 [r268321] Tilghman Lesher <tlesher@digium.com>
* /, configs/voicemail.conf.sample: Merged revisions 268320 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r268320 | tilghman | 2010-06-04 21:49:52 -0500 (Fri, 04 Jun 2010)
| 3 lines Rest In Peace
http://www.outandaboutnewspaper.com/article/4061 ........
2010-06-04 22:37 +0000 [r268205-268281] David Vossel <dvossel@digium.com>
* channels/chan_sip.c: fixes compile error from uninitialized
variable
* channels/chan_sip.c: RFC3261 compliant sip unreliable retransmit
timing + 'registerattempts' option tweak Changes. 1. RFC 3261
states in section 17.1.2.2 and 17.1.1.2 that retransmission
timers should initially be set to timer T1. T1 by default is
500ms. Asterisk was starting the retransmission timers at T1*2
which shouldn't cause any problems, but is not RFC compliant. 2.
RFC 3261 states in section 17.1.2.2 that for a non-INVITE client
transaction, if the retransmit timer fires while in the
proceeding state that the request must be retransmitted. Asterisk
currently ack's requests for both INVITE and non-INVITE
transactions when a 1XX response is received, this patch changes
this for non-INVITE requests. 3. The 'registerattempts' option in
sip.conf is supposed to set how many registry attempts will be
made before giving up. When this option is set to 1, I would
expect only one registry attempt to be made before stopping
because of a failure, but instead two are made. In my opinion
this is not expected behavior. This option does not indicate that
these are re-attempts. The logic behind this option has been
changed to only attempt registers the exact number of times this
option is set to. If this option is 0, it still continues to
re-attempt the registration forever. Review:
https://reviewboard.asterisk.org/r/687/
2010-06-04 20:42 +0000 [r267972-268127] Tilghman Lesher <tlesher@digium.com>
* /, configure, configure.ac: Merged revisions 268126 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r268126 | tilghman | 2010-06-04 15:41:24 -0500 (Fri, 04
Jun 2010) | 2 lines AC_CONFIG_SUBDIRS has a bad side-effect on
cross-compiles. ........
* Makefile, /, makeopts.in: Merged revisions 268050 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r268050 | tilghman | 2010-06-04 14:38:57 -0500 (Fri, 04
Jun 2010) | 6 lines Build menuselect with the build environment's
compiler, not the host (target)'s compiler. (closes issue #17464)
Reported by: pprindeville Tested by: tilghman ........
* /, configure, configure.ac, autoconf/libcurl.m4: Merged revisions
267971 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r267971 | tilghman | 2010-06-04 11:27:02 -0500 (Fri, 04 Jun 2010)
| 2 lines As-fixiate the build process ........
2010-06-04 14:45 +0000 [r267928] Richard Mudgett <rmudgett@digium.com>
* channels/sig_pri.c: Incoming overlap dialing no longer works
after sig_pri extraction. The problem would manifest itself if
your dialplan matching could accept more digits to match than
were actually dialed. The time out waiting for overlap digits
disconnected the call instead of matching any accumulated digits
to the dialplan. Accidental conversion of a break out of loop as
a break out of switch. (closes issue #17401) Reported by:
avalentin Patches: issue17401_digit_timeout.patch uploaded by
rmudgett (license 664) Tested by: avalentin, rmudgett
2010-06-04 03:20 +0000 [r267877] Tilghman Lesher <tlesher@digium.com>
* include/asterisk/slin.h: As signed linear audio data is accessed
as 16-bit values, certain processors require the values to be
aligned in memory. (closes issue #16912) Reported by:
michaelevdokimov Patches: asterisk.patch uploaded by
michaelevdokimov (license 997) Tested by: michaelevdokimov
2010-06-04 03:11 +0000 [r267863] Terry Wilson <twilson@digium.com>
* channels/chan_sip.c: Send an ACK for every final response
received for an INVITE From issue ABE-2247. RFC 3261 compliance
for sections 13.2.24 and 17.1.1.2. Review:
https://reviewboard.asterisk.org/r/692/
2010-06-04 02:58 +0000 [r267775-267862] Tilghman Lesher <tlesher@digium.com>
* include/asterisk/slin.h: As signed linear audio data is accessed
as 16-bit values, certain processors require the values to be
aligned in memory. (closes issue #16912) Reported by:
michaelevdokimov
* configure, autoconf/ast_ext_lib.m4: If there's a default, turn it
on, even when the option isn't specified.
* /, configure, include/asterisk/autoconfig.h.in, configure.ac:
Merged revisions 267759 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r267759 | tilghman | 2010-06-03 20:16:26 -0500 (Thu, 03 Jun 2010)
| 7 lines Make the default install path appear to be /usr on
Linux, instead of /usr/local. Also, reorganize the options, so
that they're more alphabetical. (closes issue #17013) Reported
by: klaus3000 ........
2010-06-03 20:41 +0000 [r267714] Russell Bryant <russell@digium.com>
* main/ccss.c: Remove a LOG_WARNING. This came up when using the
sample configs, and just indicates expected behavior.
2010-06-03 19:46 +0000 [r267669] Tilghman Lesher <tlesher@digium.com>
* funcs/func_odbc.c: Handle OOM errors more gracefully. (closes
issue #17084) Reported by: falves11 Patches:
issue17084_162_A.diff uploaded by falves11 (license 374) Tested
by: falves11
2010-06-03 18:53 +0000 [r267624] Leif Madsen <lmadsen@digium.com>
* UPGRADE.txt, CHANGES: Update UPGRADE.txt and CHANGE for CDR
functionality changes. Updated the UPGRADE.txt and CHANGES file
stating that CDR records will not be explicity written unless
cdr.conf exists and is configured. (closes issue #17373) Reported
by: wdoekes Tested by: pabelanger
2010-06-03 18:38 +0000 [r267622] Richard Mudgett <rmudgett@digium.com>
* codecs/codec_dahdi.c: Make compile again.
2010-06-03 17:31 +0000 [r267537] Russell Bryant <russell@digium.com>
* channels/chan_usbradio.c: Don't stop Asterisk if chan_usbradio
isn't configured.
2010-06-03 17:09 +0000 [r267492] Mark Michelson <mmichelson@digium.com>
* codecs/codec_lpc10.c, codecs/codec_g722.c, codecs/codec_adpcm.c,
codecs/codec_alaw.c, main/translate.c, codecs/codec_g726.c,
codecs/codec_gsm.c, codecs/codec_ulaw.c, codecs/codec_dahdi.c,
include/asterisk/translate.h: Remove unnecessary code relating to
PLC. The logic for handling generic PLC is now handled in
ast_write in channel.c instead of in translation code. Review:
https://reviewboard.asterisk.org/r/683/
2010-06-03 17:05 +0000 [r267445-267490] Russell Bryant <russell@digium.com>
* channels/chan_usbradio.c: Remove a line that was killing Asterisk
on startup.
* channels/h323/Makefile.in: Comment out a rule that likes to run
implicitly unnecessarily, breaking builds
2010-06-03 00:02 +0000 [r267399] Richard Mudgett <rmudgett@digium.com>
* channels/sig_pri.h, channels/chan_dahdi.c,
configs/chan_dahdi.conf.sample, configure,
include/asterisk/autoconfig.h.in, configure.ac, CHANGES,
channels/sig_pri.c: Add ETSI Message Waiting Indication (MWI)
support. Add the ability to report waiting messages to ISDN
endpoints (phones). Relevant specification: EN 300 650 and EN 300
745 Review: https://reviewboard.asterisk.org/r/599/
2010-06-02 22:46 +0000 [r267352] Russell Bryant <russell@digium.com>
* channels/Makefile, channels/h323/Makefile.in: try to fix some
random chan_h323 compilation failures After some debugging, the
random chan_h323 build failures appear to be due to complications
introduced by some chan_h323 specific build stuff getting
triggered during a clean. Simplify this by moving the h323 clean
commands down into channels/makefile.
2010-06-02 22:28 +0000 [r267350] Richard Mudgett <rmudgett@digium.com>
* main/channel.c, configure, include/asterisk/autoconfig.h.in,
configure.ac, include/asterisk/channel.h, CHANGES,
channels/sig_pri.c: Add ETSI Malicious Call ID support. Add the
ability to report malicious callers as an AMI event in the call
event class. Relevant specification: EN 300 180 Review:
https://reviewboard.asterisk.org/r/576/
2010-06-02 21:44 +0000 [r267303-267305] Russell Bryant <russell@digium.com>
* utils/extconf.c: Fix a build error on mac.
* main/Makefile: Ensure the -Wno-strict-aliasing flag makes it,
even if ASTCFLAGS has been specified. When ASTCFLAGS was
specified with the make command, Makefile.rules was using the
specified value from the command line and not the one here,
making it so this flag would go missing.
2010-06-02 21:05 +0000 [r267261] Richard Mudgett <rmudgett@digium.com>
* channels/sig_pri.h, channels/chan_dahdi.c,
configs/chan_dahdi.conf.sample, configure,
include/asterisk/autoconfig.h.in, configure.ac, CHANGES,
channels/sig_pri.c: Add ETSI Call Waiting support. Add the
ability to announce a call to an endpoint when there are no B
channels available. A call waiting call is a SETUP message with
no B channel selected. Relevant specification: EN 300 056, EN 300
057, EN 300 058 For DAHDI/ISDN channels, the CHANNEL() dialplan
function now supports the "no_media_path" option. * Returns "0"
if there is a B channel associated with the call. * Returns "1"
if no B channel is associated with the call. The call is either
on hold or is a call waiting call. If you are going to allow
incoming call waiting calls then you need to use
CHANNEL(no_media_path) do determine if you must drop a call to
accept the new call. Review:
https://reviewboard.asterisk.org/r/568/
2010-06-02 19:33 +0000 [r267181] David Vossel <dvossel@digium.com>
* CHANGES, doc/advice_of_charge.txt: Update CHANGES and aoc help
doc to reflect AOC additions
2010-06-02 18:53 +0000 [r267138] Russell Bryant <russell@digium.com>
* main/cli.c: Add a CLI command that blocks until Asterisk has
fully booted. Review: https://reviewboard.asterisk.org/r/684/
2010-06-02 18:13 +0000 [r267097] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Prevent use of uninitialized values. Two
struct sockaddr_ins are created when applying directmedia host
access rules. The addresses of these are passed to the RTP engine
to be filled in. However, the RTP engine inspects the fields of
the structs before actually taking action. This inspection caused
valgrind to be a bit unhappy.
2010-06-02 18:10 +0000 [r267096] Richard Mudgett <rmudgett@digium.com>
* apps/app_dial.c, configs/chan_dahdi.conf.sample,
include/asterisk/aoc.h (added), channels/chan_sip.c,
configs/manager.conf.sample, main/aoc.c (added),
apps/app_queue.c, channels/sig_pri.c, doc/advice_of_charge.txt
(added), main/channel.c, channels/sig_pri.h,
channels/chan_dahdi.c, main/manager.c, main/features.c,
tests/test_aoc.c (added), configs/sip.conf.sample,
include/asterisk/frame.h, main/asterisk.c,
channels/sip/include/sip.h: Generic Advice of Charge. Asterisk
Generic AOC Representation - Generic AOC encode/decode routines.
(Generic AOC must be encoded to be passed on the wire in the
AST_CONTROL_AOC frame) - AST_CONTROL_AOC frame type to represent
generic encoded AOC data - Manager events for AOC-S, AOC-D, and
AOC-E messages Asterisk App Support - app_dial AOC-S pass-through
support on call setup - app_queue AOC-S pass-through support on
call setup AOC Unit Tests - AOC Unit Tests for encode/decode
routines - AOC Unit Test for manager event representation. SIP
AOC Support - Pass-through of generic AOC-D and AOC-E messages to
snom phones via the snom AOC specification. - Creation of
chan_sip page3 flags for the addition of the new
'snom_aoc_enabled' sip.conf option. IAX AOC Support - Natively
supports AOC pass-through through the use of the new
AST_CONTROL_AOC frame type DAHDI AOC Support - ETSI PRI full AOC
Pass-through support - 'aoc_enable' chan_dahdi.conf option for
independently enabling pass-through of AOC-S, AOC-D, AOC-E. -
'aoce_delayhangup' option for retrieving AOC-E on disconnect. -
DAHDI A() dial string option for requesting AOC services. example
usage: ;requests AOC-S, AOC-D, and AOC-E on call setup
exten=>1111,1,Dial(DAHDI/g1/1112/A(s,d,e)) Review:
https://reviewboard.asterisk.org/r/552/
2010-06-02 17:57 +0000 [r267093] Russell Bryant <russell@digium.com>
* apps/app_voicemail.c: Silence a compiler warning.
2010-06-02 17:29 +0000 [r267065] Jeff Peeler <jpeeler@digium.com>
* include/asterisk/slin.h: Fix infinite loop when loading codec
speex This changes the sample slinear frame data to contain
non-zero data so that translation calculations for speex works
when preprocessing and VAD is turned on. The encoder expects
samples to be returned, but when attempted with the mentioned two
options and silent sample frames everything was discarded.
(closes issue #17240) Reported by: seandarcy Review:
https://reviewboard.asterisk.org/r/682/
2010-06-02 17:25 +0000 [r267041] Paul Belanger <paul.belanger@polybeacon.com>
* /, main/ast_expr2.y: Merged revisions 267009 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r267009 | pabelanger | 2010-06-02 13:14:37 -0400 (Wed, 02 Jun
2010) | 7 lines Cleanup error/warning messages in AEL2 parser
(closes issue #16684) Reported by: Silmaril Patches:
patch_ael2_logmsg.diff uploaded by Silmaril (license 979)
........
2010-06-02 17:13 +0000 [r266926-267008] Richard Mudgett <rmudgett@digium.com>
* main/manager.c, configure, include/asterisk/autoconfig.h.in,
configure.ac, configs/manager.conf.sample, CHANGES,
channels/sig_pri.c, include/asterisk/manager.h: Add ETSI Advice
Of Charge (AOC) event reporting. This feature generates AMI
events in the new aoc event class from the events passed up by
libpri. Review: https://reviewboard.asterisk.org/r/537/
* channels/sig_pri.h, channels/chan_dahdi.c,
configs/chan_dahdi.conf.sample, configure,
include/asterisk/autoconfig.h.in, configure.ac, CHANGES,
channels/sig_pri.c: Add ETSI Explicit Call Transfer (ECT)
support. Added ability to send and receive ETSI Explicit Call
Transfer (ECT) messages to eliminate tromboned calls. Note:
Asterisk already supported initiating the transfer of calls to
eliminate tromboned calls to libpri so there was nothing to do
for the asterisk portion. Review:
https://reviewboard.asterisk.org/r/520/
2010-06-02 13:32 +0000 [r266877] Paul Belanger <paul.belanger@polybeacon.com>
* main/bridging.c: pthread_join to assure the thread is really gone
(closes issue #15465) Reported by: fnordian Patches:
bridging.patch uploaded by fnordian (license 110) Tested by:
lmadsen, fnordian, peterh Review:
https://reviewboard.asterisk.org/r/679/
2010-06-01 22:14 +0000 [r266832] Terry Wilson <twilson@digium.com>
* res/res_calendar_exchange.c: Use the correct ical.h file
2010-06-01 21:28 +0000 [r266828] Tilghman Lesher <tlesher@digium.com>
* configure, include/asterisk/autoconfig.h.in, tests/test_locale.c
(added), configure.ac, configs/voicemail.conf.sample,
include/asterisk/localtime.h, main/stdtime/localtime.c, CHANGES,
apps/app_voicemail.c: Support setting locale per-mailbox (changes
date/time languages for email, pager messages). (closes issue
#14333) Reported by: klaus3000 Patches:
20090515__issue14333.diff.txt uploaded by tilghman (license 14)
app_voicemail.c-svn-trunk-rev211675-patch.txt uploaded by
klaus3000 (license 65) Tested by: klaus3000
2010-06-01 21:12 +0000 [r266786] Terry Wilson <twilson@digium.com>
* apps/app_dial.c, UPGRADE.txt: Set app and appdata fields when a
Dial is redirected (closes issue #17204) Reported by: one47
Tested by: twilson, one47
2010-06-01 18:02 +0000 [r266592-266735] Tilghman Lesher <tlesher@digium.com>
* res/res_smdi.c: Don't register functions until the last possible
point, so they're not unloaded unnecessarily. (closes issue
#15996) Reported by: junky Patches: sdmi_wait.diff uploaded by
junky (license 177)
* main/manager.c: Eliminate stale manager events after a set
interval, even if AMI clients don't query for them. Actions (or
failures to act) by external clients should not cause memory
leaks in Asterisk, especially when those continued leaks could
cause Asterisk to misbehave later. (closes issue #17234) Reported
by: mav3rick Patches: 20100510__issue17234.diff.txt uploaded by
tilghman (license 14) 20100517__issue17234__trunk.diff.txt
uploaded by tilghman (license 14) Tested by: mav3rick, davidw
(closes issue #17365) Reported by: davidw
* /, main/asterisk.c: Merged revisions 266585 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r266585 | tilghman | 2010-06-01 10:17:46 -0500 (Tue, 01 Jun 2010)
| 11 lines Prevent CLI prompt from distorting output of lines
shorter than the prompt. Uses the VT100 method of clearing the
line from the cursor position to the end of the line: Esc-0K
(closes issue #17160) Reported by: coolmig Patches:
20100531__issue17160.diff.txt uploaded by tilghman (license 14)
Tested by: coolmig ........
2010-05-30 20:18 +0000 [r266438-266522] Tilghman Lesher <tlesher@digium.com>
* funcs/func_env.c: Needs to be wrapped in <para>
* contrib/init.d/rc.debian.asterisk, /: Merged revisions 266437 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r266437 | tilghman | 2010-05-29 23:43:28 -0500 (Sat, 29 May 2010)
| 2 lines Reverting patch and reopening issue #16784, as patch
breaks color display. ........
2010-05-28 22:54 +0000 [r266386] Terry Wilson <twilson@digium.com>
* res/res_calendar_icalendar.c, configure, configure.ac,
res/res_calendar_caldav.c: Fix ical library handling (again)
Newer versions of libical (which we require) store the header
file in a libical/ subfolder and include an ical.h file that does
a #warning for deprecation and then #includes <libical/ical.h>.
Since we now test for libical/ical.h, we can change the #includes
back to <libical/ical.h> and remove the test which specifically
adds /usr/include/libical as an include directory.
2010-05-28 22:50 +0000 [r266337-266385] Tilghman Lesher <tlesher@digium.com>
* funcs/func_env.c, UPGRADE.txt, main/asterisk.c: Setup environment
variables for the benefit of child processes and disallow
changing them. (closes issue #14899) Reported by: jmls Patches:
20090916__issue14899.diff.txt uploaded by tilghman (license 14)
Tested by: jmls
* main/asterisk.c: Only report swap on platforms which can examine
those statistics
2010-05-28 17:55 +0000 [r266292] David Vossel <dvossel@digium.com>
* channels/chan_sip.c: fixes crash when creation of UDPTL fails
(closes issue #17264) Reported by: falves11 Patches:
issue_17264_reviewboard_fix.diff uploaded by dvossel (license
671) issue_17264_1.6.2_reviewboard_fix.diff uploaded by dvossel
(license 671) Tested by: falves11
2010-05-28 17:34 +0000 [r266289] Terry Wilson <twilson@digium.com>
* configure, configure.ac, makeopts.in: More build fixes for
ical/neon and res_calendar_ews
2010-05-27 20:08 +0000 [r266240] Jeff Peeler <jpeeler@digium.com>
* pbx/pbx_realtime.c: fix compile error
2010-05-27 19:25 +0000 [r266146-266238] Tilghman Lesher <tlesher@digium.com>
* pbx/pbx_realtime.c, CHANGES: Cache query results for one second.
Queries from the PBX core come in 3's. Caching avoids the
additional performance penalty from those two additional queries
hitting the database. (closes issue #16521) Reported by: tilghman
Patches: 20091229__issue16521.diff.txt uploaded by tilghman
(license 14) Tested by: Hubguru, tilghman
* /, main/logger.c, utils/extconf.c, main/asterisk.c: Merged
revisions 266142 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r266142 | tilghman | 2010-05-26 16:11:44 -0500 (Wed, 26 May 2010)
| 14 lines Use sigaction for signals which should persist past
the initial trigger, not signal. If you call signal() in a
Solaris signal handler, instead of just resetting the signal
handler, it causes the signal to refire, because the signal is
not marked as handled prior to the signal handler being called.
This effectively causes Solaris to immediately exceed the
threadstack in recursive signal handlers and crash. (closes issue
#17000) Reported by: rmcgilvr Patches:
20100526__issue17000.diff.txt uploaded by tilghman (license 14)
Tested by: rmcgilvr ........
2010-05-26 20:17 +0000 [r266092-266098] Mark Michelson <mmichelson@digium.com>
* apps/app_dial.c: Remove redundant ast_conntected_line_free call.
This wouldn't cause any problems, but it's certainly not needed
either.
* res/res_musiconhold.c: Remove unrelated MOH change from previous
commit. Thanks Kevin!
* main/channel.c, res/res_musiconhold.c: Fix misspelling of macro
args.
2010-05-26 19:46 +0000 [r266006-266090] David Vossel <dvossel@digium.com>
* channels/chan_sip.c, main/app.c, channels/sip/config_parser.c,
channels/sip/include/sip.h: do all sip registry parsing before
transmit_register This patch breaks up every part of the sip
registry string during config parsing and removes all parsing
from transmit_register(). Thanks to Nick_Lewis for contributing
this patch! (closes issue #14331) Reported by: Nick_Lewis
Patches: chan_sip.c-domparse.patch uploaded by Nick Lewis
(license 657) chan_sip.c.patch uploaded by Nick Lewis (license
657) chan_sip.c.domainparse3.patch uploaded by Nick Lewis
(license 657) chan_sip.c-domparse4.patch uploaded by Nick Lewis
(license 657) chan_sip.c-domparse5.patch uploaded by Nick Lewis
(license 657) nicklewispatch.diff uploaded by dvossel (license
671) Tested by: Nick_Lewis, dvossel Review:
https://reviewboard.asterisk.org/r/628/
* channels/chan_sip.c: fixes failed SIP Directed pickup resulting
in dead channel (closes issue #17339) Reported by: one47 Patches:
sip_magic_pickup2 uploaded by one47 (license 23) Tested by:
one47, dvossel
2010-05-26 16:23 +0000 [r265894-265923] Tilghman Lesher <tlesher@digium.com>
* res/res_config_pgsql.c, /: Merged revisions 265910 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r265910 | tilghman | 2010-05-26 11:21:00 -0500 (Wed, 26
May 2010) | 7 lines Not finding rows in the DB does not rise to
the level of a warning. (closes issue #17062) Reported by:
drookie Patches: 20100525__issue17062.diff.txt uploaded by
tilghman (license 14) ........
* res/res_config_pgsql.c, configs/res_pgsql.conf.sample: Construct
socket name, according to the Postgres docs, and document as
such. (closes issue #17392) Reported by: dps Patches:
20100525__issue17392.diff.txt uploaded by tilghman (license 14)
Tested by: dps
2010-05-26 14:45 +0000 [r265842-265844] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: .......
* channels/chan_sip.c: Re-enable "always" option for videosupport
option in sip.conf. (closes issue #17016) Reported by: twilson
Patches: 17016.patch uploaded by mmichelson (license 60) Tested
by: devmod
2010-05-26 05:33 +0000 [r265793] Terry Wilson <twilson@digium.com>
* build_tools/menuselect-deps.in, configure,
include/asterisk/autoconfig.h.in, configure.ac,
res/res_calendar_ews.c: Ensure that libneon > 0.29.0 is installed
for res_calendar_ews This uses a modified version of pabelanger's
patch that checks for NTLM support instead, which was added in
0.29.0 which is what is required for res_calendar_ews. (closes
issue #17391) Reported by: loloski Patches: issue17391.patch.v2
uploaded by pabelanger (license 224) Tested by: twilson
2010-05-26 00:29 +0000 [r265747] Tilghman Lesher <tlesher@digium.com>
* res/res_calendar_exchange.c, res/res_calendar_icalendar.c,
configure, include/asterisk/autoconfig.h.in, configure.ac,
pbx/pbx_lua.c, res/res_calendar_caldav.c, res/res_calendar_ews.c:
Use configure to determine the prefixes and include directories
properly. This ensures cross-platform compatibility, even among
Linux distributions, which don't always put headers in the same
place. (closes issue #17391) Reported by: loloski
2010-05-25 20:59 +0000 [r265698] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Properly use peer's outboundproxy for
outbound REGISTERs. The logic used in transmit_register to get
the outboundproxy for a peer was flawed since this value would be
overridden shortly afterwards when create_addr was called. In
addition, this also fixes some logic used when parsing users.conf
so that the peer name is placed in the internally-generated
register string so that an outboundproxy set in the Asterisk GUI
will be used for outbound REGISTERs.
2010-05-25 17:00 +0000 [r265611] Matthew Nicholson <mnicholson@digium.com>
* /, apps/app_queue.c: Merged revisions 265610 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r265610 | mnicholson | 2010-05-25 11:48:19 -0500 (Tue, 25 May
2010) | 8 lines Don't mark the cdr records of unanswered queue
calls with "NOANSWER". This restores the behavior prior to
r258670. (closes issue #17334) Reported by: jvandal Patches:
queue-cdr-fixes1.diff uploaded by mnicholson (license 96) Tested
by: aragon, jvandal ........
2010-05-25 16:23 +0000 [r265608] Richard Mudgett <rmudgett@digium.com>
* main/channel.c: Memory leak in connected line data when SIP blond
transfer done. The handling of the control subclass
AST_CONTROL_READ_ACTION frame leaked connected line string memory
in __ast_read(). Also in __ast_read() the frame type switch
should not have had a case for AST_CONTROL_READ_ACTION.
AST_CONTROL_READ_ACTION is not a frame type.
2010-05-25 08:31 +0000 [r265525] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
* addons/ooh323c/src/oochannels.c: Typos: 'succesful' (lintian)
2010-05-24 22:21 +0000 [r265467] Terry Wilson <twilson@digium.com>
* doc/manager_1_1.txt, main/manager.c, main/asterisk.c: Merge the
rest of the FullyBooted patch
2010-05-24 22:16 +0000 [r265449-265453] Mark Michelson <mmichelson@digium.com>
* apps/app_senddtmf.c: Allow SendDTMF to play digits to a specified
channel. Patch supplied by reporter was modified to use
autoservice and prevent a potential channel ref leak but is
otherwise as the reporter uploaded it. (closes issue #17182)
Reported by: rcasas Patches: app_senddtmf.c.patch_trunk uploaded
by rcasas (license 641)
* channels/h323/ast_h323.cxx: Print openh323 log to the Asterisk
console. (closes issue #17109) Reported by: under Patches:
logstream.diff uploaded by under (license 914)
* channels/chan_sip.c: Allow type=user SIP endpoints to be loaded
properly from realtime. (closes issue #16021) Reported by:
Guggemand Patches: realtime-type-fix.patch uploaded by Guggemand
(license 897) (altered by me slightly to avoid ref leaks) Tested
by: Guggemand
2010-05-24 20:08 +0000 [r265367] Richard Mudgett <rmudgett@digium.com>
* apps/app_rpt.c: Make app_rpt.c able to compile again.
2010-05-24 19:42 +0000 [r265366] David Vossel <dvossel@digium.com>
* channels/chan_sip.c: reverses incorrect logic introduced by
r243200 The decoding of the replace_id did not need to be broken
up in this instance. This was brought to my attention again
because it caused a segfault when the from or to tags were not
present in the "Replaces" header.
2010-05-24 19:06 +0000 [r265317-265320] Terry Wilson <twilson@digium.com>
* doc/tex/manager.tex: Add the FullyBooted AMI event It is possible
to connect to the manager interface before all Asterisk modules
are loaded. To ensure that an application does not send AMI
actions that might require a module that has not yet loaded, the
application can listen for the FullyBooted manager event. It will
be sent upon connection if all modules have been loaded, or as
soon as loading is complete. The event: Event: FullyBooted
Privilege: system,all Status: Fully Booted Review:
https://reviewboard.asterisk.org/r/639/
* CREDITS, configs/calendar.conf.sample, CHANGES,
res/res_calendar_ews.c (added), res/res_calendar.c: Calendaring
support for Exchange Server 2007+ via EWS This commit adds
support for calendaring with Exchange Server 2007+ via Exchange
Web Services. Full write support and for querying attendees. Many
thanks to Jan Kaláb for the feature. (closes issue #17022)
Reported by: pitel Patches: res_calendar_ews.c uploaded by pitel
(license 1008) Tested by: pitel, twilson Review:
https://reviewboard.asterisk.org/r/557/ Review:
https://reviewboard.asterisk.org/r/668/
2010-05-24 18:19 +0000 [r265316] Tilghman Lesher <tlesher@digium.com>
* main/asterisk.c: On systems with a LOT of RAM, a signed integer
sometimes printed negative. (closes issue #16837) Reported by:
jlpedrosa Patches: 20100504__issue16837.diff.txt uploaded by
tilghman (license 14)
2010-05-24 16:10 +0000 [r265273] David Vossel <dvossel@digium.com>
* main/channel.c: fixes segfault when using generic plc
2010-05-23 18:23 +0000 [r265227] Alexandr Anikin <may@telecom-service.ru>
* addons/chan_ooh323.c: small changes to avoiding 'freeing unused
memory...'
2010-05-21 22:46 +0000 [r265174] Richard Mudgett <rmudgett@digium.com>
* main/channel.c: Channel initialization failure causes crashes.
__ast_channel_alloc_ap() has several points in the initialization
of a new channel structure where it could fail. Since the channel
structure is now an ao2 object, the destructor callback needs to
be able to handle clean up when the structure setup is
incomplete. Problems corrected: 1) Failing to setup the alertpipe
would not unreference the structure but free it directly. Doing
this to an ao2_object is very bad. 2) File descriptors need to be
initialized to -1 before a construction failure could occur so
the destructor will not close unopened descriptors. 3) The
destructor needs to check that the string field has been
initialized before using any string field values. Crashes
expected. 4) The destructor should not notify devstate if the
device name is empty. It is a waste of cycles and a couple ERROR
log messages are generated. Review:
https://reviewboard.asterisk.org/r/675/
2010-05-21 21:08 +0000 [r264953-265090] Mark Michelson <mmichelson@digium.com>
* include/asterisk/file.h, /, apps/app_queue.c: Merged revisions
265089 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r265089 | mmichelson | 2010-05-21 15:59:14 -0500 (Fri, 21 May
2010) | 8 lines Don't hang up on a queue caller if the file we
attempt to play does not exist. This also fixes a documentation
mistake in file.h that made my original attempt to correct this
problem not work correctly. (closes issue #17061) Reported by:
RoadKill ........
* channels/chan_sip.c: Be sure to set the sin_family on the proxy
when allocating. (closes issue #17157) Reported by: stuarth
* /, include/asterisk/channel.h: Merged revisions 264999 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r264999 | mmichelson | 2010-05-21 11:53:53 -0500 (Fri, 21 May
2010) | 3 lines Fix grammatical error in comment. ........
* main/channel.c, main/autoservice.c, /,
include/asterisk/channel.h: Merged revisions 264996 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r264996 | mmichelson | 2010-05-21 11:28:34 -0500 (Fri,
21 May 2010) | 32 lines Allow ast_safe_sleep to defer specific
frames until after the sleep has concluded. From reviewboard
Background: A Digium customer discovered a somewhat odd bug. The
setup is that parties A and B are bridged, and party A places
party B on hold. While party B is listening to hold music, he
mashes a bunch of DTMF. Party A takes party B off hold while this
is happening, but party B continues to hear hold music. I could
reproduce this about 1 in 5 times. The issue: When DTMF features
are enabled and a user presses keys, the channel that the DTMF is
streamed to is placed in an ast_safe_sleep for 100 ms, the
duration of the emulated tone. If an AST_CONTROL_UNHOLD frame is
read from the channel during the sleep, the frame is dropped.
Thus the unhold indication is never made to the channel that was
originally placed on hold. The fix: Originally, I discussed with
Kevin possible ways of fixing the specific problem reported.
However, we determined that the same type of problem could happen
in other situations where ast_safe_sleep() is used. Using
autoservice as a model, I modified ast_safe_sleep_conditional()
to defer specific frame types so they can be re-queued once the
sleep has finished. I made a common function for determining if a
frame should be deferred so that there are not two identical
switch blocks to maintain. Review:
https://reviewboard.asterisk.org/r/674/ ........
* res/res_fax.c, include/asterisk/res_fax.h,
res/res_fax.exports.in, res/res_fax_spandsp.c: Log spandsp's fax
debug output to the FAX logger level. Review:
https://reviewboard.asterisk.org/r/658
2010-05-21 01:00 +0000 [r264905] Terry Wilson <twilson@digium.com>
* channels/chan_sip.c: Take dup'd code for directmedia ACLs and
make utility func The same code was repeated in lots of different
places, so I made a utility fuction for it. This should make the
merge in the v6-new branch easier.
2010-05-20 23:29 +0000 [r264828] Richard Mudgett <rmudgett@digium.com>
* /, main/callerid.c: Merged revisions 264820 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r264820 | rmudgett | 2010-05-20 18:23:21 -0500 (Thu, 20 May 2010)
| 6 lines ast_callerid_parse() had a path that left name
uninitialized. Several callers of ast_callerid_parse() do not
initialize the name parameter before calling thus there is the
potential to use an uninitialized pointer. ........
2010-05-20 22:23 +0000 [r264752-264779] Tilghman Lesher <tlesher@digium.com>
* main/pbx.c: Let ExtensionState resolve dynamic hints. (closes
issue #16623) Reported by: tilghman Patches:
20100116__issue16623.diff.txt uploaded by tilghman (license 14)
Tested by: lmadsen
* apps/app_stack.c: Error message fix. (closes issue #17356)
Reported by: kenner Patches: app_stack.c.diff uploaded by kenner
(license 1040)
2010-05-20 20:49 +0000 [r264669-264711] Richard Mudgett <rmudgett@digium.com>
* main/ccss.c: Avoid crash in generic CC agent init if caller name
or number is NULL.
* apps/app_dial.c, apps/app_queue.c: Dial and queue connected line
update macro not always run when expected. The connected line
update macro would not get run if the connected line number
string was empty. The number could be empty if the connected line
update did not update a number but the name. It should be run if
there was an AST_CONTROL_CONNECTED_LINE frame received for
pending dials and queues. Renamed and added some more comments
for some confusing identifiers directly connected to the related
code. Also fixed a memory leak in app_queue. Review:
https://reviewboard.asterisk.org/r/669/
2010-05-20 17:54 +0000 [r264626] Terry Wilson <twilson@digium.com>
* channels/chan_sip.c, configs/sip.conf.sample, CHANGES,
channels/sip/include/sip.h: Add support for direct media ACLs
directmediapermit/directmediadeny support to restrict which peers
can do directmedia based on ip address. In some networks not all
phones are fully routed, i.e. not all phones can ping each other.
This patch adds a way to restrict directmedia for certain peers
between certain networks. (closes issue #16645) Reported by:
raarts Patches: directmediapermit.patch uploaded by raarts
(license 937) Tested by: raarts Review:
https://reviewboard.asterisk.org/r/467/
2010-05-20 15:30 +0000 [r264497-264540] Kevin P. Fleming <kpfleming@digium.com>
* addons/ooh323c/src/h323, addons/ooh323c/src: Ignore pre-processed
source files generated during DONT_OPTIMIZE dev-mode builds.
* main/logger.c: Correct 'all logger levels' patch to work
properly. Nick Lewis pointed out that the patch as committed
wouldn't actually include dynamic logger levels, which was missed
by the other reviewers. Thanks!
2010-05-19 21:29 +0000 [r264452] Mark Michelson <mmichelson@digium.com>
* main/channel.c, channels/chan_sip.c, include/asterisk/_private.h,
include/asterisk/options.h, main/asterisk.c, main/loader.c: Fix
transcode_via_sln option with SIP calls and improve PLC usage.
From reviewboard: The problem here is a bit complex, so try to
bear with me... It was noticed by a Digium customer that generic
PLC (as configured in codecs.conf) did not appear to actually be
having any sort of benefit when packet loss was introduced on an
RTP stream. I reproduced this issue myself by streaming a file
across an RTP stream and dropping approx. 5% of the RTP packets.
I saw no real difference between when PLC was enabled or disabled
when using wireshark to analyze the RTP streams. After analyzing
what was going on, it became clear that one of the problems faced
was that when running my tests, the translation paths were being
set up in such a way that PLC could not possibly work as
expected. To illustrate, if packets are lost on channel A's read
stream, then we expect that PLC will be applied to channel B's
write stream. The problem is that generic PLC can only be done
when there is a translation path that moves from some codec to
SLINEAR. When I would run my tests, I found that every single
time, read and write translation paths would be set up on channel
A instead of channel B. There appeared to be no real way to
predict which channel the translation paths would be set up on.
This is where Kevin swooped in to let me know about the
transcode_via_sln option in asterisk.conf. It is supposed to work
by placing a read translation path on both channels from the
channel's rawreadformat to SLINEAR. It also will place a write
translation path on both channels from SLINEAR to the channel's
rawwriteformat. Using this option allows one to predictably set
up translation paths on all channels. There are two problems with
this, though. First and foremost, the transcode_via_sln option
did not appear to be working properly when I was placing a SIP
call between two endpoints which did not share any common
formats. Second, even if this option were to work, for PLC to be
applied, there had to be a write translation path that would go
from some format to SLINEAR. It would not work properly if the
starting format of translation was SLINEAR. The one-line change
presented in this review request in chan_sip.c fixed the first
issue for me. The problem was that in sip_request_call, the
jointcapability of the outbound channel was being set to the
format passed to sip_request_call. This is nativeformats of the
inbound channel. Because of this, when
ast_channel_make_compatible was called by app_dial, both channels
already had compatibly read and write formats. Thus, no
translation path was set up at the time. My change is to set the
jointcapability of the sip_pvt created during sip_request_call to
the intersection of the inbound channel's nativeformats and the
configured peer capability that we determined during the earlier
call to create_addr. Doing this got the translation paths set up
as expected when using transcode_via_sln. The changes presented
in channel.c fixed the second issue for me. First and foremost,
when Asterisk is started, we'll read codecs.conf to see the value
of the genericplc option. If this option is set, and ast_write is
called for a frame with no data, then we will attempt to fill in
the missing samples for the frame. The implementation uses a
channel datastore for maintaining the PLC state and for creating
a buffer to store PLC samples in. Even when we receive a frame
with data, we'll call plc_rx so that the PLC state will have
knowledge of the previous voice frame, which it can use as a
basis for when it comes time to actually do a PLC fill-in. So,
reviewers, now I ask for your help. First off, there's the one
line change in chan_sip that I have put in. Is it right? By my
logic it seems correct, but I'm sure someone can tell me why it
is not going to work. This is probably the change I'm least
concerned about, though. What concerns me much more is the set of
changes in channel.c. First off, am I even doing it right? When I
run tests, I can clearly see that when PLC is activated, I see a
significant increase in RTP traffic where I would expect it to
be. However, in my humble opinion, the audio sounds kind of
crappy whenever the PLC fill-in is done. It sounds worse to me
than when no PLC is used at all. I need someone to review the
logic I have used to be sure that I'm not misusing anything. As
far as I can see my pointer arithmetic is correct, and my use of
AST_FRIENDLY_OFFSET should be correct as well, but I'm sure
someone can point out somewhere where I've done something
incorrectly. As I was writing this review request up, I decided
to give the code a test run under valgrind, and I find that for
some reason, calls to plc_rx are causing some invalid reads.
Apparently I'm reading past the end of a buffer somehow. I'll
have to dig around a bit to see why that is the case. If it's
obvious to someone reviewing, speak up! Finally, I have one other
proposal that is not reflected in my code review. Since without
transcode_via_sln set, one cannot predict or control where a
translation path will be up, it seems to me that the current
practice of using PLC only when transcoding to SLINEAR is not
useful. I recommend that once it has been determined that the
method used in this code review is correct and works as expected,
then the code in translate.c that invokes PLC should be removed.
Review: https://reviewboard.asterisk.org/r/622/
2010-05-19 20:30 +0000 [r264400] David Vossel <dvossel@digium.com>
* main/udptl.c: fixes infinite loop during udptl.c's
decode_open_type When decode_length returns the length there is a
check to see if that length is negative, if so the decode loop
breaks as this means the limit has been reached. The problem here
is that length is an unsigned int, so length can never be
negative. This resulted in an infinite loop. (issue #17352)
2010-05-19 20:26 +0000 [r264335-264379] Matthew Nicholson <mnicholson@digium.com>
* main/udptl.c: Cast an unsigned int to a signed int when comparing
it with 0. (AST-377)
* /, apps/app_speech_utils.c: Merged revisions 264334 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r264334 | mnicholson | 2010-05-19 15:01:38 -0500 (Wed,
19 May 2010) | 5 lines Set quieted flag when receiving a dtmf
tone during playback in speechbackground. (closes issue #16966)
Reported by: asackheim ........
2010-05-19 19:21 +0000 [r264331] David Vossel <dvossel@digium.com>
* channels/chan_sip.c: fixes crash in check_rtp_timeout During
deadlock avoidance the sip dialog pvt is locked and unlocked.
When this occurs we have no guarantee the pvt's owner is still
valid. We were trying to access the pvt's owner after this
without checking to see if it still existed first. (closes issue
#17271) Reported by: under Patches: check_rtp_timeout.diff
uploaded by under (license 914) Tested by: dvossel
2010-05-19 17:48 +0000 [r264204-264249] Tilghman Lesher <tlesher@digium.com>
* /, configure, include/asterisk/autoconfig.h.in, configure.ac,
include/asterisk/options.h: Merged revisions 264248 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r264248 | tilghman | 2010-05-19 12:41:29 -0500 (Wed, 19
May 2010) | 17 lines Internal timing is now on by default, if
you're using DAHDI 2.3 or above. The reason for ensuring DAHDI
2.3 or above is that this version ensures that a timer is always
available, whereas in previous versions, it was possible for
DAHDI to be loaded, but have no drivers to actually generate
timing. If internal_timing was turned on in this circumstance, a
complete lack of audio would result. This is the reason why
internal_timing was not on by default. However, now that DAHDI
ensures the availability of a timer, there is no reason for this
setting to be off (and in fact, it solves a great many initial
user problems). (closes issue #15932) Reported by: dimas Patches:
20100519__issue15932.diff.txt uploaded by tilghman (license 14)
Tested by: tilghman ........
* main/dsp.c: Keep track of digit duration, when we're decoding
inband to pass DTMF frames. (closes issue #17235) Reported by:
frawd Patches: new_dtmf_dsp_len.patch uploaded by frawd (license
610) 20100518__issue17235.diff.txt uploaded by tilghman (license
14) Tested by: frawd
2010-05-19 15:39 +0000 [r264161] Leif Madsen <lmadsen@digium.com>
* main/cli.c: Fix compilation problem with previous commit. (issue
#16009)
2010-05-19 15:29 +0000 [r264160] Kevin P. Fleming <kpfleming@digium.com>
* main/logger.c, configs/logger.conf.sample: Add ability for logger
channels to include *all* levels. Now that Asterisk modules can
dynamically create and destroy logger levels on demand, it's
useful to be able to configure a logger channel (console, file,
whatever) to be able to accept log messages from *all* levels,
even levels created dynamically. This patch adds support for
this, by allowing the '*' level name to be used in logger.conf.
Review: https://reviewboard.asterisk.org/r/663/
2010-05-19 15:12 +0000 [r264117] Leif Madsen <lmadsen@digium.com>
* CHANGES, main/cli.c: Add ability to hangup all channels from the
CLI. Added the keyword 'all' to the 'channel hangup request' CLI
command so that you can request all channels to be hungup without
having to restart Asterisk. (closes issue #16009) Reported by:
moy Patches: hangup-all-rev-221688.patch uploaded by moy (license
222) Tested by: moy, russell
2010-05-19 14:38 +0000 [r264114] David Vossel <dvossel@digium.com>
* res/res_rtp_asterisk.c: fixes crash during dtmf During the
processing of Cisco dtmf the dtmf samples were not being
calculated correctly. In an attempt to determine what sample rate
was being used, a NULL frame was processed which caused a crash.
This patch resolves this. (closes issue #17248) Reported by:
falves11 Patches: issue_17248.diff uploaded by dvossel (license
671)
2010-05-19 08:09 +0000 [r264031] Alec L Davis <sivad.a@paradise.net.nz>
* configs/indications.conf.sample: fix incorrectly typed
indications for [nz] stutter and dialrecall (closes issue #17359)
Reported by: alecdavis Patches: bug17359.diff.txt uploaded by
alecdavis (license 585)
2010-05-19 06:41 +0000 [r263905-263950] Tilghman Lesher <tlesher@digium.com>
* /, main/dsp.c: Merged revisions 263949 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r263949 | tilghman | 2010-05-19 01:32:27 -0500 (Wed, 19 May 2010)
| 8 lines Because progress is called multiple times, across
several frames, we must persist states when detecting multitone
sequences. (closes issue #16749) Reported by: dant Patches:
dsp.c-bug16749-1.patch uploaded by dant (license 670) Tested by:
dant ........
* configure, configure.ac, build_tools/sha1sum-sh (added),
makeopts.in, sounds/Makefile: Add an sha1sum-workalike for
platforms which don't have it (like Mac OS X)
2010-05-18 22:48 +0000 [r263904] David Vossel <dvossel@digium.com>
* main/strings.c: fixes segfault on logging (closes issue #17331)
Reported by: under Patches: utils.diff uploaded by under (license
914) segfault_on_logging.diff uploaded by dvossel (license 671)
Tested by: under, dvossel
2010-05-18 21:09 +0000 [r263860] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Be sure to heap-allocate the redirecting to
tag so as not to cause crashiness.
2010-05-18 20:49 +0000 [r263858] Tilghman Lesher <tlesher@digium.com>
* res/res_timing_kqueue.c: Make happy green color come back
2010-05-18 20:09 +0000 [r263810] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Fix memory leaks in redirecting structures
in chan_sip.c Thanks to Richard for pointing this out.
2010-05-18 19:30 +0000 [r263807-263808] Jeff Peeler <jpeeler@digium.com>
* CHANGES: put changes with the correct version
* /, CHANGES, apps/app_directory.c: Merged revisions 263769 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r263769 | jpeeler | 2010-05-18 13:54:58 -0500 (Tue, 18 May 2010)
| 10 lines Modify directory name reading to be interrupted with
operator or pound escape. In the case of accidentally entering
the wrong first three letters for the reading, users could be
very frustrated if the name listing is very long. This allows
interrupting the reading by pressing 0 or #. 0 will attempt to
execute a configured operator (o) extension and # will exit and
proceed in the dialplan. ABE-2200 ........
2010-05-17 23:49 +0000 [r263724] Tilghman Lesher <tlesher@digium.com>
* configure, include/asterisk/autoconfig.h.in, configure.ac,
makeopts.in, sounds/Makefile, autoconf/ast_ext_lib.m4: Cache
sound tarfiles in a common directory, such that a clean reinstall
does not force a re-download of the tarballs. (closes issue
#15370) Reported by: pprindeville Patches:
asterisk-trunk-bugid15370.patch uploaded by pprindeville (license
347) Tested by: pprindeville, tilghman, seanbright
2010-05-17 22:08 +0000 [r263640] Mark Michelson <mmichelson@digium.com>
* /, main/devicestate.c: Merged revisions 263639 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r263639 | mmichelson | 2010-05-17 17:00:28 -0500 (Mon, 17 May
2010) | 10 lines Fix logic error when checking for a devstate
provider. When using strsep, if one of the list of specified
separators is not found, it is the first parameter to strsep
which is now NULL, not the pointer returned by strsep. This issue
isn't especially severe in that the worst it is likely to do is
waste some cycles when a device with no '/' and no ':' is passed
to ast_device_state. ........
2010-05-17 19:31 +0000 [r263589] Tilghman Lesher <tlesher@digium.com>
* apps/app_voicemail.c: With IMAP backend, messages in INBOX were
counted twice for MWI. (closes issue #17135) Reported by:
edhorton Patches: 20100513__issue17135.diff.txt uploaded by
tilghman (license 14) 17135_2.diff uploaded by ebroad (license
878) Tested by: edhorton, ebroad
2010-05-17 15:36 +0000 [r263541] Mark Michelson <mmichelson@digium.com>
* apps/app_dial.c, channels/chan_local.c, main/rtp_engine.c,
channels/chan_sip.c, include/asterisk/channel.h,
configs/misdn.conf.sample, apps/app_queue.c,
funcs/func_redirecting.c, channels/misdn_config.c,
main/channel.c, main/dial.c, channels/chan_dahdi.c,
channels/misdn/isdn_lib.h, channels/chan_misdn.c,
channels/misdn/chan_misdn_config.h, main/features.c,
funcs/func_connectedline.c, include/asterisk/frame.h,
funcs/func_callerid.c, channels/sip/include/sip.h: Enhancements
to connected line and redirecting work. From reviewboard: Digium
has a commercial customer who has made extensive use of the
connected party and redirecting information present in later
versions of Asterisk Business Edition and which is to be in the
upcoming 1.8 release. Through their use of the feature, new
problems and solutions have come about. This patch adds several
enhancements to maximize usage of the connected party and
redirecting information functionality. First, Asterisk trunk
already had connected line interception macros. These macros
allow you to manipulate connected line information before it was
sent out to its target. This patch adds the same feature except
for redirecting information instead. Second, the ast_callerid and
ast_party_id structures have been enhanced to provide a "tag."
This tag can be set with func_callerid, func_connectedline,
func_redirecting, and in the case of DAHDI, mISDN, and SIP
channels, can be set in a configuration file. The idea behind the
callerid tag is that it can be set to whatever value the
administrator likes. Later, when running connected line and
redirecting macros, the admin can read the tag off the
appropriate structure to determine what action to take. You can
think of this sort of like a channel variable, except that
instead of having the variable associated with a channel, the
variable is associated with a specific identity within Asterisk.
Third, app_dial has two new options, s and u. The s option lets a
dialplan writer force a specific caller ID tag to be placed on
the outgoing channel. The u option allows the dialplan writer to
force a specific calling presentation value on the outgoing
channel. Fourth, there is a new control frame subclass called
AST_CONTROL_READ_ACTION added. This was added to correct a very
specific situation. In the case of SIP semi-attended (blond)
transfers, the party being transferred would not have the
opportunity to run a connected line interception macro to
possibly alter the transfer target's connected line information.
The issue here was that during a blond transfer, the SIP transfer
code has no bridged channel on which to queue the connected line
update. The way this was corrected was to add this new control
frame subclass. Now, we queue an AST_CONTROL_READ_ACTION frame on
the channel on which the connected line interception macro should
be run. When ast_read is called to read the frame, ast_read
responds by calling a callback function associated with the
specific read action the control frame describes. In this case,
the action taken is to run the connected line interception macro
on the transferee's channel. Review:
https://reviewboard.asterisk.org/r/652/
2010-05-17 15:14 +0000 [r263375-263460] Leif Madsen <lmadsen@digium.com>
* main/manager.c: Missing newlines added to Set-Cookie line in
manager.c Sean Bright pointed out that we lost a set of newline
characters in commit 190349 on a line I had recently changed. Yay
for code review on commits. (issue #17231, #10961)
* main/manager.c, /: Recorded merge of revisions 263456 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r263456 | lmadsen | 2010-05-17 09:35:18 -0500 (Mon, 17 May 2010)
| 11 lines Manager cookies are not compatible with RFC2109. The
Version field in the cookies we're setting contain quotes around
the version number which is not compatible with RFC2109 and
breaks some implementations. (closes issue #17231) Reported by:
ecarruda Patches: manager_rfc2109-trunk-v1.patch uploaded by
ecarruda (license 559) manager_rfc2109-1.6.2-v1.patch uploaded by
ecarruda (license 559) Tested by: ecarruda, russell ........
* /, sounds/Makefile: Merged revisions 263374 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r263374 | lmadsen | 2010-05-17 09:04:57 -0500 (Mon, 17 May 2010)
| 8 lines Update link to new version of core sounds. The latest
version of the core sounds files 1.4.19 now includes the missing
queue-minute sound file which is called by app_queue but which
has been missing. (closes issue #17123) Reported by: n8ideas
........
2010-05-17 13:05 +0000 [r263294] David Vossel <dvossel@digium.com>
* CHANGES: Update CHANGES to reflect DAHDI buffer dialstring option
backport to 1.6.2
2010-05-16 16:31 +0000 [r263250] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
* contrib/scripts/live_ast: live_ast: add commands 'rsync' and
'gen-live-asterisk' This adds the following two commands to
live_ast: * rsync [user]@host directory Copy over all generated
files to <directory> at remote host. Would allow running live_ast
there. Hence allows separating a build machine from a test
machine. * gen-live-asteris: regenerate live/asterisk . Useful if
copying over files to a different directory.
2010-05-16 11:14 +0000 [r263208] Kevin P. Fleming <kpfleming@digium.com>
* main/astobj2.c: Improve some very confusing structure names in
astobj2.c As pointed out by 'akshayb' on #asterisk-dev, the code
here called a list of bucket entries a 'bucket', and the entries
within the bucket were called 'bucket_list'. This made the code
very hard to understand without reading all of it... so I've
renamed 'bucket_list' to 'bucket_entry' to clarify the purpose of
the structure.
2010-05-14 18:53 +0000 [r263151] David Vossel <dvossel@digium.com>
* channels/chan_iax2.c: fix iax_frame double free Very unfortunate
things happen if we add an iax_frame to the frame queue and let
go of the lock before scheduling the frame's transmit... There is
a race condition that exists where the frame can be removed from
the frame_queue and freed before the transmit is scheduled if we
do not hold on to that lock. This results in a freed frame being
scheduled for transmit later.
2010-05-13 22:01 +0000 [r263069] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c: Fix inverted logic in cli command: ss7 set
debug on/off
2010-05-13 20:25 +0000 [r263028] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
* configure, configure.ac: Remove "untested" feature PRI_VERSION
Nobody seems to actually test PRI_VERSION. It is only useful for
failing PRI support in chan_dahdi.
2010-05-13 17:49 +0000 [r262940-262987] Tilghman Lesher <tlesher@digium.com>
* res/res_timing_kqueue.c: For FreeBSD
* res/res_timing_kqueue.c: Hmmm, probably should have read the
manpage more thoroughly.
2010-05-13 15:36 +0000 [r262895-262897] Russell Bryant <russell@digium.com>
* channels/chan_console.c: Fix an off by one error that causes a
crash. Thanks to Raymond Burke for pointing it out.
* main/stdtime/localtime.c: Fix build on linux.
* pbx/pbx_spool.c: Fix build on linux.
2010-05-13 05:37 +0000 [r262852] Tilghman Lesher <tlesher@digium.com>
* Makefile, pbx/pbx_spool.c, tests/test_time.c,
build_tools/menuselect-deps.in, configure,
include/asterisk/autoconfig.h.in, configure.ac,
main/stdtime/localtime.c, res/res_timing_kqueue.c (added): Add
kqueue(2) implementation to Asterisk in various places. This will
save a considerable amount of CPU on the BSDs, including Mac OS
X, as it eliminates several places in the code that we previously
used a busy loop. Additionally, this adds a res_timing interface,
using kqueue timers. Review:
https://reviewboard.asterisk.org/r/543/
2010-05-12 19:59 +0000 [r262800] Paul Belanger <paul.belanger@polybeacon.com>
* main/loader.c, main/cli.c: Notify CLI when modules is loaded /
unloaded (closes issue #17308) Reported by: pabelanger Patches:
cli.modules.patch uploaded by pabelanger (license 224) Tested by:
pabelanger, russell
2010-05-12 19:53 +0000 [r262796-262798] Leif Madsen <lmadsen@digium.com>
* res/ael/pval.c: Revert previous WARNING message removal.
Marquis42 suggested a better method of doing what I wanted
because I ended up removing the WARNING message for all instances
when really I just wanted to remove it for the 'return' keyword,
not everything. (issue #17145)
* res/ael/pval.c: Remove unnecessary WARNING message in ael/pval.c
(closes issue #17145) Reported by: okrief
2010-05-12 18:01 +0000 [r262744] David Vossel <dvossel@digium.com>
* /, apps/app_meetme.c: Merged revisions 262662 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r262662 | dvossel | 2010-05-12 12:00:04 -0500 (Wed, 12 May 2010)
| 11 lines fixes app_meetme dsp error We attempted to detect
silence after translating a frame from signed linear. This caused
a flooding of errors. To resolve this the code to detect silence
was moved before the translation. (closes issue #17133) Reported
by: jsdyer ........
2010-05-12 17:57 +0000 [r262661-262743] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c: Don't crash when destroying chan_dahdi
pseudo channels. Must do a deep copy of the cc_params in
duplicate_pseudo(). Otherwise, when the duplicate pseudo channel
is destroyed, it frees the original pseudo channel cc_params. The
original pseudo channel is then left with a dangling pointer for
when the next duplicated pseudo channel is created.
* channels/chan_misdn.c: Merged revisions 262657,262660 from
https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
.......... r262660 | rmudgett | 2010-05-12 11:46:47 -0500 (Wed,
12 May 2010) | 4 lines Forgot some conditionals around the
callrerouting facility help text. JIRA ABE-2223 ..........
r262657 | rmudgett | 2010-05-12 11:26:49 -0500 (Wed, 12 May 2010)
| 22 lines Add mISDN Call rerouting facility for point-to-point
ISDN lines (exchange line) In the case of ISDN
point-to-multipoint (multidevice) you can use the mISDN "facility
calldeflect" application for call diversions from external (PSTN)
to external (PSTN). In that case this is the only way to get rid
of the two call legs to the PBX and let the calling number at the
C party become the number of the A party. In the case of ISDN
point-to-point (exchange line) the call deflection facility may
not be used. Instead a call rerouting facility has to be used.
This patch for chan_misdn.c is an extension to realize this
service (facility rerouting application). It can accept either
spelling: "callrerouting" or "callrerouteing". The patch is
tested towards Deutsche Telekom and requires a modified version
of mISDN from Digium, Inc. Patches:
misdn_rerouteing_corrected.patch (Slightly modified.) JIRA
ABE-2223
2010-05-12 16:23 +0000 [r262656] Tilghman Lesher <tlesher@digium.com>
* apps/app_privacy.c: Ensure the arguments are initialized. Also
miscellaneous CG cleanup. (closes issue #16576) Reported by:
uxbod Patches: 20100505__issue16576.diff.txt uploaded by tilghman
(license 14) Tested by: uxbod
2010-05-12 01:00 +0000 [r262613] Paul Belanger <paul.belanger@polybeacon.com>
* channels/chan_sip.c, include/asterisk/cli.h: Convert to
AST_CLI_YESNO and AST_CLI_ONOFF Clean up chan_sip.c to use new
AST_CLI functions (closes issue #17287) Reported by: pabelanger
Patches: issue17287.patch uploaded by pabelanger (license 224)
Tested by: russell
2010-05-11 23:18 +0000 [r262569] Richard Mudgett <rmudgett@digium.com>
* configure, include/asterisk/autoconfig.h.in, configure.ac,
channels/sig_pri.c: Dialing an invalid extension causes
incomplete hangup sequence. Revision -r1489 of the libpri 1.4
branch corrected a deviation from Q.931 Section 5.3.2. However,
this resulted in an unexpected behaviour change to the upper
layer (Asterisk). This change uses pri_hangup_fix_enable() to
follow Q.931 Section 5.3.2 call hangup better if the version of
libpri supports it. (issue #17104) Reported by: shawkris Tested
by: rmudgett
2010-05-11 21:25 +0000 [r262513] Tilghman Lesher <tlesher@digium.com>
* include/asterisk/causes.h: Move cause 200 to cause 26, as
specified in Q.850. Also cleanup the formatting and add a few
more that seem like good candidates. (closes issue #16157)
Reported by: wimpy
2010-05-11 19:57 +0000 [r262422] Jason Parker <jparker@digium.com>
* /, res/Makefile: Merged revisions 262421 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r262421 | qwell | 2010-05-11 14:55:42 -0500 (Tue, 11 May 2010) |
11 lines Use a less silly method for modifying a flex-generated
file. The sed syntax that was used wasn't actually valid, causing
some versions to choke. This is the method that is used in 1.6.x+
for similar changes. (closes issue #16696) Reported by: bklang
Patches: 16696-sedfix.diff uploaded by qwell (license 4) Tested
by: qwell ........
2010-05-11 19:40 +0000 [r262414-262419] Paul Belanger <paul.belanger@polybeacon.com>
* pbx/pbx_config.c: Improve logging by displaying line number
(closes issue #16303) Reported by: dant Patches:
issue16303.patch.v2 uploaded by pabelanger (license 224) Tested
by: dant, lmadsen, pabelanger
* channels/chan_sip.c: Improve logging information for
misconfigured contexts (closes issue #17238) Reported by:
pprindeville Patches: chan_sip-bug17238.patch uploaded by
pprindeville (license 347) Tested by: pprindeville
2010-05-11 17:23 +0000 [r262330] Tilghman Lesher <tlesher@digium.com>
* /, Makefile.rules, apps/app_voicemail.c: Merged revisions 262321
via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r262321 | tilghman | 2010-05-11 12:22:07 -0500 (Tue, 11 May 2010)
| 2 lines Fix issue #17302 a slightly different way (mad props to
Qwell) ........
2010-05-11 16:43 +0000 [r262299] Jason Parker <jparker@digium.com>
* bootstrap.sh: Allow bootstrap script to work on Solaris. As
usual, the way they do things is different, so we need to account
for that. automake is versioned ala BSD/Linux, but autoconf is
not. We don't actually need to specify a version there, since
AC_PREREQ will cover it for us. Things will fail pretty loudly if
AC_PREREQ isn't met. (closes issue #16341) Reported by: bklang
Patches: opensolaris_bootstrap.sh uploaded by bklang (license
919)
2010-05-10 19:06 +0000 [r262236-262240] David Vossel <dvossel@digium.com>
* apps/app_directed_pickup.c: fixes PickupChan application (closes
issue #16863) Reported by: schern Patches:
app_directed_pickup.c.patch uploaded by schern (license 995)
for_trunk.diff uploaded by cjacobsen (license 1029) Tested by:
Graber, cjacobsen, lathama, rickead2000, dvossel
* channels/chan_console.c: fixes crash in chan_console There is a
race condition between console_hangup() and start_stream(). It is
possible for console_hangup() to be called and then the stream
thread to begin after the hangup. To avoid this a check in
start_stream() to make sure the pvt-owner still exists while the
pvt lock is held is made. If the owner is gone that means the
channel hung up and start_stream should be aborted.
2010-05-10 16:36 +0000 [r262152] Tilghman Lesher <tlesher@digium.com>
* /, Makefile.rules: Merged revisions 262151 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r262151 | tilghman | 2010-05-10 11:34:21 -0500 (Mon, 10 May 2010)
| 10 lines Allow compilation on Mac OS X 10.4 (Tiger) (closes
issue #17297) Reported by: jcovert Patches:
20100506__issue17297.diff.txt uploaded by tilghman (license 14)
(closes issue #17302) Reported by: jcovert ........
2010-05-09 02:14 +0000 [r262048-262102] Tilghman Lesher <tlesher@digium.com>
* autoconf/ast_c_define_check.m4, configure,
include/asterisk/autoconfig.h.in, autoconf/ast_ext_lib.m4,
autoconf/ast_c_compile_check.m4: Cleanup a bit more by getting
rid of useless version defines. Also make library detection use
passed CFLAGS. (closes issue #17309) Reported by: stuarth
* configure, configure.ac: Use CPPFLAGS to pass PTHREAD_CFLAGS for
vpb only
2010-05-07 23:54 +0000 [r262005] Alec L Davis <sivad.a@paradise.net.nz>
* UPGRADE.txt, apps/app_voicemail.c: VoicemailMain and
VMauthenticate, allow escape to the 'a' extension when a single
'*' is entered Where a site uses VoicemailMain(mailbox) the users
have to be at their own extension to clear their voicemail, they
have no way of escaping VoicemailMain to allow entry of new
boxnumber. This patch, allows a site to include to 'a' priority
in the VoicemailMain context, to allow an escape. If the 'a'
priority doesn't exist in the context that VoicemailMain was
called from then it acts as the old behaviour. Reported by:
alecdavis Tested by: alecdavis Patch vm_a_extension.diff2.txt
uploaded by alecdavis (license 585) Review:
https://reviewboard.asterisk.org/r/489/
2010-05-07 22:09 +0000 [r261913-261964] Tilghman Lesher <tlesher@digium.com>
* addons/ooh323c/src/ooh323.c: Fix build on Linux
* funcs/func_odbc.c: Double free crash (closes issue #17245)
Reported by: thedavidfactor Patches:
20100426__issue17245.diff.txt uploaded by tilghman (license 14)
Tested by: murraytm
* configure, include/asterisk/autoconfig.h.in, configure.ac: Use
the detected pthread building flags in every place, instead of
hardcoding -lpthread. We nicely detect the right flags on each
system for building Asterisk with pthreads, then ignore it for
every other build option that requires us to build with pthreads.
This caused some items to return a false negative. Also cleanup
some minor naming issues that caused "library library" redundancy
in the output. (closes issue #17303) Reported by: stuarth
Patches: 20100507__issue17303.diff.txt uploaded by tilghman
(license 14) Tested by: stuarth
2010-05-07 16:05 +0000 [r261867] Leif Madsen <lmadsen@digium.com>
* UPGRADE-1.6.txt: Update UPGRADE-1.6.txt stating insecure=very has
been removed. (closes issue #17282) Reported by: stuarth Tested
by: stuarth
2010-05-07 15:33 +0000 [r261866] Jeff Peeler <jpeeler@digium.com>
* channels/sig_pri.c: Fix deadlock in sig_pri when hanging up. The
pri_dchannel thread currently violates locking order by locking
the private and then attempting to queue a frame, which needs to
lock the channel. Queueing a frame is unneccesary though and is
actually a regression since sig_pri. All the places that
currently use ast_softhangup_nolock now will just set the
softhangup value directly as before. (closes issue #17216)
Reported by: lmsteffan Patches: bug17216.patch uploaded by
jpeeler (license 325)
2010-05-06 23:41 +0000 [r261822] Richard Mudgett <rmudgett@digium.com>
* channels/sig_pri.c: Some code optimizations. * Made more places
use pri_queue_control() instead of pri_queue_frame() and a local
frame variable. * Made pri_queue_frame() use
sig_pri_lock_owner(). pri_queue_frame() no longer releases the
libpri access lock unless it is required. * Made the
pri_queue_frame() and pri_queue_control() parameter list similar
to sig_pri_lock_owner().
2010-05-06 20:11 +0000 [r261736] Jeff Peeler <jpeeler@digium.com>
* /, apps/app_voicemail.c: Merged revisions 261735 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r261735 | jpeeler | 2010-05-06 15:10:59 -0500 (Thu, 06
May 2010) | 8 lines Only allow the operator key to be accepted
after leaving a voicemail. Or rather disallow the operator key
from being accepted when not offered, such as after finishing a
recording from within the mailbox options menu. ABE-2121 SWP-1267
........
2010-05-06 17:06 +0000 [r261609] Jason Parker <jparker@digium.com>
* /, sounds/Makefile: Merged revisions 261608 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r261608 | qwell | 2010-05-06 11:56:02 -0500 (Thu, 06 May 2010) |
4 lines Use the versioned MOH tarballs, now that we have them.
This makes for more reproducibility. Prompted by a discussion in
#asterisk-dev ........
2010-05-06 15:39 +0000 [r261560] Tilghman Lesher <tlesher@digium.com>
* channels/sip/include/sip.h: Permit more lines within a SIP body
to be parsed. The example given within the related issue showed
120 lines, which was mostly a result of the body being XML.
(closes issue #17179) Reported by: khw
2010-05-06 14:15 +0000 [r261496-261500] Russell Bryant <russell@digium.com>
* tests/test_heap.c: Add test case for removing random elements
from a heap. I modified the original patch for trunk to use the
unit test API. (issue #17277) Reported by: cappucinoking Patches:
test_heap.diff uploaded by cappucinoking (license 1036) Tested
by: cappucinoking, russell
* main/heap.c: Fix handling of removing nodes from the middle of a
heap. This bug surfaced in 1.6.2 and does not affect code in any
other released version of Asterisk. It manifested itself as SIP
qualify not happening when it should, causing peers to go
unreachable. This was debugged down to scheduler entries
sometimes not getting executed when they were supposed to, which
was in turn caused by an error in the heap code. The problem only
sometimes occurs, and it is due to the logic for removing an
entry in the heap from an arbitrary location (not just popping
off the top). The scheduler performs this operation frequently
when entries are removed before they run (when ast_sched_del() is
used). In a normal pop off of the top of the heap, a node is
taken off the bottom, placed at the top, and then bubbled down
until the max heap property is restored (see max_heapify()). This
same logic was used for removing an arbitrary node from the
middle of the heap. Unfortunately, that logic is full of fail.
This patch fixes that by fully restoring the max heap property
when a node is thrown into the middle of the heap. Instead of
just pushing it down as appropriate, it first pushes it up as
high as it will go, and _then_ pushes it down. Lastly, fix a
minor problem in ast_heap_verify(), which is only used for
debugging. If a parent and child node have the same value, that
is not an error. The only error is if a parent's value is less
than its children. A huge thanks goes out to cappucinoking for
debugging this down to the scheduler, and then producing an
ast_heap test case that demonstrated the breakage. That made it
very easy for me to focus on the heap logic and produce a fix.
Open source projects are awesome. (closes issue #16936) Reported
by: ib2 Tested by: cappucinoking, crjw (closes issue #17277)
Reported by: cappucinoking Patches: heap-fix.rev2.diff uploaded
by russell (license 2) Tested by: cappucinoking, russell
2010-05-06 07:27 +0000 [r261451] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
* channels/chan_dahdi.c: When failing to configure, don't destroy
'cfg' twice Fixes a crash when some config section had an
incorrect channel config.
2010-05-05 22:22 +0000 [r261405] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c: Avoid a crash on SS7 channels.
2010-05-05 20:48 +0000 [r261364] Russell Bryant <russell@digium.com>
* Makefile, configs/asterisk.conf.sample: Restore previous
asterisk.conf syntax, where the directories aren't commented out.
This fixes some breakage in the test suite, that uses the
contents of asterisk.conf to discover the install layout on the
system.
2010-05-05 19:13 +0000 [r261316] David Vossel <dvossel@digium.com>
* channels/chan_sip.c: fixes sip native transfer The Refer-To
header field containing the Replaces header in the URI was not
being decoded properly. This caused invalid parsing between the
caller id field and the domain resulting in a failed transfer.
(closes issue #17284) Reported by: dvossel
2010-05-05 18:43 +0000 [r261314] Paul Belanger <paul.belanger@polybeacon.com>
* /, channels/chan_sip.c: Merged revisions 261274 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r261274 | pabelanger | 2010-05-05 12:42:22 -0400 (Wed, 05 May
2010) | 12 lines Registration fix for SIP realtime. Make sure
realtime fields are not empty. (closes issue #17266) Reported by:
Nick_Lewis Patches: chan_sip.c-realtime.patch uploaded by Nick
Lewis (license 657) Tested by: Nick_Lewis, sberney Review:
https://reviewboard.asterisk.org/r/643/ ........
2010-05-05 18:28 +0000 [r261313] Mark Michelson <mmichelson@digium.com>
* channels/sip/dialplan_functions.c: Prevent unnecessary warnings
when getting rtpsource or rtpdest. If a recognized media type was
present, but the media type was not enabled for the channel, then
a warning would be emitted. For instance, attempting to get
CHANNEL(rtpsource,video) on a call with no video would cause a
warning message to appear. With this change, the warning will
only appear if the stream argument is not recognized as being a
media type that can be specified.
2010-05-05 15:42 +0000 [r261124-261232] Paul Belanger <paul.belanger@polybeacon.com>
* apps/app_queue.c: 'queue reset stats' erroneously clears
wrapuptime configuration. Resets each member's lastcall to 0 now.
(closes issue #17262) Reported by: rain Patches:
wrapuptime_reset_fix.diff uploaded by rain (license 327) Tested
by: rain
* main/manager.c, include/asterisk/cli.h, CHANGES,
include/asterisk/manager.h: New 'manager show settings' CLI
command. See the CHANGES file for more details. (closes issue
#16343) Reported by: pabelanger Patches: issue16343.patch.v5
uploaded by pabelanger (license 224) Tested by: pabelanger,
tilghman, lmadsen Review: https://reviewboard.asterisk.org/r/630/
* Makefile, configs/asterisk.conf.sample (added): New static
asterisk.conf.sample file. This simply moves the functionality
from the Makefile (cleaning it up) into an external
asterisk.conf.samples file. Also updates formatting (easier to
read) and grammar changes to asterisk.conf.samples. (closes issue
#17027) Reported by: pabelanger Patches:
0017027.asterisk.conf.v6.patch uploaded by pabelanger (license
224) Tested by: qwell, lmadsen, pabelanger, chappell Review:
https://reviewboard.asterisk.org/r/616/
2010-05-04 23:51 +0000 [r261095] Tilghman Lesher <tlesher@digium.com>
* main/channel.c, /: Merged revisions 261093-261094 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r261093 | tilghman | 2010-05-04 18:36:53 -0500 (Tue, 04
May 2010) | 7 lines Protect against overflow, when calculating
how long to wait for a frame. (closes issue #17128) Reported by:
under Patches: d.diff uploaded by under (license 914) ........
r261094 | tilghman | 2010-05-04 18:47:08 -0500 (Tue, 04 May 2010)
| 2 lines Add a tiny corner case to the previous commit ........
2010-05-04 22:46 +0000 [r261051] Mark Michelson <mmichelson@digium.com>
* configs/queues.conf.sample, CHANGES, apps/app_queue.c: Add new
possible value to autopause option to allow members to be
autopaused in all queues. See the CHANGES file and
queues.conf.sample for more details. (closes issue #17008)
Reported by: jlpedrosa Patches: queues.autopause_en_review.diff
uploaded by jlpedrosa (license 1002) Review:
https://reviewboard.asterisk.org/r/581/
2010-05-04 21:10 +0000 [r261007] Richard Mudgett <rmudgett@digium.com>
* channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_analog.c,
channels/sig_analog.h, channels/sig_pri.c: The inalarm flag is
not passed up from the sig_analog and sig_pri submodules. The CLI
"dahdi show channel" command was not correctly reporting the
InAlarm status. The inalarm flag is now consistently passed
between chan_dahdi and submodules.
2010-05-04 18:51 +0000 [r260924] Jeff Peeler <jpeeler@digium.com>
* /, apps/app_voicemail.c: Merged revisions 260923 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r260923 | jpeeler | 2010-05-04 13:46:46 -0500 (Tue, 04
May 2010) | 12 lines Voicemail transfer to operator should occur
immediately, not after main menu. There were two scenarios in the
advanced options that while using the operator=yes and review=yes
options, the transfer occurred only after exiting the main menu
(after sending a reply or leaving a message for an extension).
Now after the audio is processed for the reply or message the
transfer occurs immediately as expected. ABE-2107 ABE-2108
........
2010-05-04 15:49 +0000 [r260802] Jason Parker <jparker@digium.com>
* /, build_tools/make_build_h: Merged revisions 260801 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r260801 | qwell | 2010-05-04 10:49:27 -0500 (Tue, 04 May
2010) | 1 line Fix fallout from removing from configure script.
Pointed out by philipp64 on #asterisk-dev ........
2010-05-03 22:13 +0000 [r260757] Jeff Peeler <jpeeler@digium.com>
* apps/app_meetme.c, CHANGES: Add new admin features to meetme:
Roll call, eject all, mute all, record in-conf This patch adds
the following in-conference admin DTMF features: *81 - Roll call
(or simply user count if INTROUSER isn't enabled) *82 - Eject all
non-admins *83 - Mute/unmute all non-admins *84 - Start recording
the conference on the fly FWIW, this code uses newly recorded
prompts. (closes issue #16379) Reported by: rfinnie Patches:
meetme-enhancements-232771-v1.patch uploaded by rfinnie (license
940) modified slightly by me
2010-05-03 17:06 +0000 [r260663] Paul Belanger <paul.belanger@polybeacon.com>
* Makefile, /: Merged revisions 260661-260662 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r260661 | pabelanger | 2010-05-03 12:41:30 -0400 (Mon, 03 May
2010) | 10 lines non-root make install PREFIX=/tmp fails. Prepend
libdir when executing mkpkgconfig allowing non-root installs to
work. (closes issue #17268) Reported by: pabelanger Patches:
issue17268.patch uploaded by pabelanger (license 224) Tested by:
pabelanger ........ r260662 | pabelanger | 2010-05-03 12:54:41
-0400 (Mon, 03 May 2010) | 3 lines Should have removed /usr/lib/
part. Thanks Qwell. ........
2010-05-03 14:58 +0000 [r260570] Leif Madsen <lmadsen@digium.com>
* doc/HOWTO_collect_debug_information.txt: Merged revisions 260569
via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r260569 | lmadsen | 2010-05-03 09:57:39 -0500 (Mon, 03 May 2010)
| 1 line Minor typo pointed out by pabelanger on IRC. ........
2010-05-02 02:52 +0000 [r260521] Eliel C. Sardanons <eliels@gmail.com>
* main/data.c, include/asterisk/data.h: Avoid making AstData depend
on libxml2 to compile. We have some functions inside the AstData
API to get the tree in XML form, but it is not required at the
moment to compile asterisk and we can disable that part of the
API if we don't have libxml2 support.
2010-04-30 22:36 +0000 [r260437] Jeff Peeler <jpeeler@digium.com>
* channels/chan_dahdi.c, channels/sig_analog.c, /,
channels/sig_analog.h: Merged revisions 260434 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r260434 | jpeeler | 2010-04-30 17:22:46 -0500 (Fri, 30 Apr 2010)
| 11 lines Ensure channel state is not incorrectly set in the
case of a very early answer. The needringing bit was being read
in dahdi_read after answering thereby setting the state to
ringing from up. This clears needringing upon answering so that
is no longer possible. (closes issue #17067) Reported by: tzafrir
Patches: needringing.diff uploaded by tzafrir (license 46)
........
2010-04-30 22:24 +0000 [r260435] Richard Mudgett <rmudgett@digium.com>
* channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_pri.c:
Separate the uses of NUM_DCHANS and MAX_CHANNELS into PRI, SS7,
and MFCR2 users. Created SIG_PRI_MAX_CHANNELS, SIG_PRI_NUM_DCHANS
SIG_SS7_MAX_CHANNELS, SIG_SS7_NUM_DCHANS SIG_MFCR2_MAX_CHANNELS
Also fixed the declaration of pollers[] in mfcr2_monitor(). It
was dimensioned to the number of bytes in struct
dahdi_mfcr2.pvts[] and not to the same dimension of the struct
dahdi_mfcr2.pvts[].
2010-04-30 20:11 +0000 [r260344-260346] Mark Michelson <mmichelson@digium.com>
* /, res/res_musiconhold.c: Merged revisions 260345 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r260345 | mmichelson | 2010-04-30 15:08:15 -0500 (Fri,
30 Apr 2010) | 18 lines Fix potential crash from race condition
due to accessing channel data without the channel locked. In
res_musiconhold.c, there are several places where a channel's
stream's existence is checked prior to calling ast_closestream on
it. The issue here is that in several cases, the channel was not
locked while checking the stream. The result was that if two
threads checked the state of the channel's stream at
approximately the same time, then there could be a situation
where both threads attempt to call ast_closestream on the
channel's stream. The result here is that the refcount for the
stream would go below 0, resulting in a crash. I have added
proper channel locking to res_musiconhold.c to ensure that we do
not try to check chan->stream without the channel locked. A
Digium customer has been using this patch for several weeks and
has not had any crashes since applying the patch. ABE-2147
........
* apps/app_queue.c: Fix logic reversal error when queue callers
join the queue. When a specific position is specified for the
queue, the idea was that the caller cannot be placed ahead of
higher-priority callers. Unfortunately, the logic was reversed so
that the caller could ONLY be placed ahead of higher priority
callers. Discovered while writing a unit test.
2010-04-30 06:19 +0000 [r260280-260292] Tilghman Lesher <tlesher@digium.com>
* main/strcompat.c: Don't allow file descriptors to go above 64k,
when we're closing them in a fork(2). This saves time, when, even
though the system allows the process limit to be that high, the
practical limit is much lower. Also introduce an additional
optimization, in the form of using the CLOEXEC flag to close
descriptors at the right time. (closes issue #17223) Reported by:
dbackeberg Patches: 20100423__issue17223.diff.txt uploaded by
tilghman (license 14) Tested by: dbackeberg
* configs/extensions.conf.sample: Logic fixups for a sample FREENUM
dialplan context. (closes issue #17263) Reported by: pprindeville
Patches: freenum-dialplan.patch#3 uploaded by pprindeville
(license 347)
2010-04-29 22:44 +0000 [r260231] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c, channels/sig_analog.c, /: Merged revisions
260195 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r260195 | rmudgett | 2010-04-29 17:11:47 -0500 (Thu, 29 Apr 2010)
| 26 lines DTMF CallerID detection problems. The code handling
DTMF CallerID drops digits on long CallerID numbers and may
timeout waiting for the first ring with shorter numbers. The DTMF
emulation mode was not turned off when processing DTMF CallerID.
When the emulation code gets behind in processing the DTMF digits
it can skip a digit. For shorter numbers, the timeout may have
been too short. I increased it from 2 seconds to 4 seconds. Four
seconds is a typical time between rings for many countries.
(closes issue #16460) Reported by: sum Patches: issue16460.patch
uploaded by rmudgett (license 664) issue16460_v1.6.2.patch
uploaded by rmudgett (license 664) Tested by: sum, rmudgett
Review: https://reviewboard.asterisk.org/r/634/ JIRA SWP-562 JIRA
AST-334 JIRA SWP-901 ........
2010-04-29 18:15 +0000 [r260148] Tilghman Lesher <tlesher@digium.com>
* configs/extensions.conf.sample: Pattern match fail.
2010-04-29 15:33 +0000 [r260050] David Vossel <dvossel@digium.com>
* /, include/asterisk/audiohook.h, main/audiohook.c: Merged
revisions 260049 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r260049 | dvossel | 2010-04-29 10:31:02 -0500 (Thu, 29 Apr 2010)
| 14 lines Fixes crash in audiohook_write_list The middle_frame
in the audiohook_write_list function was being freed if a
audiohook manipulator returned a failure. This is incorrect
logic. This patch resolves this and adds detailed descriptions of
how this function should work and why manipulator failures must
be ignored. (closes issue #17052) Reported by: dvossel Tested by:
dvossel (closes issue #16196) Reported by: atis Review:
https://reviewboard.asterisk.org/r/623/ ........
2010-04-29 00:35 +0000 [r260007] Richard Mudgett <rmudgett@digium.com>
* include/asterisk/extconf.h: Fix comment.
2010-04-28 22:34 +0000 [r259957] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c, channels/sip/include/sip.h: Don't override
peer context with domain context. (closes issue #17040) Reported
by: pprindeville Patches: asterisk-1.6-bugid17040.patch uploaded
by pprindeville (license 347) Tested by: pprindeville Review:
https://reviewboard.asterisk.org/r/565/
2010-04-28 21:20 +0000 [r259870] David Vossel <dvossel@digium.com>
* main/channel.c, channels/chan_local.c, /: Merged revisions 259858
via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r259858 | dvossel | 2010-04-28 16:16:03 -0500 (Wed, 28 Apr 2010)
| 33 lines resolves deadlocks in chan_local Issue_1. In the
local_hangup() 3 locks must be held at the same time... pvt,
pvt->chan, and pvt->owner. Proper deadlock avoidance is done when
the channel to hangup is the outbound chan_local channel, but
when it is not the outbound channel we have an issue... We
attempt to do deadlock avoidance only on the tech pvt, when both
the tech pvt and the pvt->owner are locked coming into that loop.
By never giving up the pvt->owner channel deadlock avoidance is
not entirely possible. This patch resolves that by doing deadlock
avoidance on both the pvt->owner and the pvt when trying to get
the pvt->chan lock. Issue_2. ast_prod() is used in
ast_activate_generator() to queue a frame on the channel and make
the channel's read function get called. This function is used in
ast_activate_generator() while the channel is locked, which
mean's the channel will have a lock both from the generator code
and the frame_queue code by the time it gets to chan_local.c's
local_queue_frame code... local_queue_frame contains some of the
same crazy deadlock avoidance that local_hangup requires, and
this recursive lock prevents that deadlock avoidance from
happening correctly. This patch removes ast_prod() from the
channel lock so only one lock is held during the
local_queue_frame function. (closes issue #17185) Reported by:
schmoozecom Patches: issue_17185_v1.diff uploaded by dvossel
(license 671) issue_17185_v2.diff uploaded by dvossel (license
671) Tested by: schmoozecom, GameGamer43 Review:
https://reviewboard.asterisk.org/r/631/ ........
2010-04-28 21:08 +0000 [r259853] Leif Madsen <lmadsen@digium.com>
* /, config.guess: Merged revisions 259852 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r259852 | lmadsen | 2010-04-28 16:07:48 -0500 (Wed, 28 Apr 2010)
| 6 lines Update config.guess. Updating config.guess because
after installing Ubuntu Server 9.10 and running all the update
scripts, running ./configure would not continue because it was
unable to determine what kind of system I had. After updating
config.guess things started working again. ........
2010-04-28 20:32 +0000 [r259760-259848] Jason Parker <jparker@digium.com>
* /, configure, configure.ac: Merged revisions 259847 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r259847 | qwell | 2010-04-28 15:30:21 -0500 (Wed, 28 Apr
2010) | 1 line Add AC_CONFIG_AUX_DIR to configure script, so
systems without install can use install-sh from our source dir.
........
* /, makeopts.in: Merged revisions 259833 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r259833 | qwell | 2010-04-28 15:25:36 -0500 (Wed, 28 Apr 2010) |
1 line Missed this when removing $ID ........
* Makefile, /, configure, configure.ac: Merged revisions 259748 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r259748 | qwell | 2010-04-28 14:17:38 -0500 (Wed, 28 Apr 2010) |
7 lines Remove usage of `id` since it isn't useful and was
causing breakge. Solaris `id` doesn't support the -u argument.
Instead of figuring out how to fix this to work on Solaris, I
decided to check why it was necessary and where else it was used.
It was only used in one place, and it hasn't been needed for a
very long time (I question whether it was ever needed). ........
2010-04-28 17:18 +0000 [r259672] Jeff Peeler <jpeeler@digium.com>
* /, apps/app_voicemail.c: Merged revisions 259664 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r259664 | jpeeler | 2010-04-28 12:13:29 -0500 (Wed, 28
Apr 2010) | 4 lines Do not play goodbye prompt after timeout of
message review. ABE-2124 ........
2010-04-27 22:47 +0000 [r259587-259617] Jason Parker <jparker@digium.com>
* res/res_agi.c: Fix compile on systems without
HAVE_NULLSAFE_PRINTF defined.
* channels/sip/dialplan_functions.c: Be more explicit about field
naming in a test.
2010-04-27 22:18 +0000 [r259538] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c, /: Merged revisions 259531 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r259531 | rmudgett | 2010-04-27 16:53:07 -0500 (Tue, 27
Apr 2010) | 11 lines DAHDI "WARNING" message is confusing and
vague "WARNING[28406]: chan_dahdi.c:6873 ss_thread: CallerID feed
failed: Success" Changed the warning to "Failed to decode
CallerID on channel 'name'". The message before it is likely more
specific about why the CallerID decode failed. SWP-501 AST-283
........
2010-04-27 22:11 +0000 [r259533] Mark Michelson <mmichelson@digium.com>
* main/ccss.c: Shuffle some casts to make builds on bamboo happier.
2010-04-27 21:49 +0000 [r259527] Leif Madsen <lmadsen@digium.com>
* /, sounds/Makefile: Merged revisions 259526 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r259526 | lmadsen | 2010-04-27 16:48:47 -0500 (Tue, 27 Apr 2010)
| 15 lines Update sounds files. * Add additional sounds prompts
for say_enumeration * Update the English conference sounds
prompts so they are better quality and all sound more consistent
* Clean up the core-sounds-XX.txt and extra-sounds-XX.txt files
to include all present sound files Both core (en, fr, es) and
extra (en, fr) sounds files have been updated. (closes issue
#16200) Reported by: murf (closes issue #17137) Reported by:
lmadsen ........
2010-04-27 21:18 +0000 [r259439-259451] Jason Parker <jparker@digium.com>
* /: Block 259441 instead of recording it as merged.
* /: Recorded merge of revisions 259441 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r259441 | qwell | 2010-04-27 16:15:46 -0500 (Tue, 27 Apr 2010) |
1 line Add gar to the check for AR for those silly OSes (Solaris)
that don't have ar. ........
* main/editline/configure, main/editline/Makefile.in,
main/editline/configure.in: Add gar to the check for AR for those
silly OSes (Solaris) that don't have ar. autoconf2.13 couldn't
handle AC_PROG_GREP, so I removed it. This is fine, since we
don't need to use anything that the configure script doesn't.
2010-04-27 21:10 +0000 [r259438] Leif Madsen <lmadsen@digium.com>
* include/asterisk/doxygen/mantisworkflow.h: Update the Mantis
Workflow document in doxygen. (closes issue #17175) Reported by:
lmadsen Patches: Bug_Tracker_Workflow.v2.txt uploaded by
pabelanger (license 224) Tested by: pabelanger, lmadsen
2010-04-27 19:52 +0000 [r259357] Mark Michelson <mmichelson@digium.com>
* main/ccss.c: Change cc_ref and cc_unref from macros to inline
functions. The hope is that Solaris won't be as whiny after this
change.
2010-04-27 19:31 +0000 [r259353] Jason Parker <jparker@digium.com>
* /, configure, configure.ac: Merged revisions 259352 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r259352 | qwell | 2010-04-27 14:29:26 -0500 (Tue, 27 Apr
2010) | 5 lines Support the silly OSes that don't have ar and
strip. Since AC_PATH_TOOL is equiv to AC_CHECK_TOOL when path
isn't specified, and AC_PATH_TOOLS doesn't exist, we'll just
switch to AC_CHECK_TOOLS. ........
2010-04-27 18:29 +0000 [r259229-259307] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, /: Merged
revisions 259270 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r259270 | rmudgett | 2010-04-27 13:14:54 -0500 (Tue, 27 Apr 2010)
| 14 lines hidecalleridname parameter in chan_dahdi.conf Issue
#7321 implements a new chan_dahdi configuration option. However,
a change mentioned in the issue was never implemented. This is
the change that will allow the feature to work. I added a note to
chan_dahdi.conf.sample about the feature. (closes issue #17143)
Reported by: djensen99 Patches: diff.txt uploaded by djensen99
(license NA) (One line change) Tested by: djensen99 ........
* channels/chan_dahdi.c: Re-fix dahdi_request() iflist locking
since CCSS merged.
2010-04-27 15:25 +0000 [r259189] Tilghman Lesher <tlesher@digium.com>
* contrib/init.d/etc_default_asterisk (added): Add missing file
(pointed out by TheDavidFactor on #asterisk-dev) referenced by
revision 239231.
2010-04-26 21:45 +0000 [r259023-259105] Mark Michelson <mmichelson@digium.com>
* main/channel.c, /: Merged revisions 259104 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r259104 | mmichelson | 2010-04-26 16:44:43 -0500 (Mon, 26 Apr
2010) | 3 lines Let compilation succeed warning-free when
DONT_OPTIMIZE is turned off. ........
* main/channel.c, /: Merged revisions 259018 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r259018 | mmichelson | 2010-04-26 16:03:08 -0500 (Mon, 26 Apr
2010) | 13 lines Prevent Newchannel manager events for dummy
channels. No Newchannel manager event will be fired for channels
that are allocated to not match a registered technology type.
Thus bogus channels allocated solely for variable substitution or
CDR operations do not result in a Newchannel event. (closes issue
#16957) Reported by: atis Review:
https://reviewboard.asterisk.org/r/601 ........
2010-04-26 19:05 +0000 [r258974] David Ruggles <thedavidfactor@gmail.com>
* contrib/valgrind.supp: Line 24 missed in compatibility fix in
revision 233577 added a "fun:" prefix line 24
2010-04-26 15:59 +0000 [r258934] Leif Madsen <lmadsen@digium.com>
* channels/chan_sip.c: Small error in the T.140 RTP port verbose
log. (closes issue #16988) Reported by: frawd Patches:
chan_sip_sdp_verbose_fix.diff uploaded by frawd (license 610)
Tested by: russell
2010-04-26 14:18 +0000 [r258896] Matthew Nicholson <mnicholson@digium.com>
* res/res_fax.c, include/asterisk/res_fax.h, res/res_fax_spandsp.c:
Update res_fax and res_fax_spandsp to be compatible with Fax For
Asterisk 1.2. The fax session initilization code for T.38 faxes
has been rewritten. T.38 session initialization was removed from
generic_fax_exec, and split into two different code paths for
receive and send. Also the 'z' option (to send a T.38 reinvite if
we do not receive one) was added to sendfax. In the output of
'fax show sessions', the 'Type' column has been renamed to 'Tech'
and replaced with a new 'Tech' column that will report 'G.711' or
'T.38'. Control of ECM defaults has been added to res_fax A 'fax
show settings' CLI command has been added. Support of the new
AST_T38_REQUEST_PARMS control method request to handle channels
that have already received a T.38 reinvite before the FAX
application is start has been added. Support for the 'fax show
settings' command has been added to res_fax_spandsp and handling
of the ECM flag has been slightly altered.
2010-04-25 18:51 +0000 [r258838-258855] Alexandr Anikin <may@telecom-service.ru>
* addons/chan_ooh323.c: additional checking related to issue 17186
* addons/chan_ooh323.c: Don't pass zero length callerid to ooh323
stack Don't pass zero callerid string to ooh323 stack because it
can't encode this properly and can't generate setup message.
(closes issue #17186) Reported by: vmikhelson Patches:
zero_callerid_num.patch uploaded by may213 (license 454) Tested
by: may213
2010-04-25 18:12 +0000 [r258776] Tilghman Lesher <tlesher@digium.com>
* /, res/res_monitor.c: Merged revisions 258775 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r258775 | tilghman | 2010-04-25 13:09:05 -0500 (Sun, 25 Apr 2010)
| 6 lines When StopMonitor is called, ensure that it will not be
restarted by a channel event. (closes issue #16590) Reported by:
kkm Patches: resmonitor-16590-trunk.239289.diff uploaded by kkm
(license 888) ........
2010-04-22 22:19 +0000 [r258685] Jason Parker <jparker@digium.com>
* utils/extconf.c: Add another random function that does nothing to
make the utils/ dir happy.
2010-04-22 22:11 +0000 [r258675] Matthew Nicholson <mnicholson@digium.com>
* main/channel.c: Fix previous commit.
2010-04-22 22:10 +0000 [r258673-258674] Jason Parker <jparker@digium.com>
* utils/Makefile, utils/extconf.c: Make utils/ stuff *actually*
compile this time.
* utils/Makefile, utils/extconf.c: Let utils/ dir compile when
DEBUG_THREADS is not enabled.
2010-04-22 21:57 +0000 [r258671] Matthew Nicholson <mnicholson@digium.com>
* main/cdr.c, main/channel.c, /, main/features.c: Merged revisions
193391,258670 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r193391 | mnicholson | 2009-05-08 16:01:25 -0500 (Fri, 08 May
2009) | 8 lines Set the proper disposition on originated calls.
(closes issue #14167) Reported by: jpt Patches:
call-file-missing-cdr2.diff uploaded by mnicholson (license 96)
Tested by: dlotina, rmartinez, mnicholson ........ r258670 |
mnicholson | 2010-04-22 16:49:07 -0500 (Thu, 22 Apr 2010) | 11
lines Fix broken CDR behavior. This change allows a CDR record
previously marked with disposition ANSWERED to be set as BUSY or
NO ANSWER. Additionally this change partially reverts r235635 and
does not set the AST_CDR_FLAG_ORIGINATED flag on CDRs generated
from ast_call(). To preserve proper CDR behavior, the
AST_CDR_FLAG_DIALED flag is now cleared from all brige CDRs in
ast_bridge_call(). (closes issue #16797) Reported by:
VarnishedOtter Tested by: mnicholson ........ (closes issue
#16222) Reported by: telles Tested by: mnicholson
2010-04-22 21:06 +0000 [r258632] Russell Bryant <russell@digium.com>
* tests/test_event.c, main/event.c: Add ast_event subscription unit
test and fix some ast_event API bugs. This patch introduces
another test in test_event.c that exercises most of the
subscription related ast_event API calls. I made some minor
additions to the existing event allocation test to increase API
coverage by the test code. Finally, I made a list in a comment of
API calls not yet touched by the test module as a to-do list for
future test development. During the development of this test
code, I discovered a number of bugs in the event API. 1)
subscriptions to AST_EVENT_ALL were not handled appropriately in
a couple of different places. The API allows a subscription to
all event types, but with IE parameters, just as if it was a
subscription to a specific event type. However, the parameters
were being ignored. This affected ast_event_check_subscriber()
and event distribution to subscribers. 2) Some of the logic in
ast_event_check_subscriber() for checking subscriptions against
query parameters was wrong. Review:
https://reviewboard.asterisk.org/r/617/
2010-04-22 20:04 +0000 [r258595] Eliel C. Sardanons <eliels@gmail.com>
* apps/app_voicemail.c: Pass interactive = 0 and fix a compile
error.
2010-04-22 19:08 +0000 [r258557] Jason Parker <jparker@digium.com>
* main/lock.c (added), include/asterisk/res_odbc.h,
include/asterisk/astobj2.h, main/heap.c, include/asterisk/lock.h,
main/astobj2.c, res/res_odbc.c, include/asterisk/heap.h: Remove
ABI differences that occured when compiling with DEBUG_THREADS.
"Bad Things" would happen if Asterisk was compiled with
DEBUG_THREADS, but a loaded module was not (or vice versa). This
also immensely simplifies the lock code, since there are no
longer 2 separate versions of them. Review:
https://reviewboard.asterisk.org/r/508/
2010-04-22 18:07 +0000 [r258517] Eliel C. Sardanons <eliels@gmail.com>
* doc/manager_1_1.txt, main/channel.c, include/asterisk/doxyref.h,
include/asterisk/xml.h, main/data.c (added), main/xml.c,
include/asterisk/channel.h, include/asterisk/_private.h,
include/asterisk/data.h (added), CHANGES, apps/app_queue.c,
main/asterisk.c, apps/app_voicemail.c: Asterisk data retrieval
API. This module implements an abstraction for retrieving and
exporting asterisk data. Developed by: Brett Bryant
<brettbryant@gmail.com> Eliel C. Sardanons (LU1ALY)
<eliels@gmail.com> For the Google Summer of code 2009 Project.
Documentation can be found in doxygen format and inside the
header include/asterisk/data.h Review:
https://reviewboard.asterisk.org/r/275/
2010-04-22 17:36 +0000 [r258515] Russell Bryant <russell@digium.com>
* doc/tex/channelvariables.tex: Add MEETMEBOOKID from r256019.
2010-04-21 21:56 +0000 [r258433] Jeff Peeler <jpeeler@digium.com>
* /, apps/app_voicemail.c: Merged revisions 258432 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r258432 | jpeeler | 2010-04-21 16:45:36 -0500 (Wed, 21
Apr 2010) | 8 lines Fix looping forever when no input received in
certain voicemail menu scenarios. Specifically, prompting for an
extension (when leaving or forwarding a message) or when
prompting for a digit (when saving a message or changing
folders). ABE-2122 SWP-1268 ........
2010-04-21 19:45 +0000 [r258351-258387] Leif Madsen <lmadsen@digium.com>
* doc/tex/asterisk.tex: Missed this when reverting the bad version
change in asterisk.tex.
* doc/tex/asterisk.tex: Fix change in asterisk.tex that got merged
in after testing. (issue #17220)
* Makefile, doc/tex/security-events.tex, configure,
include/asterisk/autoconfig.h.in, doc/tex/Makefile, configure.ac,
doc/tex/phoneprov.tex, doc/tex, doc/tex/ael.tex,
build_tools/prep_tarball, doc/tex/localchannel.tex,
doc/tex/enum.tex, makeopts.in, doc/tex/asterisk.tex,
doc/tex/cel-doc.tex: Add ability to generate ASCII documentation
from the TeX files. These changes add the ability to run 'make
asterisk.txt' just like the existing 'make asterisk.pdf' commands
to generate a text document from the TeX files we have in the
doc/tex/ directory. I've also updated a few of the .tex files
because they weren't properly escaping certain characters so they
would show up as Unicode characters (like [U+021C]). Made changes
to the configure scripts so it would detect the catdvi program
which is required to convert the .dvi file generated by latex.
I've also added a few lines to the build_tools/prep_tarball
script so that the text documentation gets generated and added to
future tarballs of Asterisk releases. (closes issue #17220)
Reported by: lmadsen Patches: asterisk.txt.patch uploaded by
lmadsen (license 10) asterisk.txt.patch-v4 uploaded by pabelanger
(license 224) Tested by: lmadsen, pabelanger
2010-04-21 19:07 +0000 [r258345] Mark Michelson <mmichelson@digium.com>
* funcs/func_callcompletion.c: Add small documentation update to
func_callcompletion.c. This directs users to documents which can
help explain the concepts and configuration options settable with
the function.
2010-04-21 19:02 +0000 [r258344] Leif Madsen <lmadsen@digium.com>
* UPGRADE.txt, CHANGES, channels/chan_iax2.c: IAXpeers output now
matches SIPpeers format for manager (AMI). (closes issue #17100)
Reported by: secesh Tested by: pabelanger Review:
https://reviewboard.asterisk.org/r/594/
2010-04-21 18:13 +0000 [r258305] David Vossel <dvossel@digium.com>
* channels/chan_sip.c: fixes issue with double "sip:" in header
field This is a clear mistake in logic. Future discussions about
how to avoid having to handle uri's like this should take place
in the future, but this fix needs to go in for now. (closes issue
#15847) Reported by: ebroad Patches: doublesip.patch uploaded by
ebroad (license 878)
2010-04-21 13:26 +0000 [r258265] Leif Madsen <lmadsen@digium.com>
* res/res_calendar_exchange.c, res/res_calendar_icalendar.c,
res/res_calendar_caldav.c: Fix the \brief description in the
res_calendar_*.c files.
2010-04-21 13:24 +0000 [r258190-258256] Julian Lyndon-Smith <julian@dotr.com>
* doc/manager_1_1.txt: fix whitespace issue
* doc/manager_1_1.txt, doc/tex/manager.tex: Added NEW ACTIONS entry
for new MixMonitorMute AMI command. Added State and Direction
variables for new MixMonitorMute AMI command.
* CHANGES: Added CHANGES entry for new MixMonitorMute AMI command.
* main/frame.c, include/asterisk/audiohook.h, main/audiohook.c,
include/asterisk/frame.h, apps/app_mixmonitor.c,
res/res_mutestream.c: Added MixMonitorMute manager command Added
a new manager command to mute/unmute MixMonitor audio on a
channel. Added a new feature to audiohooks so that you can mute
either read / write (or both) types of frames - this allows for
MixMonitor to mute either side of the conversation without
affecting the conversation itself. (closes issue #16740) Reported
by: jmls Review: https://reviewboard.asterisk.org/r/487/
2010-04-20 19:02 +0000 [r258106-258149] Leif Madsen <lmadsen@digium.com>
* configs/cli_aliases.conf.sample: Add 'soft hangup' alias per
Steve Johnson on asterisk-users.
* configs/extensions.conf.sample: Add example dialplan for dialing
ISN numbers (http://www.freenum.org). Minor tweaks and
documentation added by me. (closes issue #17058) Reported by:
pprindeville Patches: freenum.patch#5 uploaded by pprindeville
(license 347) Tested by: lmadsen
* contrib/scripts/sip-friends.sql: Add missing 'useragent' field to
sip-friends.sql file. (closes issue #17171) Reported by: thehar
Patches: sip-friends.patch uploaded by thehar (license 831)
Tested by: pabelanger, thehar
2010-04-20 17:06 +0000 [r258065] Jeff Peeler <jpeeler@digium.com>
* /, apps/app_voicemail.c: Merged revisions 258029 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r258029 | jpeeler | 2010-04-20 11:16:33 -0500 (Tue, 20
Apr 2010) | 11 lines Play correct prompt when voicemail store
failure occurs after attempted forward. If a user's mailbox was
full and a message was attempted to be forwarded to said box,
warnings on the console would indicate failure. However, the
played prompt was that of success (vm-msgsaved). Now storage
failure is taken into account and the correct prompt
(vm-mailboxfull) is played when appropriate. ABE-2123 SWP-1262
........
2010-04-20 12:38 +0000 [r257988] Leif Madsen <lmadsen@digium.com>
* formats/format_pcm.c: Update supported file extensions in
doxygen. Updated the doxygen \arg line after looking at the file
for some other Asterisk documentation and noticing they weren't
up to date. Thanks to seanbright for looking at the code for me
:)
2010-04-19 21:57 +0000 [r257947-257949] Jason Parker <jparker@digium.com>
* main/indications.c: Change log message to match severity.
* main/indications.c: Don't consider a missing indications.conf to
be a critical error. There were many changes in revision 176627
which would avoid the error that a missing config would have
caused. Other than this, there are no other config files
(including asterisk.conf, surprisingly) that are required.
2010-04-19 19:23 +0000 [r257883] Tilghman Lesher <tlesher@digium.com>
* apps/app_voicemail.c: Bad merge fix
2010-04-19 18:42 +0000 [r257851] Mark Michelson <mmichelson@digium.com>
* funcs/func_srv.c: Commit compromise I suggested on review 608.
This allows for multiple SRV queries to be done from the dialplan
for the same service on a single call while still allowing one to
bypass the call to SRVQUERY if they so please. Taking action
since no comments had been left for a while. This can easily be
reverted if needed. External tests still pass.
2010-04-19 17:57 +0000 [r257810] Terry Wilson <twilson@digium.com>
* main/features.c: Fix incomplete CDR merge from r195881 Because
res/res_features.c was removed and main/cdr.c added, these
changes didn't make it to trunk and the 1.6.x branches
2010-04-18 17:25 +0000 [r257768] Tilghman Lesher <tlesher@digium.com>
* configs/cdr_odbc.conf.sample: Removing unused configuration
parameters
2010-04-16 21:22 +0000 [r257713] Dwayne M. Hubbard <dwayne.hubbard@gmail.com>
* /, apps/app_mixmonitor.c: Merged revisions 257686 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r257686 | dhubbard | 2010-04-16 16:15:43 -0500 (Fri, 16
Apr 2010) | 21 lines Make the mixmonitor thread process audio
frames faster Mantis issue 17078 reports MixMonitor recordings
have shorter durations than the call duration. This was because
the mixmonitor thread was not processing frames from the
audiohook fast enough. The mixmonitor thread would slowly fall
behind the most recent audio frame and when the channel hangs up,
the mixmonitor thread would exit without processing the same
number of frames as the channel; leaving the mixmonitor recording
shorter than actual call duration. This revision fixes this issue
by moving the ast_audiohook_trigger_wait() and the subsequent
audiohook.status check into the block where the
ast_audiohook_read_frame() function returns NULL. (closes issue
#17078) Reported by: geoff2010 Patches: dw-M17078.patch uploaded
by dhubbard (license 733) Tested by: dhubbard, geoff2010 Review:
https://reviewboard.asterisk.org/r/611/ ........
2010-04-16 19:50 +0000 [r257646] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Make sure to fail a monitor if we receive a
negative response for a CC SUBSCRIBE.
2010-04-16 19:25 +0000 [r257642] Dwayne M. Hubbard <dwayne.hubbard@gmail.com>
* channels/chan_dahdi.c: Enable PRI SERVICE message support in
chan_dahdi for the 'national' switchtype Revision 1072 of libpri
added SERVICE message support for the 'national' switchtype. The
attached patch enables the use of 'pri service' CLI commands on
dahdi channels that are configured for the 'national' switchtype.
(closes issue #17142) Reported by: dhubbard Patches: dw-ni2.patch
uploaded by dhubbard (license 733) Tested by: elguero, dhubbard
Review: https://reviewboard.asterisk.org/r/612/
2010-04-15 21:26 +0000 [r257493-257560] Tilghman Lesher <tlesher@digium.com>
* include/asterisk/app.h, /, tests/test_app.c, main/app.c: Merged
revisions 257544 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r257544 | tilghman | 2010-04-15 16:23:24 -0500 (Thu, 15 Apr 2010)
| 6 lines Allow application options with arguments to contain
parentheses, through a variety of escaping techniques. Fixes
SWP-1194 (ABE-2143). Review:
https://reviewboard.asterisk.org/r/604/ ........
* /, channels/chan_sip.c: Merged revisions 257467 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r257467 | tilghman | 2010-04-15 15:24:50 -0500 (Thu, 15 Apr 2010)
| 13 lines Don't recreate peer, when responding to a repeated
deregistration attempt. When a reply to a deregistration is lost
in transmit, the client retries the deregistration. Previously,
this would cause a realtime/autocreate peer to be loaded back
into memory, after it had already been correctly purged. Instead,
we just want to resend the reply without loading the peer.
(closes issue #16908) Reported by: kkm Patches:
20100412__issue16908.diff.txt uploaded by tilghman (license 14)
Tested by: kkm ........
2010-04-15 19:41 +0000 [r257343-257427] Leif Madsen <lmadsen@digium.com>
* /, doc/backtrace.txt: Merged revisions 257426 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r257426 | lmadsen | 2010-04-15 14:40:33 -0500 (Thu, 15 Apr 2010)
| 13 lines Update backtrace.txt documentation. Update the
backtrace.txt documentation so it conforms to the same layout as
other documents we've been working on recently. Additionally, add
a bunch of new information about gathering backtraces for crashes
and deadlocks, along with ways of verifying your file before
uploading it. Create a couple of one line commands for people to
generate the files we need. (closes issue #17190) Reported by:
lmadsen Patches: backtrace.txt.patch-2 uploaded by lmadsen
(license 10) Tested by: lmadsen, pabelanger ........
* /, doc/backtrace.txt: Merged revisions 257342 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r257342 | lmadsen | 2010-04-15 08:41:45 -0500 (Thu, 15 Apr 2010)
| 1 line Update address of the bug tracker. ........
2010-04-14 22:57 +0000 [r257262] Tilghman Lesher <tlesher@digium.com>
* main/features.c, configs/features.conf.sample: Yet another issue
where the conversion of the application delimiter to comma caused
an issue. Application arguments within the feature map could
possibly contain a comma, which conflicts with the syntax of the
features.conf configuration file. This patch allows the argument
to be wrapped in parentheses or quoted, to allow the application
arguments to be interpreted as a single configuration parameter.
(closes issue #16646) Reported by: pinga-fogo Patches:
20100414__issue16646.diff.txt uploaded by tilghman (license 14)
Tested by: tilghman Review:
https://reviewboard.asterisk.org/r/547/
2010-04-13 19:17 +0000 [r257191] Tilghman Lesher <tlesher@digium.com>
* channels/chan_sip.c: Also unref the pvt when we delete the
provisional keepalive job. (closes issue #16774) Reported by:
kowalma Patches: 20100315__issue16774.diff.txt uploaded by
tilghman (license 14) Tested by: falves11, jamicque Review:
https://reviewboard.asterisk.org/r/591/
2010-04-13 18:10 +0000 [r257146] Matthew Nicholson <mnicholson@digium.com>
* main/manager.c, /, configs/manager.conf.sample: Merged revisions
257070 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r257070 | mnicholson | 2010-04-13 11:46:30 -0500 (Tue, 13 Apr
2010) | 9 lines Add an option to restore past broken behavor of
the Events manager action Before r238915, certain values for the
EventMask parameter of the Events action would result in no
response being returned. This patch adds an option to restore
that broken behavior. Also while fixing this bug I discovered
that passing an empty EventMasks parameter would also result in
no response being returned, this has been fixed as well while
being preserved when the broken behavior is requested. (closes
issue #17023) Reported by: nblasgen Review:
https://reviewboard.asterisk.org/r/602/ ........
2010-04-13 16:33 +0000 [r257065] Tilghman Lesher <tlesher@digium.com>
* cdr/cdr_sqlite3_custom.c: Ensure that we can have commas within
cdr values. (closes issue #17001) Reported by: snuffy Patches:
20100412__issue17001.diff.txt uploaded by tilghman (license 14)
Tested by: snuffy
2010-04-13 16:18 +0000 [r256985-257032] Mark Michelson <mmichelson@digium.com>
* configs/sip.conf.sample: Update sample dialstrings in
sip.conf.sample file.
* funcs/func_srv.c: Address Russell's comments on func_srv from
reviewboard. * Change copyright date * Place channel in
autoservice when doing SRV lookup * Get rid of trailing
whitespace * Change logic in load_module function
* main/ccss.c: Fix issue where recall would not happen when it
should. Specifically, the situation would happen when multiple
callers would request CC for a single generically-monitored
device. If the monitored device became available but the caller
did not answer the recall, then there was nothing that would poke
the CC core to let it know that it should attempt to recall
someone else instead. After careful consideration, I came to the
conclusion that the only area of Asterisk that needed to be
touched was the generic CC monitor. All other types of CC would
require something outside of Asterisk to invoke a recall for a
separate device. This was accomplished by changing the generic
monitor destructor to poke other generic monitor instances if the
device is currently available and the specific instance was
currently not suspended. In order to not accidentally trigger
recalls at bad times, the fit_for_recall flag was also added to
the generic_monitor_instance_list struct. This gets set as soon
as a monitored device becomes available. It gets cleared if a
CCNR request triggers the creation of a new generic monitor
instance. By doing this, we don't accidentally try to recall a
device when the monitored device was being monitored for CCNR and
never actually became available for recall in the first place.
This error was discovered by Steve Pitts during in-house testing
at Digium.
2010-04-12 17:29 +0000 [r256860-256901] Leif Madsen <lmadsen@digium.com>
* /, doc/HOWTO_collect_debug_information.txt (added): Merged
revisions 256900 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r256900 | lmadsen | 2010-04-12 12:29:26 -0500 (Mon, 12 Apr 2010)
| 15 lines Add How-To document on collecting debugging info for
issues.asterisk.org Paul Belanger has been helping a lot with bug
tracking recently and created this document that we can now point
to when additional debugging information is required. This
document will help those filing issues to know how to get the
information required when filing their issues. This will make
things easier on the developers. Initial text and changes by
pabelanger. Tweaks and editing by myself. (closes issue #17159)
Reported by: pabelanger Patches:
HOWTO_collect_debug_information.txt.patch uploaded by lmadsen
(license 10) Tested by: tzafrir, pabelanger, lmadsen ........
* apps/app_voicemail.c: Remove silly debug message that is not
useful. (issue #17159)
2010-04-12 14:47 +0000 [r256823] David Vossel <dvossel@digium.com>
* channels/chan_sip.c: gives channel reference before unlocking it
and using setvar helper. To guarantee the channel is valid when
calling setvar on the MASTER_CHANNEL dialplan function, a channel
reference must be taken before unlocking. Thanks to russell for
pointing out the error.
2010-04-12 14:39 +0000 [r256821] Leif Madsen <lmadsen@digium.com>
* main/logger.c: CLI command logger set level auto complete. A
simple patch to enable auto tab complete. (closes issue #17152)
Reported by: pabelanger Patches: 0017152.patch uploaded by
pabelanger (license 224)
2010-04-12 02:19 +0000 [r256745-256783] Russell Bryant <russell@digium.com>
* tests/test_substitution.c: test_substitution expects func_curl to
be present to work.
* tests/test_pbx.c: Add ASTERISK_FILE_VERSION() macro
2010-04-10 08:33 +0000 [r256704] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
* contrib/scripts/safe_asterisk.8, doc/asterisk.8,
contrib/scripts/autosupport.8, contrib/scripts/astgenkey.8: fix
hyphen vs. minus in man pages In troff '-' is used for a hyphen.
A minus is denoted by '\-' . This is normally also used for a
dash. This patch converts all '-'-s that are minuses or dashes to
'\-'.
2010-04-09 22:20 +0000 [r256646-256661] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c, main/ccss.c: Remove status_response
callbacks where they are not needed.
* channels/chan_local.c: Prevent crash when originating a call to a
local channel. Call completion code tries to grab the call
completion parameters from the requesting channel during
local_request. When originating a call to a local channel,
however, this channel is NULL. This was causing an issue for me
when trying to run a test script.
2010-04-09 19:46 +0000 [r256569-256608] Richard Mudgett <rmudgett@digium.com>
* doc/CCSS_architecture.pdf (added): Merge CCSS architecture
document from CCSS branch.
* channels/sig_pri.h, configure, include/asterisk/autoconfig.h.in:
Remove PRI CCSS BUGBUG message and update configure script.
2010-04-09 16:04 +0000 [r256485-256530] Mark Michelson <mmichelson@digium.com>
* channels/sip/reqresp_parser.c, channels/sip/include/sip.h,
channels/sip/include/reqresp_parser.h: Add routines for parsing
SIP URIs consistently. From the original issue report opened by
Nick Lewis: Many sip headers in many sip methods contain the ABNF
structure name-andor-addr = name-addr / addr-spec Examples
include the to-header, from-header, contact-header,
replyto-header At the moment chan_sip.c makes various different
attempts to parse this name-andor-addr structure for each header
type and for each sip method with sometimes limited degrees of
success. I recommend that this name-andor-addr structure be
parsed by a dedicated function and that it be used irrespective
of the specific method or header that contains the
name-andor-addr structure Nick has also included unit tests for
verifying these routines as well, so...heck yeah. (closes issue
#16708) Reported by: Nick_Lewis Patches:
reqresp_parser-nameandoraddr2.patch uploaded by Nick Lewis
(license 657 Review: https://reviewboard.asterisk.org/r/549
* channels/chan_sip.c, tests/test_gosub.c, funcs/func_srv.c: Fix
some compiler errors that popped up after the CCSS merge.
* apps/app_dial.c, configs/chan_dahdi.conf.sample,
include/asterisk/devicestate.h, include/asterisk/xml.h,
channels/chan_local.c, doc/tex/ccss.tex (added), main/ccss.c
(added), channels/chan_sip.c, configure.ac, main/xml.c,
include/asterisk/channel.h, configs/manager.conf.sample,
include/asterisk/channelstate.h (added),
include/asterisk/manager.h, CHANGES, channels/sig_pri.c,
channels/sig_pri.h, main/channel.c, channels/chan_dahdi.c,
main/manager.c, funcs/func_callcompletion.c (added),
channels/sig_analog.c, channels/sig_analog.h,
configs/ccss.conf.sample (added), include/asterisk/rtp_engine.h,
include/asterisk/frame.h, include/asterisk/ccss.h (added),
doc/tex/asterisk.tex, main/asterisk.c,
channels/sip/include/sip.h: Merge Call completion support into
trunk. From Reviewboard: CCSS stands for Call Completion
Supplementary Services. An admittedly out-of-date overview of the
architecture can be found in the file doc/CCSS_architecture.pdf
in the CCSS branch. Off the top of my head, the big differences
between what is implemented and what is in the document are as
follows: 1. We did not end up modifying the Hangup application at
all. 2. The document states that a single call completion monitor
may be used across multiple calls to the same device. This proved
to not be such a good idea when implementing protocol-specific
monitors, and so we ended up using one monitor per-device
per-call. 3. There are some configuration options which were
conceived after the document was written. These are documented in
the ccss.conf.sample that is on this review request. For some
basic understanding of terminology used throughout this code, see
the ccss.tex document that is on this review. This implements
CCBS and CCNR in several flavors. First up is a "generic"
implementation, which can work over any channel technology
provided that the channel technology can accurately report device
state. Call completion is requested using the dialplan
application CallCompletionRequest and can be canceled using
CallCompletionCancel. Device state subscriptions are used in
order to monitor the state of called parties. Next, there is a
SIP-specific implementation of call completion. This method uses
the methods outlined in draft-ietf-bliss-call-completion-06 to
implement call completion using SIP signaling. There are a few
things to note here: * The agent/monitor terminology used
throughout Asterisk sometimes is the reverse of what is defined
in the referenced draft. * Implementation of the draft required
support for SIP PUBLISH. I attempted to write this in a
generic-enough fashion such that if someone were to want to write
PUBLISH support for other event packages, such as dialog-state or
presence, most of the effort would be in writing callbacks
specific to the event package. * A subportion of supporting
PUBLISH reception was that we had to implement a PIDF parser. The
PIDF support added is a bit minimal. I first wrote a validation
routine to ensure that the PIDF document is formatted properly.
The rest of the PIDF reading is done in-line in the
call-completion-specific PUBLISH-handling code. In other words,
while there is PIDF support here, it is not in any state where it
could easily be applied to other event packages as is. Finally,
there are a variety of ISDN-related call completion protocols
supported. These were written by Richard Mudgett, and as such I
can't really say much about their implementation. There are notes
in the CHANGES file that indicate the ISDN protocols over which
call completion is supported. Review:
https://reviewboard.asterisk.org/r/523
* main/srv.c, channels/chan_sip.c, funcs/func_srv.c (added),
CHANGES, include/asterisk/srv.h: func_srv and explicit
specification of a remote IP for SIP. From Review Board: There
are two interrelated changes here. First, there is the
introduction of func_srv. This adds two new read-only dialplan
functions, SRVQUERY and SRVRESULT. They work very similarly to
the ENUMQUERY and ENUMRESULT functions, except that this allows
one to query SRV records instead. In order to facilitate this
work, I added a couple of new API calls to srv.h.
ast_srv_get_record_count tells the number of records returned by
an SRV lookup. This number is calculated at the time of the SRV
lookup. ast_srv_get_nth_record allows one to get a numbered SRV
record. Second, there is the modification to chan_sip that allows
one to specify a hostname or IP address (along with a port) to
send an outgoing INVITE to when dialing a SIP peer. This goes
hand-in-hand with func_srv. You can query SRV records and then
use the host and port from the results to dial via a specific
host instead of what is configured in sip.conf. Review:
https://reviewboard.asterisk.org/r/608 SWP-1200
2010-04-08 16:35 +0000 [r256428] Kevin P. Fleming <kpfleming@digium.com>
* /, Makefile.rules, build_tools/make_linker_version_script: Ensure
that linker version scripts (used for symbol export control)
always exist. Using wildcard matching in the Makefile is not
adequate to determine whether an export file should exist for a
module or not, so instead we'll just create one if the module
needs one, or copy the default one if it does not.
2010-04-06 19:28 +0000 [r256370] Tilghman Lesher <tlesher@digium.com>
* configure, include/asterisk/autoconfig.h.in, configure.ac,
include/asterisk/lock.h: Mac OS X does not support comparing a
mutex to its initializer. Create a test for this.
2010-04-06 14:42 +0000 [r256319] David Vossel <dvossel@digium.com>
* channels/chan_sip.c: fixes deadlock in chan_sip caused by usage
of MASTER_CHANNEL dialplan function (closes issue #16767)
Reported by: lmsteffan Patches: deadlock_16767v3.diff uploaded by
dvossel (license 671) Review:
https://reviewboard.asterisk.org/r/606/
2010-04-06 00:39 +0000 [r256265] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c, /: Merged revisions 256225 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r256225 | rmudgett | 2010-04-05 19:10:16 -0500 (Mon, 05
Apr 2010) | 5 lines DAHDI/PRI call to pri_channel_bridge() not
protected by PRI lock. SWP-1231 ABE-2163 ........
2010-04-05 15:14 +0000 [r256161] Leif Madsen <lmadsen@digium.com>
* doc/tex/localchannel.tex: Fix for localchannel.tex to allow PDFs
to be generated again.
2010-04-03 02:12 +0000 [r256103-256104] Richard Mudgett <rmudgett@digium.com>
* apps/app_dial.c, channels/chan_local.c, channels/chan_sip.c,
include/asterisk/channel.h, main/cel.c, channels/sig_pri.c,
channels/chan_iax2.c, apps/app_queue.c, channels/chan_oss.c,
funcs/func_redirecting.c, main/channel.c, main/dial.c,
channels/chan_dahdi.c, channels/chan_misdn.c,
apps/app_dumpchan.c, res/res_agi.c, channels/chan_h323.c,
res/snmp/agent.c, apps/app_amd.c, funcs/func_callerid.c:
Consolidate ast_channel.cid.cid_rdnis into
ast_channel.redirecting.from.number. SWP-1229 ABE-2161 * Ensure
chan_local.c:local_call() will not leak cid.cid_dnid when
copying.
* apps/app_dial.c: Using the Dial application f option when the
call is forwarded will likely crash. Fix app_dial.c:do_forward()
OPT_FORCECLID setting cid.cid_num with a stack allocated string
instead of a heap allocated string.
2010-04-02 23:55 +0000 [r256010-256019] Russell Bryant <russell@digium.com>
* apps/app_meetme.c: Export MEETMEBOOKID and fix pin-less
conferences with realtime conferences (closes issue #16866)
Reported by: DEA Patches: rt-meetme-options.txt uploaded by DEA
(license 3) Tested by: DEA Review:
https://reviewboard.asterisk.org/r/582/
* channels/chan_local.c, /: Merged revisions 256014 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r256014 | russell | 2010-04-02 18:45:56 -0500 (Fri, 02
Apr 2010) | 9 lines Resolve a deadlock that occurs due to a
pointless call to ast_bridged_channel() (closes issue #16840)
Reported by: bzing2 Patches: patch.txt uploaded by bzing2
(license 902) issue_16840.rev1.diff uploaded by russell (license
2) Tested by: bzing2, russell ........
* main/channel.c, /: Merged revisions 256009 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r256009 | russell | 2010-04-02 18:30:15 -0500 (Fri, 02 Apr 2010)
| 2 lines Remove extremely verbose debug message. ........
2010-04-02 20:19 +0000 [r255952] Tilghman Lesher <tlesher@digium.com>
* main/asterisk.c: Pass the PID of the Asterisk process, not the
PID of the canary. (closes issue #17065) Reported by:
globalnetinc Patches: astcanary.patch uploaded by makoto (license
38) Tested by: frawd, globalnetinc
2010-04-02 18:57 +0000 [r255906] Kevin P. Fleming <kpfleming@digium.com>
* res/res_ael_share.exports.in (added), codecs,
res/res_pktccops.exports.in (added), utils,
res/res_monitor.exports.in (added), Makefile.moddir_rules,
res/res_smdi.exports.in (added), Makefile.rules, cdr,
res/res_agi.exports.in (added), formats, main/asterisk.exports
(removed), res/res_odbc.exports (removed),
res/res_calendar.exports (removed), apps/app_voicemail.exports
(removed), bridges, res/res_odbc.exports.in (added),
main/asterisk.exports.in (added), apps/app_voicemail.exports.in
(added), res/res_calendar.exports.in (added),
res/res_features.exports (removed), res/res_fax.exports.in
(added), pbx, res/res_adsi.exports.in (added),
res/res_jabber.exports (removed), res/res_pktccops.exports
(removed), channels, res/res_jabber.exports.in (added),
main/Makefile, res/res_smdi.exports (removed), tests, apps, cel,
res/res_agi.exports (removed), addons, res/res_speech.exports
(removed), Makefile, funcs, res/res_speech.exports.in (added),
res/res_fax.exports (removed), main, res/res_adsi.exports
(removed), res/res_features.exports.in (added),
res/res_ael_share.exports (removed),
build_tools/make_linker_version_script (added), res,
res/res_monitor.exports (removed): Allow symbol export filtering
to work properly on platforms that have symbol prefixes. Some
platforms prefix externally-visible symbols in object files
generated from C sources (most commonly, '_' is the prefix). On
these platforms, the existing symbol export filtering process
ends up suppressing all the symbols that are supposed to be left
visible. This patch allows the prefix string to be supplied to
the top-level Makefile in the LINKER_SYMBOL_PREFIX variable, and
then generates the linker scripts as required to include the
prefix supplied.
2010-04-02 06:45 +0000 [r255850-255851] Michiel van Baak <michiel@vanbaak.info>
* channels/chan_skinny.c: Ignore Redial softkey when no previous
dialed number is known (closes issue #17126) Reported by: wedhorn
Patches: skinny79xx_redial1.diff uploaded by wedhorn (license 30)
* channels/chan_skinny.c: Cleanup transmit_* functions Bulk lot of
generally trivial changes for cleaning up the transmit stuff.
Line state request has been modified for line only responses.
(closes issue #16994) Reported by: wedhorn Patches:
skinny-clean07.diff uploaded by wedhorn (license 30) Tested by:
wedhorn
2010-04-01 18:16 +0000 [r255796] Tilghman Lesher <tlesher@digium.com>
* include/asterisk/lock.h: Fix DEBUG_THREADS build on Darwin.
(closes issue #16828) Reported by: oej Patches:
20100331__issue16828.diff.txt uploaded by tilghman (license 14)
2010-04-01 16:09 +0000 [r255751] Matthew Nicholson <mnicholson@digium.com>
* configs/sip.conf.sample: Removed documentation of the non
existent 'both' option to 'faxdetect' in sip.conf
2010-03-31 22:35 +0000 [r255701] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Fix improper comaparison of anonymous URI
when getting P-Asserted-Identity. There was a bug where we split
the URI on the @ sign and then attempted to compare to
"anonymous@anonymous.invalid" afterwards. This comparison could
never evaluate true. So now we keep a copy of the URI prior to
the split so that the comparison is valid.
2010-03-31 19:13 +0000 [r255592] Tilghman Lesher <tlesher@digium.com>
* /, apps/app_voicemail.c: Recorded merge of revisions 255591 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r255591 | tilghman | 2010-03-31 14:09:46 -0500 (Wed, 31 Mar 2010)
| 15 lines Ensure line terminators in email are consistent. Fixes
an issue with certain Mail Transport Agents, where attachments
are not interpreted correctly. (closes issue #16557) Reported by:
jcovert Patches: 20100308__issue16557__1.4.diff.txt uploaded by
tilghman (license 14) 20100308__issue16557__1.6.0.diff.txt
uploaded by tilghman (license 14)
20100308__issue16557__trunk.diff.txt uploaded by tilghman
(license 14) Tested by: ebroad, zktech Reviewboard:
https://reviewboard.asterisk.org/r/544/ ........
2010-03-31 17:48 +0000 [r255504] Leif Madsen <lmadsen@digium.com>
* apps/app_dial.c, /, configs/sip.conf.sample: Add documentation
clarifying when 't' and 'T' can be used. (closes issue #17021)
Reported by: kovzol Tested by: lmadsen, kovzol, davidw, ebroad
2010-03-30 20:56 +0000 [r255323-255410] Russell Bryant <russell@digium.com>
* /, channels/chan_h323.c: Merged revisions 255409 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r255409 | russell | 2010-03-30 15:56:00 -0500 (Tue, 30
Mar 2010) | 2 lines Don't kill Asterisk if the H323 listener does
not start. ........
* /, pbx/pbx_dundi.c: Merged revisions 255322 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r255322 | russell | 2010-03-30 11:06:06 -0500 (Tue, 30 Mar 2010)
| 2 lines Don't make Asterisk not start if pbx_dundi fails to
initialize. ........
2010-03-29 14:07 +0000 [r255281] Jared Smith <jaredsmith@jaredsmith.net>
* apps/app_confbridge.c, CHANGES: This patch adds custom device
state handling for ConfBridge conferences, matching the devstate
handling of the MeetMe conferences. Review:
https://reviewboard.asterisk.org/r/572/ Closes issue #16972
2010-03-29 05:10 +0000 [r255240] Russell Bryant <russell@digium.com>
* main/event.c: Remove a debugging log entry.
2010-03-27 23:51 +0000 [r255199] Alexandr Anikin <may@telecom-service.ru>
* addons/ooh323c/src/ooh323.c, addons/ooh323c/src/ooGkClient.c,
addons/chan_ooh323.c, addons/ooh323c/src/ooh323.h,
addons/ooh323c/src/ooq931.c, addons/ooh323c/src/ooCalls.c:
corrections in gk interface, small fixes in call clearing.
2010-03-27 14:44 +0000 [r255158] Sean Bright <sean@malleable.com>
* apps/app_voicemail.c: We need to inclde sys/wait.h on OpenBSD to
get WEXITSTATUS.
2010-03-27 06:09 +0000 [r255117] Tilghman Lesher <tlesher@digium.com>
* pbx/pbx_spool.c: inotify support for pbx_spool This should give a
good speed boost, in that one particular thread isn't waking up
once a second to read directory contents. Reviewboard:
https://reviewboard.asterisk.org/r/137/
2010-03-26 19:27 +0000 [r255021-255066] Leif Madsen <lmadsen@digium.com>
* configs/sip.conf.sample: Replace some documentation from 1.6.x
back into trunk. This documentation associated wth tlsbindaddr is
still useful so lets synchronize it between trunk and 1.6.x
branches. (issue #17054)
* configs/sip.conf.sample: Update confusing documentation for
tlsbindaddr. Update some confusing documentation for the
tlsbindaddr option in sip.conf.sample. Point at a link instead
which has better documentation. (closes issue #17054) Reported
by: klaus3000
2010-03-26 16:27 +0000 [r254976] Sean Bright <sean@malleable.com>
* contrib/scripts/live_ast: Work around a bug in dash on Ubuntu by
checking the number of arguments before shift'ing. Reported and
tested by pabelanger.
2010-03-25 23:38 +0000 [r254931] Kevin P. Fleming <kpfleming@digium.com>
* addons/chan_ooh323.h, addons/ooh323c/src/ooasn1.h,
addons/ooh323c/src/ooLogChan.c, addons/ooh323c/src/ooStackCmds.c,
addons/ooh323c/src/errmgmt.c, addons/ooh323c/src/ooTimer.c,
addons/ooh323c/src/dlist.c, addons/ooh323c/src/eventHandler.c,
addons/ooh323c/src/ooCapability.c, addons/ooh323cDriver.c,
addons/mp3/interface.c, addons/ooh323cDriver.h,
addons/ooh323c/src/rtctype.c, addons/ooh323c/src/ooCalls.c,
addons/ooh323c/src/encode.c, addons/ooh323c/src/ooUtils.c,
addons/ooh323c/src/ooGkClient.c, addons/ooh323c/src/ooh323ep.c,
addons/ooh323c/src/ooports.c, addons/mp3/decode_ntom.c,
addons/ooh323c/src/memheap.c, addons/ooh323c/src/ooh323.c,
addons/ooh323c/src/ooh245.c, addons/mp3/common.c,
addons/ooh323c/src/decode.c, addons/ooh323c/src/context.c,
addons/ooh323c/src/perutil.c, addons/mp3/layer3.c,
addons/ooh323c/src/oochannels.c,
addons/ooh323c/src/ooCmdChannel.c,
addons/ooh323c/src/printHandler.c, addons/ooh323c/src/ooq931.c,
addons/ooh323c/src/ootrace.c: Use "local" instead of "system"
header file inclusion. Now that these files are in the tree, they
should prefer the tree's local copy of all Asterisk headers over
any that may be installed.
2010-03-25 21:39 +0000 [r254884] Russell Bryant <russell@digium.com>
* addons/ooh323c/src/ooSocket.c, addons/ooh323c/src/ooSocket.h: Fix
a number of other build problems on Mac OS X.
2010-03-25 20:41 +0000 [r254802] Jason Parker <jparker@digium.com>
* utils/Makefile, /: Merged revisions 254800 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r254800 | qwell | 2010-03-25 15:41:15 -0500 (Thu, 25 Mar 2010) |
1 line Don't remove local copies of utils in uninstall. ........
2010-03-25 20:41 +0000 [r254718-254801] Russell Bryant <russell@digium.com>
* addons/chan_ooh323.h: Resolve compiler warning on FreeBSD.
* addons/ooh323c/src/ooh323.c, addons/Makefile,
addons/ooh323c/src/ooq931.c, addons/ooh323c/src/ootrace.c: Fix
chan_ooh323 so it works on Mac OS X, as well.
* channels/chan_usbradio.c: chan_usbradio depends on alsa.
2010-03-25 18:38 +0000 [r254636-254638] Kevin P. Fleming <kpfleming@digium.com>
* .cleancount: Bump cleancount due to ast_channel change.
* include/asterisk/channel.h: Remove no-longer-used (and unsafe)
field in ast_channel for linked lists. The ast_channel structure
had a field used for linking a channel into a linked list, but
now that ast_channel structures are ao2 objects, this is no
longer needed, and could be harmful as ao2 objects really
shouldn't ever be placed into linked lists (since those lists
don't assist with reference count management on the objects).
* addons/Makefile: Get chan_ooh323 building again after recent
build system changes.
2010-03-25 17:52 +0000 [r254454-254557] Mark Michelson <mmichelson@digium.com>
* tests/test_acl.c (added): Add unit test for testing ACL
functionality. There are two unit tests contained here. 1.
"Invalid ACL" This attempts to read a bunch of badly formatted
ACL entries and add them to a host access rule. The goal of this
test is to be sure that all invalid entries are rejected as they
should be. 2. "ACL" This sets up four ACLs. One is a permit all,
one is a deny all, and the other two have specific rules about
which subnets are allowed and which are not. Then a set of test
addresses is used to determine whether we would allow those
addresses to access us when each ACL is applied. This test, by
the way, was what resulted in AST-2010-003's creation. Review:
https://reviewboard.asterisk.org/r/532
* include/asterisk/acl.h, /: Merged revisions 254552 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r254552 | mmichelson | 2010-03-25 12:33:35 -0500 (Thu,
25 Mar 2010) | 5 lines Add doxygen for acl.h Review:
https://reviewboard.asterisk.org/r/528 ........
* channels/sip/dialplan_functions.c: Add new rtpsource options to
the CHANNEL function. This adds rtpsource options analogous to
the rtpdest functions that already exist. In addition, this fixes
potential crashes which could result due to trying to read values
from nonexistent RTP streams.
* res/res_rtp_asterisk.c, /: Recorded merge of revisions 254452 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r254452 | mmichelson | 2010-03-25 10:59:56 -0500 (Thu, 25 Mar
2010) | 44 lines Several fixes regarding RFC2833 DTMF detection.
Here is a copy and paste of the details from my request on
reviewboard that dealt with these changes: Fix 1. The first
change in place is to fix Mantis issue 15811, which deals with a
situation where Asterisk will incorrectly interpret out of order
RFC2833 frames as duplicate DTMF digits. For instance, we would
receive a sequence like: seqno 1: DTMF 1 seqno 2: DTMF 1 seqno 3:
DTMF 1 seqno 4: DTMF 1 seqno 6: DTMF 1 (end) seqno 5: DTMF 1
seqno 7: DTMF 1 (end) seqno 8: DTMF 1 (end) Prior to this patch
when we received the frame with seqno 5, we would interpret this
as a new DTMF 1. With this patch, we will check the seqno of the
incoming digit and not process the frame if the seqno is lower
than the last recorded seqno. Note that we do not record the
seqno of the dropped DTMF frame for future processing. While the
above situation is what was designed to be fixed, the patch is
written in such a way that the following would also be fixed too:
seqno 9: DTMF 1 seqno 10: DTMF 1 (end) seqno 11: DTMF 1 (end)
seqno 13: DTMF 2 seqno 12: DTMF 1 (end) seqno 14: DTMF 2 seqno
15: DTMF 2 (end) seqno 16: DTMF 2 (end) seqno 17: DTMF 2 (end) In
this second situation, the beginning of the DTMF 2 arrives before
the final end frame of the DTMF 1. With the patch, seqno 12 is no
processed and thus we properly interpret the DTMF. Fix 2. The
second change in place is to fix an issue like the following:
seqno 1: DTMF 1 seqno 2: DTMF 1 seqno 3: DTMF 1 (end) *packet
lost* seqno 4: DTMF 1 (end) *packet lost* seqno 5: DTMF 1 (end)
*packet lost* seqno 6: DTMF 2 When we receive seqno 6, we had
code in place that was supposed to properly end the previously
unended DTMF 1. The problem was that the code was essentially a
no-op. The code would set up an end frame for the DTMF 1 but
would immediately overwrite the frame with the begin for DTMF 2.
I changed process_dtmf_rfc2833() so that instead of returning a
single frame, it is given as an output parameter a list of
frames. Each frame that needs to be returned is appended to this
list. Fix 3. The final change is a minor one where an
AST_CONTROL_SRCCHANGE frame could get lost. If we process a cisco
DTMF or an RFC 3389 frame and no frame was returned, then we
would return &ast_null_frame. The problem is that earlier in the
function, we may have generated an AST_CONTROL_SRCCHANGE frame
and put it in the list of frames we wish to return. This frame
would be lost in such a case. The patch fixes this problem
........
2010-03-25 16:03 +0000 [r254453] Terry Wilson <twilson@digium.com>
* /, main/file.c: Merged revisions 254451 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r254451 | twilson | 2010-03-25 10:57:29 -0500 (Thu, 25 Mar 2010)
| 2 lines Handle new SRCCHANGE control message here too ........
2010-03-25 15:27 +0000 [r254450] Kevin P. Fleming <kpfleming@digium.com>
* main/channel.c, channels/chan_sip.c, res/res_fax.c,
configs/sip.conf.sample, include/asterisk/frame.h,
channels/sip/include/sip.h: Improve handling of T.38 re-INVITEs
that arrive before a T.38-capable application is executing on a
channel. This patch addresses an issue found during working with
end-users using res_fax. If an incoming call is answered in the
dialplan, or jumps to the 'fax' extension due to reception of a
CNG tone (with faxdetect enabled), and then the remote endpoint
sends a T.38 re-INVITE, it is possible for the channel's T.38
state to be 'T38_STATE_NEGOTIATING' when the application starts
up. Unfortunately, even if the application wants to use T.38, it
can't respond to the peer's negotiation request, because the
AST_CONTROL_T38_PARAMETERS control frame that chan_sip sent
originally has been lost, and the application needs the content
of that frame to be able to formulate a reply. This patch adds a
new 'request' type to AST_CONTROL_T38_PARAMETERS,
AST_T38_REQUEST_PARMS. If the application sends this request,
chan_sip will re-send the original control frame (with
AST_T38_REQUEST_NEGOTIATE as the request type), and the
application can respond as normal. If this occurs within the five
second timeout in chan_sip, the automatic cancellation of the
peer reinvite will be stopped, and the application will 'own' the
negotiation process from that point onwards. This also improves
the code path in chan_sip to allow sip_indicate(), when called
for AST_CONTROL_T38_PARAMETERS, to be able to return a non-zero
response, which should have been in place before since the
control frame *can* fail to be processed properly. It also
modifies ast_indicate() to return whatever result the channel
driver returned for this control frame, rather than converting
all non-zero results into '-1'. Finally, the new request type
intentionally returns a positive value, so that an application
that sends AST_T38_REQUEST_PARMS can know for certain whether the
channel driver accepted it and will be replying with a control
frame of its own, or whether it was ignored (if the
sip_indicate()/ast_indicate() path had properly supported failure
responses before, this would not be necessary). This patch also
modifies res_fax to take advantage of the new request. In
addition, this patch makes sip_t38_abort() actually lock the
private structure before doing its work... bad programmer, no
donut. This patch also enhances chan_sip's 'faxdetect' support to
allow triggering on T.38 re-INVITEs received as well as CNG tone
detection. Review: https://reviewboard.asterisk.org/r/556/
2010-03-25 15:21 +0000 [r254446] Leif Madsen <lmadsen@digium.com>
* res/res_agi.c: handle_speechset has 4 arguments. Update code to
reflect that handle_speechset has 4 arguments. (closes issue
#17093) Reported by: gpatri Patches: res_agi.patch uploaded by
gpatri (license 1014) Tested by: pabelanger, mmichelson
2010-03-25 10:09 +0000 [r254406] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
* channels/chan_dahdi.c: remove unneeded explicit channel in dahdi
ioctls This patch removes some cases where the channel number for
an ioctl was passed as a member in a struct rather then through
the file descriptor. The gain setting functions passed around a
channel which is always 0, and thus this parameter is simply
dropped. Review: https://reviewboard.asterisk.org/r/584/
2010-03-24 21:10 +0000 [r254362] Mark Michelson <mmichelson@digium.com>
* main/pbx.c: Fix potential invalid reads that could occur in pbx.c
Here is a cut and paste of my review request for this change:
This past weekend, Russell ran our current suite of unit tests
for Asterisk under valgrind. The PBX pattern match test caused
valgrind to spew forth two invalid read errors. This patch
contains two changes that shut valgrind up and do not cause any
new memory leaks. Change 1: In
ast_context_remove_extension_callerid2, valgrind reported an
invalid read in the for loop close to the function's end.
Specifically, one of the the strcmp calls in the loop control was
reading invalid memory. This was because the caller of
ast_context_remove_extension_callerid2 (__ast_context destroy in
this case) passed as a parameter a shallow copy of an ast_exten's
exten field. This same ast_exten was what was destroyed inside
the for loop, thus any iterations of the for loop beyond the
destruction of the ast_exten would result in invalid reads. My
fix for this is to make a copy of the ast_exten's exten field and
pass the copy to ast_context_remove_extension_callerid2. In
addition, I have also acted similarly with the ast_exten's
matchcid field. Since in this case a NULL is handled quite
differently than an empty string, I needed to be a bit more
careful with its handling. Change 2: In __ast_context_destroy, we
iterated over a hashtab and called
ast_context_remove_extension_callerid2 on each item.
Specifically, the hashtab over which we were iterating was an
ast_exten's peer_table. Inside of
ast_context_remove_extension_callerid2, we could possibly destroy
this ast_exten, which also caused the hashtab to be freed.
Attempting to call ast_hashtab_end_traversal on the hashtab
iterator caused an invalid read to occur when trying to read the
iterator->tab->do_locking field since iterator->tab had already
been freed. My handling of this problem is a bit less
straightforward. With each iteration over the hashtab's contents,
we set a variable called "end_traversal" based on the return of
ast_context_remove_extension_callerid2. If 0 is ever returned,
then we know that the extension was found and destroyed. Because
of this, we cannot call ast_hashtab_end_traversal because we will
be guaranteeing a read of invalid memory. In such a case, we
forego calling ast_hashtab_end_traversal and instead call
ast_free on the hashtab iterator. Review:
https://reviewboard.asterisk.org/r/585
2010-03-24 18:13 +0000 [r254277-254321] Jeff Peeler <jpeeler@digium.com>
* configs/voicemail.conf.sample, CHANGES, apps/app_voicemail.c:
Allow configuration of minsecs and nextaftercmd per mailbox.
Previously only configurable globally. A unit test has also been
written to provide protection against parse failures for
supported mailbox options. (closes issue #16864) Reported by:
kobaz Patches: voicemail2.patch uploaded by kobaz (license 834)
Review: https://reviewboard.asterisk.org/r/555/
* /, res/res_monitor.c: Merged revisions 254235 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r254235 | jpeeler | 2010-03-23 19:37:23 -0500 (Tue, 23 Mar 2010)
| 72 lines Ensure that monitor recordings are written to the
correct location (again) This is an extension to 248860. As such
the dialplan test has been extended: ; non absolute path, not
combined exten => 5040, 1, monitor(wav,tmp/jeff/monitor_test)
exten => 5040, n, dial(sip/5001) ; absolute path, not combined
exten => 5041, 1, monitor(wav,/tmp/jeff/monitor_test2) exten =>
5041, n, dial(sip/5001) ; no path, not combined exten => 5042, 1,
monitor(wav,monitor_test3) exten => 5042, n, dial(sip/5001) ;
combined: changemonitor from non absolute to no path (leaves
tmp/jeff) exten => 5043, 1, monitor(wav,tmp/jeff/monitor_test4,m)
exten => 5043, n, changemonitor(monitor_test5) exten => 5043, n,
dial(sip/5001) ; combined: changemonitor from no path to non
absolute path exten => 5044, 1, monitor(wav,monitor_test6,m)
exten => 5044, n, changemonitor(tmp/jeff/monitor_test7) ; this
wasn't possible before exten => 5044, n, dial(sip/5001) ; non
absolute path, combined exten => 5045, 1,
monitor(wav,tmp/jeff/monitor_test8,m) exten => 5045, n,
dial(sip/5001) ; absolute path, combined exten => 5046, 1,
monitor(wav,/tmp/jeff/monitor_test9,m) exten => 5046, n,
dial(sip/5001) ; no path, combined exten => 5047, 1,
monitor(wav,monitor_test10,m) exten => 5047, n, dial(sip/5001) ;
combined: changemonitor from non absolute to absolute (leaves
tmp/jeff) exten => 5048, 1,
monitor(wav,tmp/jeff/monitor_test11,m) exten => 5048, n,
changemonitor(/tmp/jeff/monitor_test12) exten => 5048, n,
dial(sip/5001) ; combined: changemonitor from absolute to non
absolute (leaves /tmp/jeff) exten => 5049, 1,
monitor(wav,/tmp/jeff/monitor_test13,m) exten => 5049, n,
changemonitor(tmp/jeff/monitor_test14) exten => 5049, n,
dial(sip/5001) ; combined: changemonitor from no path to absolute
exten => 5050, 1, monitor(wav,monitor_test15,m) exten => 5050, n,
changemonitor(/tmp/jeff/monitor_test16) exten => 5050, n,
dial(sip/5001) ; combined: changemonitor from absolute to no path
(leaves /tmp/jeff) exten => 5051, 1,
monitor(wav,/tmp/jeff/monitor_test17,m) exten => 5051, n,
changemonitor(monitor_test18) exten => 5051, n, dial(sip/5001) ;
not combined: changemonitor from non absolute to no path (leaves
tmp/jeff) exten => 5052, 1, monitor(wav,tmp/jeff/monitor_test19)
exten => 5052, n, changemonitor(monitor_test20) exten => 5052, n,
dial(sip/5001) ; not combined: changemonitor from no path to non
absolute exten => 5053, 1, monitor(wav,monitor_test21) exten =>
5053, n, changemonitor(tmp/jeff/monitor_test22) exten => 5053, n,
dial(sip/5001) ; not combined: changemonitor from non absolute to
absolute (leaves tmp/jeff) exten => 5054, 1,
monitor(wav,tmp/jeff/monitor_test23) exten => 5054, n,
changemonitor(/tmp/jeff/monitor_test24) exten => 5054, n,
dial(sip/5001) ; not combined: changemonitor from absolute to non
absolute (leaves /tmp/jeff) exten => 5055, 1,
monitor(wav,/tmp/jeff/monitor_test24) exten => 5055, n,
changemonitor(tmp/jeff/monitor_test25) exten => 5055, n,
dial(sip/5001) ; not combined: changemonitor from no path to
absolute exten => 5056, 1, monitor(wav,monitor_test26) exten =>
5056, n, changemonitor(/tmp/jeff/monitor_test27) exten => 5056,
n, dial(sip/5001) ; not combined: changemonitor from absolute to
no path (leaves /tmp/jeff) exten => 5057, 1,
monitor(wav,/tmp/jeff/monitor_test28) exten => 5057, n,
changemonitor(monitor_test29) exten => 5057, n, dial(sip/5001)
........
2010-03-23 22:48 +0000 [r254162] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
* main/asterisk.c: make 'core show settings' should show all
settable directories (closes issue #17086) Reported by: tzafrir
Patches: asterisk_extra_settings_dirs.diff uploaded by tzafrir
(license 46)
2010-03-23 22:35 +0000 [r254159] Russell Bryant <russell@digium.com>
* main/test.c: Put test output for a failure in a CDATA section in
the XML results.
2010-03-23 21:17 +0000 [r254050] Jeff Peeler <jpeeler@digium.com>
* main/channel.c: Exit native bridging early for greater timing
accuracy with warnings This changes native bridging to break one
millisecond early so that the more accurate timeval calculations
done in the generic bridge can be performed using the bridge
config. Currently the time between exiting native bridging
slightly late can sometimes cause a large enough discrepancy for
warnings to be missed. For the record, 1.4 does not attempt to
native bridge at all when warnings are enabled. (closes issue
#15815) Reported by: adomjan Review:
https://reviewboard.asterisk.org/r/577/
2010-03-23 20:52 +0000 [r254045] Sean Bright <sean@malleable.com>
* apps/app_queue.c: Remove unused structure member in app_queue.
(closes issue #15494) Reported by: makoto
2010-03-23 19:19 +0000 [r254001] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
* tests/Makefile: Change the name of the category 'TEST' to match
the name of the subdir
2010-03-23 16:52 +0000 [r253958] Terry Wilson <twilson@digium.com>
* main/http.c: Don't act like an http write failed when it didn't
fwrite returns the number of items written, not the number of
bytes
2010-03-23 14:22 +0000 [r253917] Kevin P. Fleming <kpfleming@digium.com>
* codecs/Makefile, include/asterisk/logger.h, main/Makefile,
Makefile.moddir_rules, pbx/Makefile, res/Makefile, CHANGES,
channels/Makefile, include/asterisk/options.h, main/cli.c: Change
per-file debug and verbose levels to be per-module, the way users
expect them to work. 'core set debug' and 'core set verbose' can
optionally change the level for a specific filename; however,
this is actually for a specific source file name, not the module
that source file is included in. With examples like chan_sip,
chan_iax2, chan_misdn and others consisting of multiple source
files, this will not lead to the behavior that users expect. If
they want to set the debug level for chan_sip, they want it set
for all of chan_sip, and not to have to also set it for
reqresp_parser and other files that comprise the chan_sip module.
This patch changes this functionality to be module-name based
instead of file-name based. To make this work, some Makefile
modifications were required to ensure that the AST_MODULE
definition is present in each object file produced for each
module as well. Review: https://reviewboard.asterisk.org/r/574/
2010-03-22 20:32 +0000 [r253872] Mark Michelson <mmichelson@digium.com>
* main/asterisk.c: Initialize channels prior to loading "preload"
modules. We can have bad results when a module, upon being
loaded, attempts to reference the channels container if the
container hasn't yet been initialized. I saw this happen by
trying to preload pbx_config.so and having a hint defined which
referenced a non-existent SIP peer.
2010-03-22 19:52 +0000 [r253800] Matthew Nicholson <mnicholson@digium.com>
* /, main/features.c: Merged revisions 253799 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r253799 | mnicholson | 2010-03-22 14:50:00 -0500 (Mon, 22 Mar
2010) | 4 lines Unconditionally copy the caller's account code to
the called party. (related to issue #16331) ........
2010-03-22 19:05 +0000 [r253712-253758] Tilghman Lesher <tlesher@digium.com>
* contrib/scripts/dbsep.cgi: Update query should be an UPDATE, not
a SELECT.
* contrib/scripts/dbsep.cgi: Return the list for later
manipulation. This fixes an issue with the update procedure.
Debugging with mmichelson.
* contrib/scripts/dbsep.cgi, configs/dbsep.conf.sample: Accomodate
equal signs in DSNs and add documentation, based upon
mmichelson's feedback.
2010-03-20 16:50 +0000 [r253536-253579] Russell Bryant <russell@digium.com>
* funcs/func_strings.c: Fix memory corruption found by unit tests.
ast_str_reset() was being called on a potentially uninitialized
pointer. Valgrind is my hero, once again.
* cel/cel_pgsql.c, main/tcptls.c, main/manager.c, main/features.c,
main/test.c, cdr/cdr_pgsql.c, main/stdtime/localtime.c,
main/cel.c: Resolve more compiler warnings on FreeBSD.
* apps/app_voicemail.c: Include sys/wait.h on FreeBSD to get the
WEXITSTATUS() macro.
* apps/app_dial.c, apps/app_followme.c: Resolve compiler warnings
on FreeBSD.
* pbx/pbx_dundi.c: Resolve a compiler warning on FreeBSD.
* channels/chan_dahdi.c: Use SHRT_MAX instead of MAXSHORT. These
changes fix build issues I had with this module on FreeBSD.
2010-03-19 07:37 +0000 [r253490] Alec L Davis <sivad.a@paradise.net.nz>
* main/astobj2.c: prevent segfault if bad magic number is
encountered. internal_ao2_ref uses INTERNAL_OBJ which mzy report
'bad magic number', but internal_ao2_ref continues on, causing
segfault. Although AO2_MAGIC number is checked by INTERNAL_OBJ
before internal_ao2_ref is called, A02_MAGIC is being destroyed
(or a wrong pointer) by the time internal_ao2_ref uses
INTERNAL_OBJ. internal_ao2_ref now returns -1 if INTERNAL_OBJ
encouters a bad magic number. (issue #17037) Reported by:
alecdavis Patches: bug17037.diff.txt uploaded by alecdavis
(license 585) Tested by: alecdavis
2010-03-18 18:23 +0000 [r253357-253378] Russell Bryant <russell@digium.com>
* main/asterisk.c: Update comment to reflect new timeout value.
* main/asterisk.c: Increase CLI command output timeout for asterisk
-rx to 60 seconds. (closes issue #17049) Reported by: russell
Tested by: russell Review:
https://reviewboard.asterisk.org/r/573/
2010-03-18 17:52 +0000 [r253345] Leif Madsen <lmadsen@digium.com>
* apps/app_userevent.c: Change usage of pipe to comma in UserEvent
docs. Change the example usage of pipe as a separator to comma in
the UserEvent documentation. (closes issue #16961) Reported by:
jlpedrosa
2010-03-18 15:59 +0000 [r253261] Philippe Sultan <philippe.sultan@gmail.com>
* res/res_jabber.c: Prevent a crash when a buddy gets offline.
(closes issue #16760) Reported by: fiddur Patches: 248394.diff
uploaded by fiddur (license 678)i with modifications by me Tested
by: fiddur, phsultan
2010-03-18 15:46 +0000 [r253256] Leif Madsen <lmadsen@digium.com>
* /, doc/tex/localchannel.tex: Update to new Local channel
documentation. Add same changes as commit to 1.4, but convert to
TeX. (issue #16963) Reported by: kobaz Patches:
localchannel-2.txt uploaded by kobaz (license 834)
2010-03-18 15:45 +0000 [r253255] Tilghman Lesher <tlesher@digium.com>
* main/stdtime/localtime.c: Just in case of a race, send the signal
on interrupt.
2010-03-17 19:06 +0000 [r253205] Leif Madsen <lmadsen@digium.com>
* main/test.c: main/test.c reports erroneous CLI message. (closes
issue #17051) Reported by: Nick_Lewis
2010-03-17 14:16 +0000 [r253113] Tilghman Lesher <tlesher@digium.com>
* tests/test_gosub.c: Switch to using intptr_t, as suggested by
Kevin Fleming on the -dev list
2010-03-17 00:40 +0000 [r253028-253032] Leif Madsen <lmadsen@digium.com>
* main/xmldoc.c: Fix a typo.
* configs/say.conf.sample: Merged revisions 253018 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r253018 | lmadsen | 2010-03-16 19:26:19 -0500 (Tue, 16
Mar 2010) | 6 lines Add french snipset to say.conf. Add the
french snipset to say.conf. (Closes issue #15799) ........
2010-03-17 00:23 +0000 [r252976-253004] Tilghman Lesher <tlesher@digium.com>
* tests/test_gosub.c: Argh.
* configure, include/asterisk/autoconfig.h.in, tests/test_gosub.c,
configure.ac: Fix bamboo compile error by calculating an integer
with the same size as a pointer.
* tests/test_gosub.c (added), apps/app_stack.c: Mask out previous
arguments on each nested invocation of Gosub. (closes issue
#16758) Reported by: wdoekes Patches:
20100316__issue16758.diff.txt uploaded by tilghman (license 14)
Review: https://reviewboard.asterisk.org/r/561/
2010-03-16 19:36 +0000 [r252849] Russell Bryant <russell@digium.com>
* tests/test_time.c: Re-enable test_time on non-Linux.
2010-03-16 19:36 +0000 [r252848] Sean Bright <sean@malleable.com>
* res/res_clialiases.c: Include an extra newline after "Aliased CLI
command" to get back the prompt. The other issue mentioned in
this bug will be more difficult to resolve since we have no idea
(right now) of knowing if the command that is aliased has been
installed yet. (issue #16978) Reported by: jw-asterisk Tested by:
seanbright
2010-03-16 19:34 +0000 [r252846] Tilghman Lesher <tlesher@digium.com>
* tests/test_time.c, include/asterisk/localtime.h,
main/stdtime/localtime.c: Fix test_time on Mac OS X (and other
platforms without inotify) Reviewboard:
https://reviewboard.asterisk.org/r/554/
2010-03-16 19:01 +0000 [r252767] Russell Bryant <russell@digium.com>
* utils/Makefile, /: Merged revisions 252766 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r252766 | russell | 2010-03-16 14:00:43 -0500 (Tue, 16 Mar 2010)
| 6 lines Don't treat warnings as errors for muted. muted
supports OS X, but uses functions marked as deprecated in 10.6.
However, the functions are still supported, so just ignore the
warnings for now and allow the build to proceed. ........
2010-03-16 18:48 +0000 [r252762] Leif Madsen <lmadsen@digium.com>
* configs/extensions.ael.sample: Merged revisions 252761 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r252761 | lmadsen | 2010-03-16 13:46:20 -0500 (Tue, 16 Mar 2010)
| 7 lines Additional extensions.ael global variable fixes. Fixing
up a couple more overlapping global variable namespaces shared
with extensions.conf.sample. Also noticed a few of the lines that
were commented out didn't have the closing semi-colon so I added
that as well. (issue #17035) ........
2010-03-16 18:40 +0000 [r252760] Tilghman Lesher <tlesher@digium.com>
* codecs/gsm/Makefile: OSARCH is not inherited to this directory
2010-03-16 18:36 +0000 [r252759] Russell Bryant <russell@digium.com>
* tests/test_time.c: Disable this test on non-Linux for now.
2010-03-15 22:48 +0000 [r252709] Kevin P. Fleming <kpfleming@digium.com>
* res/res_fax.c: Improve handling of values supplied to
FAXOPT(ecm). Previously, values that began with whitespace were
silently treated as 'no', and all non-'yes' values were also
treated as 'no'. Now the supplied value is specifically checked
for a 'yes' or 'no' (or equivalent) value, after skipping leading
whitespace. If the value is not valid, then a warning message is
generated.
2010-03-15 22:14 +0000 [r252627] Russell Bryant <russell@digium.com>
* channels/chan_sip.c: Tell the RTP engine API about the initial
read and write format. Peer reviewed out-of-band by file.
2010-03-15 21:55 +0000 [r252623] Sean Bright <sean@malleable.com>
* apps/app_meetme.c: Resolve a crash in SLATrunk when the specified
trunk doesn't exist. Reported by philipp64 in #asterisk-dev.
2010-03-15 21:51 +0000 [r252619] Tilghman Lesher <tlesher@digium.com>
* contrib/init.d/org.asterisk.asterisk.plist, /: Merged revisions
252617 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r252617 | tilghman | 2010-03-15 16:43:14 -0500 (Mon, 15 Mar 2010)
| 2 lines Uh, yeah. Umask. I'm stupid. ........
2010-03-15 20:52 +0000 [r252534] Leif Madsen <lmadsen@digium.com>
* /, configs/extensions.ael.sample: Merged revisions 252533 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r252533 | lmadsen | 2010-03-15 15:48:56 -0500 (Mon, 15 Mar 2010)
| 7 lines Update extensions.ael file to not overlap
extensions.conf. Updated the extensions.ael file so the global
variables don't overlap those that we have in extensions.conf
(sample files). This way unexpected things won't happed hopefully
if both pbx_ael and res_config are loaded. (closes issue #17035)
Reported by: pprindeville ........
2010-03-15 16:27 +0000 [r252362-252488] Tilghman Lesher <tlesher@digium.com>
* codecs/gsm/Makefile: Make the Makefile logic more explicit and
move the Snow Leopard logic down to where it's not executed on
non-Darwin systems. (closes issue #17028) Reported by: pabelanger
Patches: issue17028_20100315.patch uploaded by seanbright
(license 71) 20100315__issue17028.diff.txt uploaded by tilghman
(license 14) Tested by: tilghman, pabelanger
* channels/chan_sip.c: THIS IS NOT PYTHON. Indentation doesn't
matter, only braces do. (closes issue #17025) Reported by:
smurfix Patches: sip.patch uploaded by smurfix (license 547)
* /: Recorded merge of revisions 252366 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r252366 | tilghman | 2010-03-14 20:39:00 -0500 (Sun, 14 Mar 2010)
| 2 lines Typo ........
* Makefile, contrib/init.d/org.asterisk.asterisk.plist (added), /,
main/asterisk.c: Merged revisions 252361 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r252361 | tilghman | 2010-03-14 20:33:50 -0500 (Sun, 14 Mar 2010)
| 4 lines Launch Asterisk on Mac OS X with launchd. Reviewboard:
https://reviewboard.asterisk.org/r/551/ ........
2010-03-14 17:43 +0000 [r252314] Sean Bright <sean@malleable.com>
* cdr/cdr_sqlite3_custom.c, cel/cel_sqlite3_custom.c: Fix building
CDR and CEL SQLite3 modules. They added a sqlite3_log() function
which was conflicting with our function names. (closes issue
#17017) Reported by: alephlg
2010-03-14 14:42 +0000 [r252277] Alexandr Anikin <may@telecom-service.ru>
* addons/ooh323c/src/ooh323.c, addons/chan_ooh323.c,
addons/ooh323c/src/ooh245.c, addons/ooh323c/src/ooCalls.h,
configs/chan_ooh323.conf.sample, addons/ooh323c/src/ooh245.h,
addons/ooh323c/src/ooSocket.c, addons/ooh323c/src/ootypes.h,
addons/ooh323c/src/ooq931.c: generate roundtrip delay requests
and responses added response to roundtrip delay requests from
opposite side added roundtrip delay request sending to opposite
side after answer, added options for sending request (interval
between request and count of unreplied requests before forced
call hangup) (closes issue #16976) Reported by: vmikhelson
Patches: rtdr-1.6.0-2.patch uploaded by may213 (license 454)
Tested by: vmikhelson, may213
2010-03-13 22:21 +0000 [r252229-252241] Russell Bryant <russell@digium.com>
* main/app.c: Resolve unit test failure that occurred on Mac OSX.
On Linux (glibc), regcomp() does not return an error for an empty
string. However, the version on OSX will return an error. The
test for channel group matching by regex now passes on the mac,
as well.
* tests/test_time.c: Resolve compiler warning by paying attention
to system() return value. This resolves the last compile failure
on bamboo.
2010-03-12 23:18 +0000 [r252133] Tilghman Lesher <tlesher@digium.com>
* tests/test_time.c (added): Test script to verify that timezone
cache is properly removed on zonefile alteration.
2010-03-12 22:04 +0000 [r252089] Terry Wilson <twilson@digium.com>
* main/channel.c, res/res_rtp_asterisk.c, addons/chan_ooh323.c,
main/rtp_engine.c, channels/chan_sip.c, channels/chan_skinny.c,
channels/chan_h323.c, configs/sip.conf.sample,
include/asterisk/frame.h, include/asterisk/rtp_engine.h,
channels/sip/include/sip.h, channels/chan_mgcp.c: Only change the
RTP ssrc when we see that it has changed This change basically
reverts the change reviewed in
https://reviewboard.asterisk.org/r/374/ and instead limits the
updating of the RTP synchronization source to only those times
when we detect that the other side of the conversation has
changed the ssrc. The problem is that SRCUPDATE control frames
are sent many times where we don't want a new ssrc, including
whenever Asterisk has to send DTMF in a normal bridge. This is
also not the first time that this mistake has been made. The
initial implementation of the ast_rtp_new_source function also
changed the ssrc--and then it was removed because of this same
issue. Then, we put it back in again to fix a different issue.
This patch attempts to only change the ssrc when we see that the
other side of the conversation has changed the ssrc. It also
renames some functions to make their purpose more clear. Review:
https://reviewboard.asterisk.org/r/540/
2010-03-12 21:57 +0000 [r252088] Moises Silva <moises.silva@gmail.com>
* channels/chan_dahdi.c: add missing mfcr2_skip_category setting
2010-03-12 19:43 +0000 [r251989] Tilghman Lesher <tlesher@digium.com>
* apps/app_voicemail.c: Don't override a user option with the
global option. (closes issue #16849) Reported by: ip-rob Patches:
20100311__issue16849.diff.txt uploaded by tilghman (license 14)
Tested by: ip-rob
2010-03-12 19:40 +0000 [r251946-251987] Richard Mudgett <rmudgett@digium.com>
* /: Merged revisions 251986 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r251986 | rmudgett | 2010-03-12 13:33:22 -0600 (Fri, 12 Mar 2010)
| 1 line Make chan_dahdi wakeup_sub() prototype not conditional.
........
* channels/chan_dahdi.c: Doxegen this chan_dahdi lock.
2010-03-11 21:07 +0000 [r251877-251884] Tilghman Lesher <tlesher@digium.com>
* apps/app_exec.c: Because ExecIf needs to reprocess arguments,
it's best if we don't remove quotes during parsing. (closes issue
#16905) Reported by: ip-rob Patches:
20100303__issue16905.diff.txt uploaded by tilghman (license 14)
Tested by: ip-rob
* tests/test_stringfields.c: Fix tests on 32-bit systems.
* apps/app_system.c: If the argument to the system application is
quoted, ensure we remove the quotes before trying to execute.
(closes issue #16842) Reported by: ip-rob Patches:
20100310__issue16842.diff.txt uploaded by tilghman (license 14)
Tested by: ip-rob
2010-03-11 18:07 +0000 [r251821] Richard Mudgett <rmudgett@digium.com>
* channels/sig_pri.h, channels/chan_dahdi.c: Minor tweaks and
comment updates to chan_dahdi.
2010-03-11 07:03 +0000 [r251779] Alec L Davis <sivad.a@paradise.net.nz>
* apps/app_directory.c: Add supporting code for app-directory pause
option. Since 1.6.1 CLI help reports that option p(n) 'initial
pause' is available. Supporting code was never implemented.
(closes issue #16751) Reported by: alecdavis Patches:
directory_pause.trunk.diff.txt uploaded by alecdavis (license
585) Tested by: alecdavis Review:
https://reviewboard.asterisk.org/r/481/
2010-03-10 23:15 +0000 [r251736] Jeff Peeler <jpeeler@digium.com>
* tests/test_stringfields.c (added), main/utils.c: Add new unit
test for stringfields. (Copied from reviewboard) Tests the
following: 1. Basic allocation and setting of string fields. 2.
Shrinking a string field and re-expanding it. 3. Growing the last
allocation in a string field pool. 4. Setting a string to a large
value such that a new string field pool must be allocated. In
each part, we make sure that the string field is accurate (has
the correct value in it), make sure that the 2 bytes before the
string field has the correct capacity for the field, and for
tests 2-4, we make sure that the string field is where we expect
it to be in memory. Also tested: 5. Shrinking a string field and
partially re-expanding it. 6. Setting strings in such a way as to
create three separate string field pools and then removing the
middle pool. There is a bug fix in the init function, which
ensures the embedded_pool is set to NULL which is important for
stack allocated structures. Review:
https://reviewboard.asterisk.org/r/185/
2010-03-10 20:54 +0000 [r251682] Tilghman Lesher <tlesher@digium.com>
* funcs/func_strings.c: Hmmm, apparently needed to be fixed in
trunk, too. (closes issue #16900) Reported by: bluecrow76
Patches: asterisk-1.6.2.4-func_strings.diff uploaded by
bluecrow76 (license 270)
2010-03-10 20:53 +0000 [r251680] Leif Madsen <lmadsen@digium.com>
* apps/app_record.c: Be less ambiguous in Record() app docs. For
some reason the documentation for the 'k' application in trunk
and 1.6.2 is different than 1.6.0 and 1.6.1, so I'm setting them
all to match. The wording in 1.6.2 and trunk was ambiguous, so
you could interpret the wording the mean that recording would
continue upon hangup indefinitely, or you could interpret it to
mean that the recorded data would not be discarded upon hangup.
This change makes it clear we mean the latter, and not the
former. Came from a discussion in #asterisk on IRC.
2010-03-10 20:51 +0000 [r251679] Jeff Peeler <jpeeler@digium.com>
* main/features.c: Fix ParkAndAnnounce not respecting parking
options. The patch ensures that if a peer does not exist, parking
settings are read from the channel. A unit test has been written
to ensure proper operation for both standard parking and parking
using masquerades. (closes issue #16592) Reported by: mwyres
Patches: bug_16592.diff uploaded by snuffy (license 35) Review:
https://reviewboard.asterisk.org/r/539/
2010-03-10 20:30 +0000 [r251677] Tilghman Lesher <tlesher@digium.com>
* tests/test_substitution.c, funcs/func_strings.c: It's amazing
what writing a test will find. (issue #16900) Reported by:
bluecrow76
2010-03-10 18:25 +0000 [r251631] Jeff Peeler <jpeeler@digium.com>
* main/abstract_jb.c: Fix jitterbuffer logging not creating
logfiles. Three changes made here: 1) Do not fail if a previous
log does not exist (in fact, this is probably expected). 2)
Ensure that the file descriptor to write to gets assigned
properly. I am at a loss as to why assigning safe_fd outside the
if fixes this, but it makes the if statement slightly less
complicated anyway. 3) Move up the failure message so that the
errno of the failure is not overwritten by fclose. (closes issue
#16917) Reported by: Artem
2010-03-10 16:55 +0000 [r251538-251585] Richard Mudgett <rmudgett@digium.com>
* channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_analog.c,
channels/sig_analog.h, channels/sig_pri.c: Simplified
dahdi_request() channel selection failed reason/cause code. Also
avoid potential crash because cause could be NULL.
* channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_pri.c:
Reduce the amount of database access for
HAVE_PRI_SERVICE_MESSAGES. Rework HAVE_PRI_SERVICE_MESSAGES to
not use the active values directly from the database. Database
access is likely expensive. Database access now only happens on
initialization, destruction, and when the B channel is taken in
or out of service. This change is not related to call waiting but
it would cause the search for a call waiting interface to be very
expensive and slow down D channel message servicing.
2010-03-09 20:30 +0000 [r251475] Tilghman Lesher <tlesher@digium.com>
* codecs/gsm/Makefile, Makefile.rules: Build system modifications
to ensure that Asterisk properly builds on Mac OS X 10.6. (closes
issue #16997) Reported by: jquinn Patches:
20100309__issue16997__2.diff.txt uploaded by tilghman (license
14) Tested by: tilghman, russell
2010-03-08 18:08 +0000 [r251310] Leif Madsen <lmadsen@digium.com>
* contrib/init.d/rc.debian.asterisk, /: Merged revisions 251309 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r251309 | lmadsen | 2010-03-08 12:07:44 -0600 (Mon, 08 Mar 2010)
| 13 lines Fix Debian init script to not use -c. When using the
init script as-is currently, it could cause issues on Debian such
as high CPU usage. This fix has worked for several people so I'm
implementing the change. (closes issue #16784) Reported by:
pabelanger Tested by: pabelanger, mnick, davidw, mutineer612
(closes issue #16887) Reported by: jlpedrosa Tested by:
jlpedrosa, mutineer612 ........
2010-03-08 05:15 +0000 [r251262-251263] Tilghman Lesher <tlesher@digium.com>
* configure, include/asterisk/autoconfig.h.in, configure.ac,
main/stdtime/localtime.c: Remove portions that weren't meant to
be committed for the OS X compat fix
* funcs/func_pitchshift.c, configure,
include/asterisk/autoconfig.h.in, main/Makefile, configure.ac,
main/stdtime/localtime.c: Change needed to make Mac OS X 10.6
happy
2010-03-07 14:53 +0000 [r251221-251222] Michiel van Baak <michiel@vanbaak.info>
* channels/chan_skinny.c: Clean transmit_* for start/stop media
transmission Small patch changing skinny_set_rtp_peer to use
transmit_stopmediatransmission and to use new
transmit_startmediatransmission. Basic testing on 30VIP's by
wedhorn Basic testing on 7960 by me (closes issue #16956)
Reported by: wedhorn Patches: skinny-clean05b.diff uploaded by
wedhorn (license 30) Tested by: wedhorn,mvanbaak
* channels/chan_skinny.c: Cleanup transmit_callstate handling Broke
the various functions included in transmit_callstate to their own
functions. Transmit_callstate now just transmits callstate.
Generally left the functionality as it was, which highlight some
minor code issues (eg multiple transmit_callstate's). I did
however revise the hint code usage of the old transmit_callstate
as it it not appropriate to put a device on hook based on the
change of a hinted device. (closes issue #16939) Reported by:
wedhorn Patches: skinny-clean04.diff uploaded by wedhorn (license
30) Tested by: mvanbaak,wedhorn
2010-03-07 00:45 +0000 [r251181] Alexandr Anikin <may@telecom-service.ru>
* addons/ooh323c/src/ooq931.c: small log issue from bug 0016664
2010-03-06 14:16 +0000 [r251137] Russell Bryant <russell@digium.com>
* channels/chan_sip.c: Fix a crash in SIP blind transfer handling
found by an automated external test. The first real test added to
the external test suite found a pretty nasty crash that occurred
in Asterisk trunk. The crash was due to a race condition between
the REFER handling and channel destruction in the channel thread.
After the transfer has been completed, we go back to the
transferrer channel and try to lock it so we can fire off a CEL
event. However, there was no guarantee that the channel was still
around at that point since it's racing against the channel
thread. Since ast_channel is a reference counted object, the fix
is simple. The code unlocks the transferrer channel before
finally completing the transfer with an async goto. At this point
the channel thread is going to start call tear down and the
channel will eventually be destroyed. To ensure that the channel
is valid when we want to fire off the CEL event, increase the
channel's reference count.
2010-03-05 21:51 +0000 [r251038-251087] David Vossel <dvossel@digium.com>
* funcs/func_pitchshift.c: fixes xml error in func_pitchshift
* funcs/func_pitchshift.c (added), CHANGES: PITCH_SHIFT dialplan
function The PITCH_SHIFT function can be used on a channel to
independently modify the pitch of both rx and tx audio streams.
Now you can improve your conference calls by assigning a random
pitch effect to everyone entering a meetme room, or just make
your day more interesting by making your co-workers sound funny.
These are just some of the numerious practical uses for this
function. Enjoy! https://reviewboard.asterisk.org/r/526/
2010-03-05 19:32 +0000 [r251022] Russell Bryant <russell@digium.com>
* build_tools/menuselect-deps.in, configure,
include/asterisk/autoconfig.h.in, configure.ac, makeopts.in,
pbx/pbx_gtkconsole.c (removed): Remove pbx_gtkconsole and related
gtk1 checks. Review: https://reviewboard.asterisk.org/r/541/
2010-03-05 19:10 +0000 [r250979] Jeff Peeler <jpeeler@digium.com>
* apps/app_followme.c: Fix app_followme playing wrong sound files.
Fixes regression introduced in 140167 that uses the wrong
variable names. (closes issue #16930) Reported by: ianc Patches:
fix_reload_followme.diff uploaded by ianc (license 998)
2010-03-05 05:03 +0000 [r250917] Russell Bryant <russell@digium.com>
* channels/chan_sip.c: Fix up some of chan_sip's usage of the RTP
engine API. The get_local_address() function for an RTP instance
was used when building an SDP, but the results were not honored.
The RTP engine activate() function was not being used once we
have determined that media will now flow.
2010-03-05 04:37 +0000 [r250913] Tilghman Lesher <tlesher@digium.com>
* apps/app_voicemail.c: Missing quote in ODBC query. (closes issue
#16953) Reported by: elguero Patches:
app_voicemail-odbc-syntax-fix.diff uploaded by elguero (license
37)
2010-03-05 02:07 +0000 [r250871] Russell Bryant <russell@digium.com>
* include/asterisk/rtp_engine.h: Fix up the ast_rtp_property enum.
The mis-placement of the latest entry meant that when it was set,
it was writing one index past the end of the properties array in
the ast_rtp_instance (which happened to be the local_address
field).
2010-03-05 01:05 +0000 [r250787] Jeff Peeler <jpeeler@digium.com>
* /, res/res_musiconhold.c: Merged revisions 250786 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r250786 | jpeeler | 2010-03-04 19:02:58 -0600 (Thu, 04
Mar 2010) | 9 lines Fix not being able to specify a URL in MOH
class directory. Don't attempt to chdir on a URL! (closes issue
#16875) Reported by: raarts Patches: moh-http.patch uploaded by
raarts (license 937) ........
2010-03-04 20:12 +0000 [r250730] Mark Michelson <mmichelson@digium.com>
* funcs/func_channel.c: Adjust XML for func_channel to indicate
that rtpdest can take a "text" argument.
2010-03-03 21:28 +0000 [r250609-250614] Leif Madsen <lmadsen@digium.com>
* /: Recorded merge of revisions 250613 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r250613 | lmadsen | 2010-03-03 16:28:02 -0500 (Wed, 03 Mar 2010)
| 11 lines Update existing Local channel documentation. A
complete re-write of the Local channel documentation has been
performed, with the existing information from localchannel.txt
and localchannel.tex merged in. (issue #16637) Reported by: kobaz
Patches: localchannel.tex uploaded by lmadsen (license 10)
localchannel.txt uploaded by lmadsen (license 10) Tested by:
lmadsen, jsmith, mmichelson ........
* doc/tex/localchannel.tex: Update existing Local channel
documentation. A complete re-write of the Local channel
documentation has been performed, with the existing information
from localchannel.txt and localchannel.tex merged in. (closes
issue #16637) Reported by: kobaz Patches: localchannel.tex
uploaded by lmadsen (license 10) localchannel.txt uploaded by
lmadsen (license 10) Tested by: lmadsen, jsmith, mmichelson
2010-03-03 19:38 +0000 [r250565] Richard Mudgett <rmudgett@digium.com>
* apps/app_dial.c, channels/chan_dahdi.c, main/dial.c,
channels/chan_local.c, include/asterisk/channel.h,
apps/app_queue.c: Removed cdrflags from ast_channel structure.
Only chan_dahdi set a value in cdrflags. Everyone else just
copied it around the system. Noone cared about any value it may
have contained.
2010-03-03 19:06 +0000 [r250481] Jeff Peeler <jpeeler@digium.com>
* channels/chan_dahdi.c, channels/sig_analog.c, /: Merged revisions
250480 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r250480 | jpeeler | 2010-03-03 13:04:11 -0600 (Wed, 03 Mar 2010)
| 15 lines Make sure to clear red alarm after polarity reversal.
From the issue: The automatic overnight line tests (or manual
ones) used on UK (BT) lines causes a red alarm on a dahdi /
TDM400P connected channel. This is because the line uses voltage
tests (battery loss) and polarity reversal. The polarity reversal
causes chan_dahdi to initiate v23 CallerID processing but during
this the event DAHDI_EVENT_NOALARM is ignored so that the alarm
is never cleared. (closes issue #14163) Reported by: jedi98
Patches: chan_dahdi-1.4-inalarm.diff uploaded by jedi98 (license
653) Tested by: mattbrown, Chainsaw, mikeeccleston ........
2010-03-03 19:02 +0000 [r250395-250478] David Vossel <dvossel@digium.com>
* main/test.c: Changes 0ms to <1ms in cli END results during 'test
execute'
* /, channels/chan_iax2.c: Merged revisions 250394 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r250394 | dvossel | 2010-03-03 12:02:27 -0600 (Wed, 03
Mar 2010) | 16 lines fixes problem with duplicate TXREQ packets
When Asterisk receives an IAX2 TXREQ packet, try_transfer() will
call store_by_transfercallno() to link the chan_iax2_pvt struct
into iax_transfercallno_pvts. If a duplicate TXREQ packet is
received for the same call, the pvt struct will be linked into
iax_transfercallno_pvts multiple times. This patch fixes this.
Thanks rain for debugging this and providing a patch! (closes
issue #16904) Reported by: rain Patches:
iax2-double-txreq-fix.diff uploaded by rain (license 327) Tested
by: rain, dvossel ........
2010-03-03 17:37 +0000 [r250392] Jeff Peeler <jpeeler@digium.com>
* channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, CHANGES:
Add new config option to control AMI alarm event reporting in
chan_dahdi. New config parameter "reportalarms" added in
chan_dahdi.conf which supports the following possible values:
"channels": report each channel alarms (current behavior, default
for backward compatibility) "spans": report an "SpanAlarm" event
when the span of any configured channel is alarmed "all": report
channel and span alarms (aggregated behavior) "none": do not
report any alarms (closes issue #16709) Reported by: nahuelgreco
Patches: chan_dahdi.c.reportalarms.patch uploaded by nahuelgreco
(license 162)
2010-03-03 16:43 +0000 [r250303-250346] Tilghman Lesher <tlesher@digium.com>
* main/editline/configure: One more fix to editline
* main/editline/configure, main/editline/Makefile.in,
main/editline/sys.h, main/editline/configure.in: Eliminate
remaining libedit warnings (shown in bamboo)
2010-03-03 15:39 +0000 [r250302] Matthew Nicholson <mnicholson@digium.com>
* res/res_fax.c, apps/app_fax.c, CHANGES, res/res_fax_spandsp.c:
Updated CHANGES file to mention res_fax and res_fax_spandsp. Also
fixed MODULEINFO depends and conflicts for app_fax, res_fax, and
res_fax_spandsp.
2010-03-03 00:18 +0000 [r250235-250246] David Vossel <dvossel@digium.com>
* channels/chan_sip.c: fixes signed to unsigned int comparision
issue for FaxMaxDatagram value.
* main/test.c: fixes assumption that test failed if it did not pass
when generating results
* tests/test_utils.c: base64 unit test
2010-03-02 23:22 +0000 [r250190-250213] Matthew Nicholson <mnicholson@digium.com>
* configs/res_fax.conf.sample (added), include/asterisk/res_fax.h
(added): Merge missed files from res_fax/res_fax_spandsp merge.
* res/res_fax.c (added), res/res_fax.exports (added),
include/asterisk/frame.h, res/res_fax_spandsp.c (added): Merge
res_fax and res_fax_spandsp.
2010-03-02 21:58 +0000 [r250141] David Vossel <dvossel@digium.com>
* apps/app_directed_pickup.c, CHANGES: adds 'p' option to
PickupChan The 'p' option allows the PickupChan app to pickup a
ringing phone by looking for the first match to a partial channel
name rather than requiring a full match. (closes issue #16613)
Reported by: syspert Patches: pickipbycallid.patch uploaded by
syspert (license 938) pickupbycallerid_v2.patch uploaded by
dvossel (license 671) Tested by: dvossel, syspert
2010-03-02 21:09 +0000 [r249950-250051] Leif Madsen <lmadsen@digium.com>
* doc/tex/imapstorage.tex: Update IMAP documentation. Update the
IMAP documentation to make it clear that storing voicemails in
the same folder as a large number of emails could potentially
cause significant slow downs when writing or retrieving
voicemails. (issue #16704) Reported by: TimeHider Tested by:
lmadsen, TimeHider
* /, configs/cdr.conf.sample: Merged revisions 250043 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r250043 | lmadsen | 2010-03-02 15:51:35 -0500 (Tue, 02
Mar 2010) | 7 lines Update documentation to clarify purpose of
unanswered option. (closes issue #16267) Reported by: elsto
Patches: cdr.conf.sample.patch.txt uploaded by lmadsen (license
10) Tested by: davidw, elsto ........
* /: Recorded merge of revisions 250041 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r250041 | lmadsen | 2010-03-02 15:45:37 -0500 (Tue, 02 Mar 2010)
| 4 lines Update documentation to not imply we support overriding
options. (issue #16855) Reported by: davidw ........
* doc/tex/configuration.tex: Update documentation to not imply we
support overriding options. (closes issue #16855) Reported by:
davidw
* apps/app_directory.c: Fix literal values wrapped in
documentation. (closes issue #16145) Reported by: tilghman
2010-03-02 19:39 +0000 [r249947] Alec L Davis <sivad.a@paradise.net.nz>
* apps/app_echo.c: revert ability to exit echo app caused a
regression, as only supported VOICE, not VIDEO etc. (issue
#16880)
2010-03-02 19:24 +0000 [r249912-249925] Leif Madsen <lmadsen@digium.com>
* main/features.c: Add missing description of the PARKINGLOT
variable in XML documentation. (closes issue #16743) Reported by:
snuffy Patches: parkingdoc.diff uploaded by snuffy (license 35)
* pbx/pbx_dundi.c: Convert some DUNDI functions to XML
documentation. (closes issue #16798) Reported by: snuffy Patches:
xml_dundi.diff uploaded by snuffy (license 35)
2010-03-02 19:08 +0000 [r249893] David Vossel <dvossel@digium.com>
* channels/chan_unistim.c, configs/chan_dahdi.conf.sample,
configs/console.conf.sample, channels/chan_local.c,
channels/chan_sip.c, configs/oss.conf.sample,
configs/usbradio.conf.sample, configs/misdn.conf.sample,
channels/chan_console.c, channels/chan_gtalk.c,
channels/chan_oss.c, channels/misdn_config.c,
include/asterisk/abstract_jb.h, configs/alsa.conf.sample,
channels/chan_jingle.c, channels/chan_usbradio.c,
channels/chan_dahdi.c, channels/chan_skinny.c,
configs/mgcp.conf.sample, main/abstract_jb.c,
channels/chan_h323.c, channels/chan_alsa.c,
configs/sip.conf.sample, channels/chan_mgcp.c: fixes adaptive
jitterbuffer configuration When configuring the adaptive
jitterbuffer, the target_extra value not only could not be set
from the configuration, but was not even being set to its proper
default. This value is required in order for the adaptive
jitterbuffer to work correctly. To resolve this a config option
has been added to expose this value to the conf files, and a
default value is provided when no config specific value is
present.
2010-03-02 19:02 +0000 [r249892] Leif Madsen <lmadsen@digium.com>
* apps/app_osplookup.c, apps/app_confbridge.c, res/res_jabber.c:
Fix several XML documentation validate errors.
2010-03-02 18:31 +0000 [r249889-249891] Jeff Peeler <jpeeler@digium.com>
* apps/app_voicemail.c: fix build by checking result of symlink in
test_voicemail_vmsayname
* CHANGES, apps/app_voicemail.c: Add new application VMSayName for
use with voicemail. VMSayName that will play the recorded name of
the voicemail user if it exists, otherwise will play the mailbox
number. A unit test has been written to verify correct
functionality called test_voicemail_vmsayname. (closes issue
#14973) Reported by: ghjm Review:
https://reviewboard.asterisk.org/r/530/
2010-03-02 07:38 +0000 [r249759-249801] Alec L Davis <sivad.a@paradise.net.nz>
* apps/app_echo.c: fixes ability to exit echo app when called from
a ISDN channel, null frames prevent '#' exit. Now only echo back
VOICE and DTMF frames (issue #16880) Reported by: alecdavis
Patches: echo_exit.diff.txt uploaded by alecdavis (license 585)
Tested by: alecdavis
* channels/chan_dahdi.c: fix asterisk setting of pritimers from
chan_dahdi.conf regression since sig_pri split. (issue #16909)
Reported by: alecdavis Patches: pritimer.asterisk.diff.txt
uploaded by alecdavis (license 585) Tested by: alecdavis
2010-03-01 19:36 +0000 [r249672] Sean Bright <sean@malleable.com>
* /, apps/app_voicemail.c: Merged revisions 249671 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r249671 | seanbright | 2010-03-01 14:35:01 -0500 (Mon,
01 Mar 2010) | 11 lines Fix crash in app_voicemail related to
message counting. We were passing a 'struct inprocess **' and
treating it like a 'struct inprocess *' causing a segfault.
(closes issue #16921) Reported by: whardier Patches:
20100301_issue16921.patch uploaded by seanbright (license 71)
Tested by: whardier ........
2010-03-01 19:33 +0000 [r249669-249670] Michiel van Baak <michiel@vanbaak.info>
* channels/chan_skinny.c: Cleanup display_*message functions. This
patch splits transmit_displaymessage into
transmit_clear_display_message and transmit_display_message which
better aligns with the skinny protocol. The new
transmit_display_message is not used in the current code, but
will be and so it is commented. Moved handle_datetime from this
function to onhook and offhook functions (display now properly
cleared at the end of a call on 30VIP). Removed skinny debug
messages from inline code as there's an ast_verb in
transmit_clear_display_message. Also, removed commentary that it
was a clear display as it is now apparent from the function name.
Split transmit_displaypromptmessage into display and clear.
(closes issue #16878) Reported by: wedhorn Patches:
skinny-clean02.diff uploaded by wedhorn (license 30)
skinny-clean03.diff uploaded by wedhorn (license 30)
* channels/chan_skinny.c: fix endianes issues in chan_skinny
(closes issue #16826) Reported by: PipoCanaja Patches:
chan_skinny.c_bigendianPatch_20100218.diff uploaded by PipoCanaja
(license 994) Tested by: wedhorn
2010-03-01 18:36 +0000 [r249623] Tilghman Lesher <tlesher@digium.com>
* apps/app_voicemail.c: Constify a bit of app_voicemail, to make
ODBC and IMAP compile once again.
2010-03-01 17:11 +0000 [r249538] Jeff Peeler <jpeeler@digium.com>
* channels/chan_local.c, /: Merged revisions 249536 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r249536 | jpeeler | 2010-03-01 11:02:03 -0600 (Mon, 01
Mar 2010) | 11 lines Modify queued frames from local channels to
not set the other side to up In this case, attended transfers
were broken due to ast_feature_request_and_dial detecting the
channel being set to up before the answer frame could be read and
therefore failing to mark the channel as ready. This fix is a
regression fix for 244785, which should continue to work properly
as well. (closes issue #16816) Reported by: jamhed Tested by:
jamhed, corruptor ........
2010-02-28 20:50 +0000 [r249491] Tilghman Lesher <tlesher@digium.com>
* apps/app_voicemail.c: Fix unit test that Alec Davis broke.
(closes issue #16927) Reported by: alecdavis
2010-02-28 16:36 +0000 [r249449] Alec L Davis <sivad.a@paradise.net.nz>
* apps/app_voicemail.c: make unit test check for NULL folder, which
then defaults to INBOX previous test, gave false level of
assurance that code was healthy. (issue #16927) Reported by:
alecdavis Patches: based on app_voicemail_test.diff.txt uploaded
by alecdavis (license 585) Tested by: alecdavis
2010-02-28 07:10 +0000 [r249405] Tilghman Lesher <tlesher@digium.com>
* include/asterisk/app.h, apps/app_voicemail.c: Properly document
voicemail API documents. Also fix a crash reported via the -dev
list.
2010-02-27 22:49 +0000 [r249320] Alec L Davis <sivad.a@paradise.net.nz>
* channels/sig_pri.c: overlap receiving: automatically send CALL
PROCEEDING when dialplan starts Following Q.931 5.2.4 When the
user has determined that sufficient call information has been
received the user shall stop T302 and send CALL PROCEEDING to the
network. Previously timeouts were possible if the dialplan took a
long time to issue any response back to the network. Verified
that our local TELCO also does the same. (issue #16789) Reported
by: alecdavis Patches: overlap_receiving_trunk.diff.txt uploaded
by alecdavis (license 585) Tested by: alecdavis
2010-02-27 14:08 +0000 [r249235] Kevin P. Fleming <kpfleming@digium.com>
* /, channels/chan_iax2.c: Merged revisions 249234 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r249234 | kpfleming | 2010-02-27 09:07:59 -0500 (Sat, 27
Feb 2010) | 1 line add a reference to the now-published IAX2 RFC
........
2010-02-26 18:41 +0000 [r249187] Tilghman Lesher <tlesher@digium.com>
* apps/app_voicemail.c: Cleanups to fix bugs in the VM count API
functions. - Urgent voicemails were not attached, because the
attachment code looked in the wrong folder. - Urgent voicemails
were sometimes counted twice when displaying the count of new
messages. - Backends were inconsistent as to which voicemails
each API counted. - Unit tests added to verify behavior in the
future. (closes issue #15654) Reported by: tomo1657 Patches:
20100225__issue15654.diff.txt uploaded by tilghman (license 14)
Tested by: tilghman (closes issue #16448) Reported by: hevad
Review: https://reviewboard.asterisk.org/r/525/
2010-02-26 18:41 +0000 [r249186] David Vossel <dvossel@digium.com>
* main/test.c: adds Time field to "test show results" cli command
2010-02-26 17:13 +0000 [r249101-249105] Mark Michelson <mmichelson@digium.com>
* main/features.c: Send a manager event when the manager
BridgeAction command is used. (closes issue #16769) Reported by:
syspert Patches: bridgeaction.patch uploaded by syspert (license
938)
* /, channels/chan_sip.c: Merged revisions 249100 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r249100 | mmichelson | 2010-02-26 11:04:29 -0600 (Fri, 26 Feb
2010) | 8 lines For T.38 reINVITEs treat a 606 the same as a 488.
(closes issue #16792) Reported by: vrban Patches: t38_606.patch
uploaded by vrban (license 756) ........
2010-02-26 08:45 +0000 [r249009-249058] Russell Bryant <russell@digium.com>
* cdr/cdr_sqlite3_custom.c, cdr/cdr_syslog.c, cdr/cdr_sqlite.c,
cdr/cdr_adaptive_odbc.c, cdr/cdr_pgsql.c, cdr/cdr_odbc.c,
cdr/cdr_radius.c, cdr/cdr_custom.c, cdr/cdr_manager.c,
cdr/cdr_tds.c, cdr/cdr_csv.c: formatting tweaks and
constification
* main/cdr.c: Trim trailing whitespace (to help reduce diff against
cdr-q branch)
* include/asterisk/cdr.h: Trim trailing whitespace, convert lists
of defines to enums
* cdr/cdr_sqlite.c: trivial formatting tweak (working on reducing
diff against trunk for cdr-q)
* cdr/cdr_sqlite3_custom.c: remove include
* cdr/cdr_csv.c: constification, remove include
* cdr/cdr_tds.c: Remove unnecessary includes, formatting tweak
* cdr/cdr_pgsql.c: constification and remove unnecessary include
2010-02-25 23:09 +0000 [r248952] Jeff Peeler <jpeeler@digium.com>
* /, res/res_monitor.c: Merged revisions 248860 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r248860 | jpeeler | 2010-02-25 15:22:06 -0600 (Thu, 25 Feb 2010)
| 18 lines Ensure that monitor recordings are written to the
correct location (again) This is an extension to 248757. As such
the dialplan test has been extended: exten => 5040, 1,
monitor(wav,tmp/jeff/monitor_test,b) exten => 5040, n,
dial(sip/5001) exten => 5041, 1,
monitor(wav,/tmp/jeff/monitor_test2,b) exten => 5041, n,
dial(sip/5001) exten => 5042, 1, monitor(wav,monitor_test3,b)
exten => 5042, n, dial(sip/5001) exten => 5043, 1,
monitor(wav,tmp/jeff/monitor_test3,m) exten => 5043, n,
changemonitor(monitor_test4) exten => 5043, n, dial(sip/5001)
exten => 5044, 1, monitor(wav,monitor_test4,m) exten => 5044, n,
changemonitor(tmp/jeff/monitor_test5) ; this looks to fail by
design and emits a warning exten => 5044, n, dial(sip/5001)
........
2010-02-25 22:41 +0000 [r248946] Mark Michelson <mmichelson@digium.com>
* main/acl.c: Fix incorrect ACL behavior when CIDR notation of "/0"
is used. AST-2010-003
2010-02-25 21:22 +0000 [r248861] Tilghman Lesher <tlesher@digium.com>
* /, main/asterisk.c: Merged revisions 248859 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r248859 | tilghman | 2010-02-25 15:21:05 -0600 (Thu, 25 Feb 2010)
| 15 lines Some platforms clear /var/run at boot, which makes
connecting a remote console... difficult. Previously, we only
created the default /var/run/asterisk directory at install time.
While we could create it in the init script, that would not work
for those who start asterisk manually from the command line. So
the safest thing to do is to create it as part of the Asterisk
boot process. This also changes the ownership of the directory,
because the pid and ctl files are created after we setuid/setgid.
(closes issue #16802) Reported by: Brian Patches:
20100224__issue16802.diff.txt uploaded by tilghman (license 14)
Tested by: tzafrir ........
2010-02-25 18:37 +0000 [r248793] Jeff Peeler <jpeeler@digium.com>
* /, res/res_monitor.c: Merged revisions 248757 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r248757 | jpeeler | 2010-02-25 12:06:54 -0600 (Thu, 25 Feb 2010)
| 15 lines Ensure that monitor recordings are written to the
correct location. Recordings should be placed in the monitor
directory when a non-absolute path is used. Exact dialplan used
for testing: exten => 5040, 1,
monitor(wav,tmp/jeff/monitor_test,b) exten => 5040, n,
dial(sip/5001) exten => 5041, 1,
monitor(wav,/tmp/jeff/monitor_test2,b) exten => 5041, n,
dial(sip/5001) exten => 5042, 1, monitor(wav,monitor_test3,b)
exten => 5042, n, dial(sip/5001) ABE-2101 ........
2010-02-24 22:44 +0000 [r248584-248667] Tilghman Lesher <tlesher@digium.com>
* channels/Makefile: Also kill the .i files, or else the build
process will not recreate them, when we change flags. Fixes a
weird symbol problem mmichelson was having in a group branch, but
also applies to trunk.
* /, main/logger.c, include/asterisk/term.h, main/term.c: Merged
revisions 248582 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r248582 | tilghman | 2010-02-24 15:02:18 -0600 (Wed, 24 Feb 2010)
| 7 lines Remove color code sequences from verbose messages that
go to logfiles. (closes issue #16786) Reported by: dodo Patches:
logger2.patch uploaded by dodo (license 989) Tested by: tilghman
........
2010-02-24 06:39 +0000 [r248533-248534] Russell Bryant <russell@digium.com>
* funcs/func_strings.c: Remove unnecessary warning message, make a
couple of formatting tweaks
* tests/test_strings.c: Add ASTERISK_FILE_VERSION macro.
2010-02-23 22:29 +0000 [r248489] Mark Michelson <mmichelson@digium.com>
* tests/test_strings.c (added): Unit test for ast_str API. Review:
https://reviewboard.asterisk.org/r/517
2010-02-23 16:34 +0000 [r248397] David Vossel <dvossel@digium.com>
* /, channels/chan_sip.c: Merged revisions 248396 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r248396 | dvossel | 2010-02-23 10:26:05 -0600 (Tue, 23 Feb 2010)
| 9 lines fixes invite with replaces deadlock (closes issue
#16862) Reported by: pwalker Patches: replaces_deadlock_1.4
uploaded by dvossel (license 671) Tested by: pwalker, dvossel
........
2010-02-22 20:19 +0000 [r248347] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Move the REF_DEBUG comment higher in the
include list. Uncommenting the REF_DEBUG definition where it was
in the source resulted in only a small part of the astobj2
references being logged to a file. Moving this up higher in the
include list causes all references to be logged as they should
be.
2010-02-22 06:45 +0000 [r248225-248226] Russell Bryant <russell@digium.com>
* include/asterisk/taskprocessor.h, main/taskprocessor.c: Minor
tweaks to comment blocks and includes. Fix the copyright lines,
tweak doxygen formatting, and remove some unnecessary includes.
* tests/test_devicestate.c: Tweak copyright and author lines.
2010-02-21 12:09 +0000 [r248184] Michiel van Baak <michiel@vanbaak.info>
* channels/chan_skinny.c: Cleanup transmit_* functions, part 1
Break transmit_tone into transmit_start_tone and
transmit_stop_tone as per the skinny protocol. (closes issue
#16874) Reported by: wedhorn Patches: skinny-clean01.diff
uploaded by wedhorn (license 30)
2010-02-20 22:37 +0000 [r248108] Olle Johansson <oej@edvina.net>
* res/res_rtp_asterisk.c: Improve support for RTCP reports without
report blocks
2010-02-19 18:38 +0000 [r248003] Moises Silva <moises.silva@gmail.com>
* channels/chan_dahdi.c: mfcr2 issue 0016844 - Fix portability bit
fields and make mfcr2_immediate_accept work again, reported and
patched by korihor
2010-02-19 17:40 +0000 [r247915] David Vossel <dvossel@digium.com>
* channels/chan_sip.c: handle_request_invite revise comment, fix
coding guideline issues I'm working with this code right now
trying to analyze a deadlock. This change is just to clean up a
few things before I make a more complex patch.
2010-02-19 17:33 +0000 [r247914] Richard Mudgett <rmudgett@digium.com>
* channels/chan_misdn.c, /: Merged revisions 247910 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
................ r247910 | rmudgett | 2010-02-19 11:18:49 -0600
(Fri, 19 Feb 2010) | 55 lines Merged revision 247904 from
https://origsvn.digium.com/svn/asterisk/be/branches/C.2-...
.......... r247904 | rmudgett | 2010-02-19 10:49:44 -0600 (Fri,
19 Feb 2010) | 49 lines Make chan_misdn DTMF processing
consistent with other channel technologies. The processing of
DTMF tones on the receiving side of an ISDN channel is
inconsistent with the way it is handled in other channels,
especially DAHDI analog. This causes DTMF tones sent from an ISDN
phone to be doubled at the connected party. We are using the
following 2 options of misdn.conf 1) astdtmf=yes 2) senddtmf=yes
Option one is necessary because the asterisk DSP DTMF detection
is better than mISDN's internal DSP. Not as many false positives.
Option two is necessary to transmit DTMF tones end to end when
mISDN channels are connected to SIP channels with out of band
DTMF for example. The symptom is that DTMF tones sent by an ISDN
phone are doubled on the way through asterisk when two mISDN
channels are connected with a Local channel in between or if it
is bridged to an analog channel. The doubling of DTMF tones is
because DTMF is passed inband to asterisk by the mISDN channel
and passed out of band once again after the release of the DTMF
tone. Passing it inband is wrong. Neither an analog channel nor
SIP channel passes DTMF inband if configured to inband DTMF.
Analog and SIP channels filter out the DTMF tones because they
use the voice frames returned by ast_dsp_process. But chan_misdn
passes the unfiltered input voice frames instead. To overcome one
aspect of the problem, the doubling of DTMF tones when two mISDN
channels are directly bridged, someone made an 'optimization',
where in that case the DTMF tone passed out-of-band to the peer
channel is not translated to an inband tone at the transmit side.
This optimization is bad because it does not work in general. For
example, analog channels or mISDN channels when bridged through
an intermediary local channel will generate DTMF tones from
out-of-band information. Also, of course, it must not be done
when there is no inband DTMF available. This patch fixes the
issue. Now chan_misdn will filter the received inband DTMF signal
the same as other channel types. Another change included: No need
to build an extra translation path because ast_process_dsp does
it if required. Patches: misdn-dtmf.patch JIRA ABE-2080
................
2010-02-18 23:13 +0000 [r247787-247841] Tilghman Lesher <tlesher@digium.com>
* res/res_speech.c: Revert an errant part of a previous cleanup, to
fix a memory corruption issue. (closes issue #16368) Reported by:
thirionjwf Patches: res_speech.c.patch uploaded by thirionjwf
(license 955)
* channels/chan_sip.c: If the peer record is from realtime, it
could be set to 0, due to MySQL not representing NULL well in
integer columns. NULL means the value is not specified for the
column, which normally means the driver uses whatever is the
default value. However, on MySQL, placing a NULL in either a
float or integer column results in a retrieval of the 0 value.
Hence, users get an errant error on load. This patch suppresses
that error and makes the value as if it was not there. Note that
this cannot be done in the realtime driver, because the lack of
difference between NULL and 0 can only be intepreted correctly by
the driver itself. If we did it in the realtime driver, then it
would be effectively impossible to set any realtime field to 0,
because it would act as if the field were unspecified and
possibly take on a different value. (closes issue #16683)
Reported by: wdoekes
2010-02-18 21:23 +0000 [r247736-247770] David Vossel <dvossel@digium.com>
* bridges/bridge_softmix.c: fixes confbridge crash when no timing
module is loaded. (closes issue #16471) Reported by: kjotte
Patches: M16471.diff uploaded by junky (license 177) Tested by:
kjotte, junky
* apps/app_queue.c: fixes Queue with C option crash (closes issue
#16475) Reported by: okrief Patches: queue_crash.diff uploaded by
dvossel (license 671)
2010-02-18 19:39 +0000 [r247652] Matthew Nicholson <mnicholson@digium.com>
* /, main/features.c: Merged revisions 247651 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r247651 | mnicholson | 2010-02-18 13:38:09 -0600 (Thu, 18 Feb
2010) | 6 lines Copy the calling party's account code to the
called party if they don't already have one. (closes issue
#16331) Reported by: bluefox Tested by: mnicholson ........
2010-02-18 18:31 +0000 [r247609] Richard Mudgett <rmudgett@digium.com>
* main/channel.c: Fix placing ISDN calls on hold preventing native
bridging from being reexamined after a transfer. Consider the
following scenario: /-- B A == * == Network \-- C Party B calls
party A (EuroISDN BRI phone) Party A puts B on hold using the
HOLD/RETRIEVE messages. Party A calls party C. Party A puts C on
hold to talk with party B again. Party A transfers B to C by
hanging up. The call does not get the opportunity to get
re-transferred into the ISDN network by the native bridge because
native bridging is not being reexamined after the initial
transfer.
2010-02-18 16:54 +0000 [r247503-247509] Leif Madsen <lmadsen@digium.com>
* /, README-SERIOUSLY.bestpractices.txt: Merged revisions 247508
via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r247508 | lmadsen | 2010-02-18 11:53:44 -0500 (Thu, 18 Feb 2010)
| 1 line Add additional link to best practices document per
jsmith. ........
* /, README-SERIOUSLY.bestpractices.txt (added): Merged revisions
247502 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r247502 | lmadsen | 2010-02-18 11:38:17 -0500 (Thu, 18 Feb 2010)
| 10 lines Add best practices documentation. (issue #16808)
Reported by: lmadsen (issue #16810) Reported by: Nick_Lewis
Tested by: lmadsen Review:
https://reviewboard.asterisk.org/r/507/ ........
2010-02-18 16:34 +0000 [r247500] Philippe Sultan <philippe.sultan@gmail.com>
* CHANGES, res/res_jabber.c: Add a new manager event for our
buddies status. The new JabberStatus event gives a concise view
of the status change to the AMI clients. Thanks fiddur! (closes
issue #16760) Reported by: fiddur Patches: 244498.2.diff uploaded
by fiddur (license 678) Tested by: fiddur, phsultan
2010-02-18 04:20 +0000 [r247423] Russell Bryant <russell@digium.com>
* Makefile, /, sounds/Makefile: Merged revisions 247422 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r247422 | russell | 2010-02-17 22:19:01 -0600 (Wed, 17 Feb 2010)
| 10 lines Tweak argument handling for wget in the sounds
Makefile. 1) Fix the check to see if we are using wget to not be
full of fail. The configure script populates this variable with
the absolute path to wget if it is found, so it didn't work. 2)
Allow some extra arguments to be passed in for wget. This is just
a simple change to allow our Bamboo build script to tell wget to
be quiet and not fill up our logs with download status output.
........
2010-02-17 22:44 +0000 [r247335-247381] Mark Michelson <mmichelson@digium.com>
* main/test.c: Fix a couple of bugs in test tab completion. 1. Add
missing unlock of lists. 2. Swap order of arguments to
test_cat_cmp in complete_test_name.
* main/test.c: Tab completion for test categories and names for
"test show registered" and "test execute" CLI commands.
* main/strings.c, include/asterisk/strings.h: Fix two problems in
ast_str functions found while writing a unit test. 1. The
documentation for ast_str_set and ast_str_append state that the
max_len parameter may be -1 in order to limit the size of the
ast_str to its current allocated size. The problem was that the
max_len parameter in all cases was a size_t, which is unsigned.
Thus a -1 was interpreted as UINT_MAX instead of -1. Changing the
max_len parameter to be ssize_t fixed this issue. 2. Once issue 1
was fixed, there was an off-by-one error in the case where we
attempted to write a string larger than the current allotted size
to a string when -1 was passed as the max_len parameter. When
trying to write more than the allotted size, the ast_str's
__AST_STR_USED was set to 1 higher than it should have been.
Thanks to Tilghman for quickly spotting the offending line of
code. Oh, and the unit test that I referenced in the top line of
this commit will be added to reviewboard shortly. Sit tight...
2010-02-17 19:51 +0000 [r247295] Jeff Peeler <jpeeler@digium.com>
* funcs/func_groupcount.c, tests/test_app.c (added), main/app.c,
CHANGES: Add support for GROUP_MATCH_COUNT regex matching on
category Current support for regex matching was previously only
available on the group. Also, error reporting for regex failures
has been added. In addition to this feature enhancement a unit
test has been written to check the regular expression logic to
ensure the count operation is working as expected. (closes issue
#16642) Reported by: kobaz Patches: groupmatch2.patch uploaded by
kobaz (license 834) Review:
https://reviewboard.asterisk.org/r/503/
2010-02-17 19:23 +0000 [r247248-247282] David Vossel <dvossel@digium.com>
* tests/test_devicestate.c: modified device2extension_test's
category
* tests/test_devicestate.c (added): unit test for combined device
state mapping and device to exten state mapping Review:
https://reviewboard.asterisk.org/r/516/
* main/features.c, CHANGES, configs/features.conf.sample: addition
of dynamic parkinglots feature This feature allows for
parkinglots to be created dynamically within the dialplan. Thanks
to all who were involved with getting this patch written and
tested! (closes issue #15135) Reported by: IgorG Patches:
features.dynamic_park.v3.diff uploaded by IgorG (license 20)
2009090400_dynamicpark.diff.txt uploaded by mvanbaak (license 7)
dynamic_parkinglot.diff uploaded by dvossel (license 671) Tested
by: eliel, IgorG, acunningham, mvanbaak, zktech Review:
https://reviewboard.asterisk.org/r/352/
2010-02-17 16:24 +0000 [r247169] Mark Michelson <mmichelson@digium.com>
* /, apps/app_queue.c: Merged revisions 247168 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r247168 | mmichelson | 2010-02-17 10:24:17 -0600 (Wed, 17 Feb
2010) | 3 lines Make sure that when autofill is disabled that
callers not in the front of the queue cannot place calls.
........
2010-02-17 07:01 +0000 [r247124-247125] Tilghman Lesher <tlesher@digium.com>
* main/loader.c: RTP documentation states that you can pass NULL as
the module, so make sure that's really the case.
* channels/sip/include/dialog.h (added), channels/chan_sip.c,
channels/sip/include/config_parser.h,
channels/sip/include/globals.h (added),
channels/sip/dialplan_functions.c (added), channels/Makefile,
channels/sip/include/sip_utils.h,
channels/sip/include/dialplan_functions.h (added): Make all of
the various rtpqos parameters in this branch available from the
CHANNEL function. Also includes a test for retrieving rtpqos
parameters, including a NULL RTP driver. Additionally, some
further separation of the SIP internal API into headers was
necessary. (closes issue #16652) Reported by: kkm Patches:
20100204__issue16652.diff.txt uploaded by tilghman (license 14)
Review: https://reviewboard.asterisk.org/r/501/
2010-02-16 23:44 +0000 [r247076] Mark Michelson <mmichelson@digium.com>
* main/strings.c: Add va_end calls to __ast_str_helper. According
to the man page for stdarg(3), "Each invocation of va_copy() must
be matched by a corresponding invocation of va_end() in the same
function." There were several cases in __ast_str_helper where
va_copy was not matched with a corresponding call to va_end.
2010-02-16 22:58 +0000 [r247035] Alexandr Anikin <may@telecom-service.ru>
* addons/ooh323c/src/ooh323.c, addons/chan_ooh323.c: generate
connected line info update from info in h.323 packets Tested by:
benngard
2010-02-16 21:15 +0000 [r246985] Mark Michelson <mmichelson@digium.com>
* include/asterisk/strings.h: Add some clarifying documentation to
the ast_str_set and ast_str_append functions.
2010-02-16 21:03 +0000 [r246980-246981] David Vossel <dvossel@digium.com>
* main/tcptls.c: swap openssl with OpenSSL in warning message.
(issue #16673)
* main/tcptls.c: warning message if openssl support is missing
while attempting tls connection (closes issue #16673) Reported
by: michaesc Patches: tls_error_msg.diff uploaded by dvossel
(license 671)
2010-02-16 18:29 +0000 [r246942] Mark Michelson <mmichelson@digium.com>
* tests/test_pbx.c (added): Add unit test for dialplan pattern
matching. This test works by reading input from arrays to build a
sample dialplan. From there, patterns are attempted to be matched
against said dialplan, with the expected match given. We then
search in our example dialplan to see if we find a match and if
what we find matches what we expected it to match. (closes issue
#16809) Reported by: lmadsen Tested by: mmichelson Review:
https://reviewboard.asterisk.org/r/504/
2010-02-16 17:07 +0000 [r246899] David Vossel <dvossel@digium.com>
* main/channel.c: fixes sample rate conversion issue with Monitor
application When using ast_seekstream with the read/write streams
of a monitor, the number of samples we are seeking must be of the
same rate as the stream or the jump calculation will be
incorrect. This patch adds logic to correctly convert the number
of samples to jump to the sample rate the read/write stream is
using. For example, if the call is G722 (16khz) and the
read/write stream is recording a 8khz wav, seeking 320 samples of
16khz audio is not the same as seeking 320 samples of 8khz audio
when performing the ast_seekstream on the stream. ABE-2044
2010-02-16 15:36 +0000 [r246710-246863] Tilghman Lesher <tlesher@digium.com>
* build_tools/cflags.xml, build_tools/cflags-devmode.xml: Revert
changes for now, pending discussion
* build_tools/cflags-devmode.xml: Add a few more targets for
DEBUG_THREADLOCALS
* build_tools/cflags.xml, channels/chan_usbradio.c,
build_tools/cflags-devmode.xml, main/strings.c,
apps/app_voicemail.c: Change the blanket rules to delete
.lastclean on all CFLAGS menuselect targets to be more
particular. This change builds upon the recent change to
menuselect to add 'touch_on_change' as an attribute of both
categories and members. This should allow only the most invasive
defines to cause a complete rebuild, while defines which only
affect a subset of modules will only cause a rebuild of that
smaller set.
* channels/chan_sip.c: Allow Timer B to be set on the peer, and
ensure SIP rules are followed (or warn) in comparison to Timer
T1. (closes issue #16643) Reported by: nahuelgreco Patches:
20100204__issue16643.diff.txt uploaded by tilghman (license 14)
Tested by: oej
* Makefile, /: Merged revisions 246709 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r246709 | tilghman | 2010-02-15 17:42:33 -0600 (Mon, 15 Feb 2010)
| 5 lines Make the menuselect instructions correct by allowing
'make menuselect' to actually solve dependency problems.
(Previously, it would fail out again with the same message about
running 'make menuselect', which was NOT at all helpful.)
........
2010-02-15 22:08 +0000 [r246669] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c: Restore triedtopribridge flag code removed
in -r211197. Ooops. Failed to note that we were inside a for loop
and pri_channel_bridge() needs to be executed only once.
2010-02-15 21:37 +0000 [r246667] Tilghman Lesher <tlesher@digium.com>
* utils/utils.xml: Instead of just automatically filtering out in
the Makefile, give an indication of dependencies in menuselect.
2010-02-15 15:45 +0000 [r246627] David Vossel <dvossel@digium.com>
* channels/chan_sip.c, channels/sip/reqresp_parser.c,
channels/sip/include/sip_utils.h,
channels/sip/include/reqresp_parser.h: chan_sip parse code
refactoring plus two new unit tests Code Refactoring Changes -
read_to_parts() moved to reqresp_parser.c and has been renamed as
get_name_and_number() - get_in_brackets() moved to
reqresp_parser.c - find_closing_quotes() added to sip_utils.h
Logic Changes - get_name_and_number() now uses parse_uri() and
get_calleridname() for parsing. Before this change only names
within quotes were found, when names not within quotes are
possible. New Unit Tests -sip_get_name_and_number_test
-sip_get_in_brackets_test (closes issue #16707) Reported by:
Nick_Lewis Patches: issue16706.diff uploaded by dvossel (license
671) Review: https://reviewboard.asterisk.org/r/499/
2010-02-12 23:32 +0000 [r246420-246546] David Vossel <dvossel@digium.com>
* main/channel.c, /: Merged revisions 246545 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r246545 | dvossel | 2010-02-12 17:30:17 -0600 (Fri, 12 Feb 2010)
| 16 lines lock channel during datastore removal On channel
destruction the channel's datastores are removed and destroyed.
Since there are public API calls to find and remove datastores on
a channel, a lock should be held whenever datastores are removed
and destroyed. This resolves a crash caused by a race condition
in app_chanspy.c. (closes issue #16678) Reported by:
tim_ringenbach Patches: datastore_destroy_race.diff uploaded by
tim ringenbach (license 540) Tested by: dvossel ........
* channels/chan_sip.c: fixes areas where port should be removed
from domain during parsing A patch was committed recently that
converted duplicate uri parsing code to use the parse_uri
function. There were two instances where this conversion did not
mimic previous behavior exactly because the port was not being
parsed off the end of the domain. In order to do this, a dummy
pointer argument needs to be passed into parse_uri so it will
know it must parse out the port from the domain. If a port output
paramenter is not present, the domain is returned with the port
still attached.
2010-02-12 08:30 +0000 [r246382] TransNexus OSP Development <support@transnexus.com>
* apps/app_osplookup.c, UPGRADE.txt, CHANGES: Updated doc for OSP
lookup application.
2010-02-11 21:57 +0000 [r246299-246338] David Vossel <dvossel@digium.com>
* tests/test_heap.c, tests/test_event.c,
channels/sip/reqresp_parser.c, channels/sip/config_parser.c:
fixes some test description formatting inconsistencies so log
file looks nice
* tests/test_astobj2.c (added), main/astobj2.c: astobj2 unit test
and bug fix A bug was discovered during the creation of the
astobj2 unit test. When OBJ_MULTIPLE | OBJ_UNLINK is used, the
objects being returned had a ref count issue. This patch resolves
that. Review: https://reviewboard.asterisk.org/r/496/
2010-02-10 23:19 +0000 [r246260] Russell Bryant <russell@digium.com>
* include/asterisk/event.h, tests/test_event.c (added),
main/event.c: Add a test module for the event API, test_event.c.
This module includes a single test so far that creates events
using two different methods and does some verification on the
result to make sure the correct data can be retrieved from the
event that was created. One bug was found in the event API while
developing this test, which makes me happy. :-) Review:
https://reviewboard.asterisk.org/r/495/
2010-02-10 23:13 +0000 [r246249] David Vossel <dvossel@digium.com>
* channels/sip/reqresp_parser.c,
channels/sip/include/reqresp_parser.h: additional parse_uri test
and documentation
2010-02-10 21:55 +0000 [r246200-246208] Tilghman Lesher <tlesher@digium.com>
* res/res_pktccops.exports (added): res_pktccops needs to be able
to export a symbol for chan_mgcp (closes issue #16782) Reported
by: nahuelgreco Patches: res_pktccops.exports uploaded by
nahuelgreco (license 162)
* funcs/func_strings.c: Fussy compiler on another machine...
* funcs/func_strings.c: Fix weird issue with unit tests on
optimized build - turned out to be a signing issue.
2010-02-10 17:49 +0000 [r246116] David Vossel <dvossel@digium.com>
* /, apps/app_queue.c: Merged revisions 246115 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r246115 | dvossel | 2010-02-10 11:44:20 -0600 (Wed, 10 Feb 2010)
| 8 lines fixes random deadlock in app_queue with use_weight
during reload (closes issue #16677) Reported by: tim_ringenbach
Patches: app_queue_use_weight_deadlock.diff uploaded by tim
ringenbach (license 540) ........
2010-02-10 16:47 +0000 [r246070] Jeff Peeler <jpeeler@digium.com>
* channels/chan_local.c: Change channel state on local channels for
busy,answer,ring. Previously local channels channel state never
changed. This became problematic when the state of the other side
of the local channel was lost, for example during a masquerade.
Changing the state of the local channel allows for the scenario
to be detected when the channel state is set to ringing, but the
peer isn't ringing. The specific problem scenario is described in
164201. Although this was noted on one of the issues, here is the
tested dialplan verified to work: exten =>
9700,1,Dial(Local/*9700@default&Local/0009700@default) exten =>
*9700,1,Set(GLOBAL(TESTCHAN)=${CHANNEL:0:${MATH(${LEN(${CHANNEL})}-1):0:2}}1)
exten => *9700,n,wait(3) ;3 works, 1 did not exten =>
*9700,n,Dial(SIP/5001) exten => 0009700,1,Wait(1) ;1 works, 3 did
not exten =>
0009700,n,ChannelRedirect(${TESTCHAN},parkedcalls,701,1) (closes
issue #14992) Reported by: davidw
2010-02-10 16:01 +0000 [r245945-246030] Tilghman Lesher <tlesher@digium.com>
* configure, include/asterisk/autoconfig.h.in, configure.ac,
res/res_agi.c: Solaris doesn't like outputting a NULL to a %s in
format strings. Detect all platforms that don't like that,
either, and ensure that when documentation is missing, we pass a
non-NULL pointer when outputting the corresponding documentation.
(closes issue #16689) Reported by: bklang Patches:
20100209__issue16689__with_tests.diff.txt uploaded by tilghman
(license 14) Review: https://reviewboard.asterisk.org/r/497/
* funcs/func_strings.c: Enable warnings on atypical conditions for
the FILTER function (suggested by mmichelson on the -dev list).
* /, funcs/func_strings.c, configs/extensions.conf.sample: Merged
revisions 245944 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r245944 | tilghman | 2010-02-10 07:37:13 -0600 (Wed, 10 Feb 2010)
| 2 lines Include examples of FILTER usage in extension patterns
where a "." may be a risk. ........
2010-02-09 23:32 +0000 [r245864] Russell Bryant <russell@digium.com>
* include/asterisk/test.h, tests/test_sha1.c (removed),
include/asterisk/utils.h, tests/test_substitution.c,
tests/test_heap.c, tests/test_ast_format_str_reduce.c,
tests/test_skel.c, tests/test_utils.c, funcs/func_math.c,
channels/sip/reqresp_parser.c, main/test.c, tests/test_md5.c
(removed), channels/sip/config_parser.c, tests/test_sched.c:
Various updates to the unit test API. 1) It occurred to me that
the difference in usage between the error ast_str and the
ast_test_update_status() usage has turned out to be a bit
ambiguous in practice. In a lot of cases, the same message was
being sent to both. In other cases, it was only sent to one or
the other. My opinion now is that in every case, I think it makes
sense to do both; we should output it to the CLI as well as save
it off for logging purposes. This change results in most of the
changes in this diff, since it required changes to all existing
unit tests. It also allowed for some simplifications of unit test
API implementation code. 2) Update ast_test_status_update() to
include the file, function, and line number for the code
providing the update. 3) There are some formatting tweaks here
and there. Hopefully they aren't too distracting for code review
purposes. Reviewboard's diff viewer seems to do a pretty good job
of pointing out when something is a whitespace change. 4) I moved
the md5_test and sha1_test into the test_utils module. It seemed
like a better approach since these tests are so tiny. 5) I
changed the number of nodes used in heap_test_2 from 1 million to
100 thousand. The only reason for this was to reduce the time it
took for this test to run. 6) Remove an unused function prototype
that was at the bottom of utils.h. 7) Simplify test_insert()
using the LIST_INSERT_SORTALPHA() macro. The one minor difference
in behavior is that it no longer checks for a test registered
with the same name. 8) Expand the code in test_alloc() to provide
specific error messages for each failure case, to clearly inform
developers if they forget to set the name, summary, description,
etc. 9) Tweak the output of the "test show registered" CLI
command. I swapped the name and category to have the category
first. It seemed more natural since that is the sort key. 10)
Don't output the status ast_str in the "test show results" CLI
command. This is going to tend to be pretty verbose, so just
leave that for the detailed test logs (test generate results).
Review: https://reviewboard.asterisk.org/r/493/
2010-02-09 23:18 +0000 [r245793-245804] David Vossel <dvossel@digium.com>
* channels/chan_iax2.c: fixes a merging error for the iaxs and
iaxsl off by one fix
* /, channels/chan_iax2.c: Merged revisions 245792 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r245792 | dvossel | 2010-02-09 16:55:38 -0600 (Tue, 09
Feb 2010) | 12 lines Fixes iaxs and iaxsl size off by one issue.
2^15 = 32768 which is the maximum allowed iax2 callnumber.
Creating the iaxs and iaxsl array of size 32768 means the maximum
callnumber is actually out of bounds. This causes a nasty crash.
(closes issue #15997) Reported by: exarv Patches: iax_fix.diff
uploaded by dvossel (license 671) ........
2010-02-09 18:06 +0000 [r245729] Tilghman Lesher <tlesher@digium.com>
* apps/app_fax.c: Ensure frames are only freed once. (closes issue
#16361) Reported by: vlad Patches: 20100208__issue16361.diff.txt
uploaded by tilghman (license 14) Tested by: kenny, bloodoff,
misaksen
2010-02-09 17:40 +0000 [r245727] Matthew Nicholson <mnicholson@digium.com>
* channels/chan_sip.c: This commit removes an extra newline in T.38
generated SDP packets. This bug was caused by the fix introduced
in r243860. (closes issue #16766) Reported by: raivisr Patches:
t38-sdp-newline-fix1.diff uploaded by mnicholson (license 96)
Tested by: raivisr
2010-02-09 16:24 +0000 [r245680] Kevin P. Fleming <kpfleming@digium.com>
* apps/app_fax.c: Don't offer MMR or JBIG transcoding during T.38
negotiation. After further discussion with Steve Underwood, we
should not (yet) be offering to receive MMR or JBIG transcoded
streams from T.38 endpoints. A future spandsp release will
support those features, and then they can be enabled during
negotiation
2010-02-08 23:43 +0000 [r245597-245624] Russell Bryant <russell@digium.com>
* main/event.c: Fix return value of get_ie_str() and
get_ie_str_hash() for non-existent IE. I found this bug while
developing a unit test for event allocation. Testing is awesome.
* tests/test_utils.c: UNREGISTER instead of REGISTER in
unload_module().
* main/pbx.c: Use memmove() instead of memcpy() for a case where
the buffers overlap. Once again, valgrind is freaking awesome.
That is all.
* channels/Makefile: Remove object files from the channels/sip/
directory on make clean.
2010-02-08 22:31 +0000 [r245578] Tilghman Lesher <tlesher@digium.com>
* main/Makefile, channels/Makefile: Actually use _ASTLDFLAGS in the
main/ and channels/ Makefiles. They were previously passed
correctly, but they simply weren't used. This caused issues with
various platforms whose builds needed to pass special linker
flags via the configure script. (closes issue #16596) Reported
by: pprindeville Patches: asterisk-1.6-astldflags.patch uploaded
by pprindeville (license 347) Tested by: tilghman
2010-02-08 20:41 +0000 [r245497] Jason Parker <jparker@digium.com>
* /, main/ast_expr2f.c, main/ast_expr2.fl: Merged revisions 245496
via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r245496 | qwell | 2010-02-08 14:39:50 -0600 (Mon, 08 Feb 2010) |
4 lines Remove reference of documentation in source directory.
People don't always build Asterisk from source (distro packages,
anybody?). ........
2010-02-08 04:51 +0000 [r245268-245385] Russell Bryant <russell@digium.com>
* contrib/scripts/install_prereq: Add the libvpb-dev package as a
dependency.
* pbx/pbx_gtkconsole.c: Add a todo for pbx_gtkconsole for updating
to gtk2. This module needs to be converted to gtk2, or we will
eventually have to just remove it from the tree. gtk1 isn't even
packaged anymore in the distro I'm using. I suspect nobody uses
this and that nobody would notice if we removed it.
* contrib/scripts/install_prereq: Add more packages required for
building Asterisk modules.
* channels/chan_usbradio.c: Make chan_usbradio compile.
* tests/test_sha1.c (added): Add a SHA1 test module. Review:
https://reviewboard.asterisk.org/r/492/
* tests/test_md5.c: Remove unnecessary include, ast_md5_hash()
comes from utils.h.
* tests/test_md5.c (added): Add an MD5 test module. Review:
https://reviewboard.asterisk.org/r/491/
* tests/test_ast_format_str_reduce.c: Fix a couple of spelling
errors, and add format module dependencies.
* channels/sip/include/config_parser.h, channels/sip/include/sip.h,
channels/sip/include/sip_utils.h,
channels/sip/include/reqresp_parser.h: Tweak formatting and add
minor updates to some comments.
* main/test.c: Remove an extra space.
2010-02-07 19:51 +0000 [r245230] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Remove parsing of constantssrc from
reload_config. This config option is already handled by the
function handle_common_options and it is unnecessary to parse the
value again.
2010-02-06 14:43 +0000 [r245192] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c, configs/sip.conf.sample: Remove useless sip
options related to hash table size. First off, these options
weren't actually doing anything. By the time the options were
parsed, the peer and dialog containers had already been allocated
with their default values. Second, hash table size is something
that doesn't really make sense to change in a config file. If a
user is that interested in changing the hashtable size, he can
modify the source itself. I have removed the parsing of the
hash_peer, hash_user, and hash_dialog options. I have removed the
hash_user_size variable altogether since it is not used at all. I
also changed hash_peer_size and hash_dialog_size to be constant,
and have changed the symbols to be in all caps as constants
typically are. I have also removed the entire section in
sip.conf.sample regarding configurable hashtable sizes.
2010-02-05 21:21 +0000 [r245147] David Vossel <dvossel@digium.com>
* include/asterisk/astobj2.h, main/astobj2.c: fixes astobj2
unlinking of multiple objects when OBJ_MULTIPLE was disabled When
OBJ_MULTIPLE was off but OBJ_UNLINK was on, all the items in a
bucket were being unlinked instead of just the first match. This
fixes that. Review: https://reviewboard.asterisk.org/r/490/
2010-02-05 19:26 +0000 [r245090] Jeff Peeler <jpeeler@digium.com>
* /, LICENSE, contrib/firmware (removed): Merged revisions 245044
via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r245044 | kpfleming | 2010-02-05 12:32:29 -0600 (Fri, 05 Feb
2010) | 5 lines Remove contrib/firmware directory as it is empty
Remove explicit license for IAXy firmware as it is no longer
included in the tree ........
2010-02-05 19:07 +0000 [r245046] Tilghman Lesher <tlesher@digium.com>
* tests/test_ast_format_str_reduce.c, main/file.c: Merge tests that
verify the same thing. (Oops.)
2010-02-05 18:12 +0000 [r245006] David Vossel <dvossel@digium.com>
* channels/chan_iax2.c: adds total call numbers available to 'iax2
show callnumber usage' cli output
2010-02-05 17:20 +0000 [r244945] Terry Wilson <twilson@digium.com>
* res/res_calendar_exchange.c, res/res_calendar_icalendar.c,
res/res_calendar_caldav.c: Fix crash on 32-bit for users not
using https (closes issue #16778) Reported by: pitel Patches:
diff.txt uploaded by twilson (license 396) Tested by: twilson,
pitel
2010-02-05 17:05 +0000 [r244927] Sean Bright <sean@malleable.com>
* /, main/asterisk.c: Merged revisions 244926 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r244926 | seanbright | 2010-02-05 12:03:35 -0500 (Fri, 05 Feb
2010) | 1 line Update main copyright date. ........
2010-02-05 16:59 +0000 [r244769-244924] David Vossel <dvossel@digium.com>
* channels/chan_sip.c, channels/sip/include/config_parser.h,
channels/sip/config_parser.c: fixes issue with sip registry not
having correct default expiry default expiry was not being set
correctly for a registry object. Thanks to ebroad for reporting
the issue and testing the patch.
* main/astobj2.c: fixes memory leak in astobj2 test
ao2_iterator_destroy was not being used on the iterator during
the test. This resulted in the container never actually being
destroyed.
* channels/chan_sip.c: parse_moved_contact tries to parse
contact_name twice parse_moved_contact attempts to remove a
quoted string twice, and the first try wasn't even being done
correctly.
2010-02-04 22:43 +0000 [r244728-244768] Tilghman Lesher <tlesher@digium.com>
* main/file.c: Try to make ast_format_str_reduce fail...
* include/asterisk/manager.h: Oops
* include/asterisk/manager.h: Define a small set of constant return
values
2010-02-04 15:36 +0000 [r244688] David Vossel <dvossel@digium.com>
* main/test.c: fix truncated format string in 'test show
registered' When using the 'test show registered' cli command the
'Test Results' category was truncating the last few characters
making it look like 'Test Resul'. I also expanded other parts of
the format to better represent how long function names and
categories will likely be.
2010-02-04 00:12 +0000 [r244647] Richard Mudgett <rmudgett@digium.com>
* channels/sip: Add ignore *.i files property to the new
channels/sip directory.
2010-02-03 20:48 +0000 [r244598] Jeff Peeler <jpeeler@digium.com>
* main/features.c, CHANGES: Add some additional option support for
non-default parking lots. The options are: parkedcallparking,
parkedcallhangup, parkedcallrecording, and parkedcalltransfers.
Previously these options were only available for the default
parking lot. (closes issue #16641) Reported by: bluecrow76
Patches: asterisk-1.6.2.1-features.c.diff uploaded by bluecrow76
(license 270)
2010-02-03 20:33 +0000 [r244597] David Vossel <dvossel@digium.com>
* channels/chan_sip.c, channels/sip/include/config_parser.h
(added), channels/sip/reqresp_parser.c (added), channels/sip
(added), channels/Makefile, channels/sip/config_parser.c (added),
channels/sip/include (added), channels/sip/include/sip.h (added),
channels/sip/include/sip_utils.h (added),
channels/sip/include/reqresp_parser.h (added): -----Changes -----
New files - channels/sip/sip.h – A new header for shared #define,
enum, and struct definitions. - channels/sip/include/sip_utils.h
– sip util functions shared among the all the sip APIs -
channels/sip/include/config_parser.h – sip config-parser API -
channels/sip/config_parser.c – Contains sip.conf parsing helper
functions with unit tests. -
channels/sip/include/reqresp_parser.h – sip request response
parser API - channels/sip/reqresp_parser.c – Contains sip request
and response parsing helper functions with unit tests. New Unit
Tests - sip_parse_uri_test - sip_parse_host_test -
sip_parse_register_line_test Code Refactoring - All reusable
#define, enum, and struct definitions were moved out of
chan_sip.c into sip.h. During this process formatting changes
were made to comments in both sip.h and chan_sip.c in order to
better adhere to the coding guidelines. - The beginnings of three
new sip APIs, sip-utils.h, config-parser.h, reqresp-parser.h
using existing chan_sip.c functions. - parse_uri() and
get_calleridname() were moved from chan_sip.c to request-parser.c
along with unit tests for both functions. - sip_parse_host() and
sip_parse_register_line() were moved from chan_sip.c to
config-parser.c along with unit tests for both functions. Changes
to parse_uri() -removal of the options parameter. It was never
used and did not behave correctly. -additional check for
[?header] field. When this field was present, the transport type
was not being set correctly. ----- Overview ----- This patch is
introduced with the hope that unit tests for all our sip parsing
functions will be written soon. chan_sip is a huge file, and with
the addition of each unit test chan_sip is going to grow larger
and harder to maintain. I'm proposing we begin refactoring
chan_sip, starting with the parsing functions. With each parsing
function we move into a separate helper file, a unit test should
accompany it. I've attempted to lay down the ground work for this
change by creating two new parser helper files (config-parser.c
and reqresp-parser.c) and moving all shared structs, enums, and
defines from chan_sip.c into a shared sip.h file. We can't verify
everything in Asterisk using unit tests, but string parsing is
one area where unit tests make the most sense. By beginning to
restructure the code in this way, chan_sip not only becomes less
bloated, but Asterisk as a whole will become more stable. Review:
https://reviewboard.asterisk.org/r/477/
2010-02-03 19:26 +0000 [r244547] Mark Michelson <mmichelson@digium.com>
* main/sched.c: Initialize counters in ast_sched_report so that
resulting data is not bogus.
2010-02-03 18:34 +0000 [r244505] Tilghman Lesher <tlesher@digium.com>
* channels/chan_dahdi.c: The chanvar= setting should inherit the
entire list of variables, not just the first one. (closes issue
#16359) Reported by: raarts Patches: dahdi-setvars.diff uploaded
by raarts (license 937) Tested by: raarts
2010-02-02 22:27 +0000 [r244443] David Vossel <dvossel@digium.com>
* main/udptl.c, channels/chan_sip.c, include/asterisk/udptl.h:
fixes crash during T.38 negotiation caused by invalid or missing
FaxMaxDatagram field AST-2010-001 (closes issue #16634) Reported
by: krn (closes issue #16724) Reported by: barthpbx (closes issue
#16517) Reported by: bklang (closes issue #16485) Reported by:
elsto
2010-02-02 20:32 +0000 [r244071-244393] Tilghman Lesher <tlesher@digium.com>
* apps/app_dial.c, CHANGES: Properly respect GOSUB_RESULT as to
what to do with the master channel. Previously, we would parse
GOSUB_RESULT, but not actually do anything with it. Also, allow
GOSUB_RETVAL to be inherited back across a peer/master channel.
(closes issue #16687) Reported by: bklang Patches:
app_dial-preserve-gosub_retval.patch uploaded by bklang (license
919) (with modifications) (closes issue #16686) Reported by:
bklang Patches: app_dial-respect-gosub_result.patch uploaded by
bklang (license 919) (with modifications)
* funcs/func_math.c: Correct some off-by-one errors, especially
when expressions don't contain expected spaces. Also include the
tests provided by the reporter, as regression tests. (closes
issue #16667) Reported by: wdoekes Patches:
astsvn-func_match-off-by-one.diff uploaded by wdoekes (license
717)
* /, apps/app_voicemail.c: Merged revisions 244242 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r244242 | tilghman | 2010-02-01 17:13:44 -0600 (Mon, 01
Feb 2010) | 11 lines Backup and restore original textfile, for
prosthesis (gerund of prepend). Also, fix menuselect such that
changing voicemail build options correctly causes rebuild.
(closes issue #16415) Reported by: tomo1657 Patches:
prepention.patch uploaded by tomo1657 (license 484) (with
modifications by me to backport to 1.4) ........
* main/channel.c, channels/chan_local.c, /: Merged revisions 244070
via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r244070 | tilghman | 2010-02-01 11:46:31 -0600 (Mon, 01 Feb 2010)
| 16 lines Revert previous chan_local fix (r236981) and fix
instead by destroying expired frames in the queue. (closes issue
#16525) Reported by: kobaz Patches: 20100126__issue16525.diff.txt
uploaded by tilghman (license 14)
20100129__issue16525__1.6.0.diff.txt uploaded by tilghman
(license 14) Tested by: kobaz, atis (closes issue #16581)
Reported by: ZX81 (closes issue #16681) Reported by: alexr1
........
2010-01-28 22:37 +0000 [r243986] Jeff Peeler <jpeeler@digium.com>
* main/manager.c: Optimization to manager events. When potentially
sending manager events, return immediately if there are no
sessions or hooks. Also, avoid locking the hooks list if it is
empty. (issue #16455) Reported by: atis Patches:
manager_hooks_trunk.patch uploaded by atis (license 242)
2010-01-28 20:00 +0000 [r243943] Tilghman Lesher <tlesher@digium.com>
* channels/iax2-parser.c: Informational message, not an error.
2010-01-28 18:35 +0000 [r243780-243860] Russell Bryant <russell@digium.com>
* channels/chan_sip.c: Add a missing line terminator for T.38 SDP.
* /, channels/chan_sip.c: Merged revisions 243779 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r243779 | russell | 2010-01-28 09:03:17 -0600 (Thu, 28 Jan 2010)
| 2 lines Fix a bogus third argument to ast_copy_string().
........
2010-01-27 20:37 +0000 [r243551-243693] Jeff Peeler <jpeeler@digium.com>
* /, apps/app_queue.c: Merged revisions 243691 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r243691 | jpeeler | 2010-01-27 14:35:56 -0600 (Wed, 27 Jan 2010)
| 5 lines Revert 243570, I should have looked at this closer.
Will reopen the issue, but am leaving the review closed as the
change was pointless. (issue #16488) ........
* CHANGES: expand code based appreviation of AST_CONFIG_DIR to
configuration directory
* /, apps/app_queue.c: Merged revisions 243570 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r243570 | jpeeler | 2010-01-27 12:47:34 -0600 (Wed, 27 Jan 2010)
| 9 lines Extend announcement URL used with Queue from 80 chars
to PATH_MAX. (closes issue #16488) Reported by: syspert Patches:
soundfilelen.pacth-2 uploaded by syspert (license 938) Review:
https://reviewboard.asterisk.org/r/475/ ........
* Makefile, CHANGES, include/asterisk/options.h, main/asterisk.c,
main/loader.c: Add new option to asterisk.conf (lockconfdir) to
protect conf dir during reloads (closes issue #16358) Reported
by: raarts Patches: lockconfdir.diff uploaded by raarts (license
937) modified by me
2010-01-27 18:08 +0000 [r243487] Mark Michelson <mmichelson@digium.com>
* main/pbx.c, /: Merged revisions 243486 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r243486 | mmichelson | 2010-01-27 12:06:43 -0600 (Wed, 27 Jan
2010) | 3 lines Use a safe list traversal while checking for
duplicate vars in pbx_builtin_setvar_helper. ........
2010-01-27 17:32 +0000 [r243482] Russell Bryant <russell@digium.com>
* funcs/func_channel.c, channels/chan_iax2.c: Fix the ability to
specify an OSP token for an outbound IAX2 call. When this patch
was originally submitted, the code allowed for the token to be
set via a channel variable. I decided that a cleaner approach
would be to integrate it into the CHANNEL() function.
Unfortunately, that is not a suitable approach. It's not possible
to get the value set on the channel soon enough using that
method. So, go back to the simple channel variable method.
(closes issue #16711) Reported by: homesick Patches: iax-svn.diff
uploaded by homesick (license 91)
2010-01-26 23:56 +0000 [r243391] David Vossel <dvossel@digium.com>
* /, main/features.c: Merged revisions 243390 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r243390 | dvossel | 2010-01-26 17:55:49 -0600 (Tue, 26 Jan 2010)
| 9 lines fixes bug with channel receiving wrong privileges after
call parking (closes issue #16429) Reported by: Yasuhiro Konishi
Patches: features.c.diff uploaded by Yasuhiro Konishi (license
947) Tested by: dvossel ........
2010-01-26 20:49 +0000 [r243346] David Ruggles <thedavidfactor@gmail.com>
* apps/app_senddtmf.c: Code clean up in app_senddtmf Pushes code
clean up done in app_externalivr back into app_senddtmf Review:
https://reviewboard.asterisk.org/r/473/
2010-01-26 18:20 +0000 [r243244-243266] Jeff Peeler <jpeeler@digium.com>
* main/channel.c, /: Merged revisions 243258 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r243258 | jpeeler | 2010-01-26 12:19:10 -0600 (Tue, 26 Jan 2010)
| 2 lines Remove unnecessary code in ast_read as issue 16058 has
been fully solved now. ........
* main/frame.c: Fix crash resulting from frames with invalid data
pointers. In ast_frdup the frame data union does not get set to
point to malloced memory if the datalen is zero, so make sure to
handle the same case in ast_frisolate appropriately. (closes
issue #16058) Reported by: atis Patches: bug16058-fix.patch
uploaded by jpeeler (license 325) Tested by: atis
2010-01-26 17:40 +0000 [r243200-243242] David Vossel <dvossel@digium.com>
* main/test.c: modify 'test show registered' cli output format In
order to improve readability, the output from 'test show
registered' has been modified to truncate fields to fit within
the format output if they are over a certain length.
* include/asterisk/utils.h, channels/chan_sip.c, tests/test_utils.c
(added), main/test.c, main/utils.c: RFC compliant uri and
display-name encode/decode 1. URI Encoding This patch changes
ast_uri_encode()'s behavior when doreserved is enabled.
Previously when doreserved was enabled only a small set of
reserved characters were encoded. This set was comprised
primarily of the reserved characters defined in RFC3261 section
25.1, but contained other characters as well. Rather than only
escaping the reserved set, doreserved now escapes all characters
not within the unreserved set as defined by RFC 3261 and RFC
2396. Also, the 'doreserved' variable has been renamed to
'do_special_char' in attempts to avoid confusion. When doreserve
is not enabled, the previous logic of only encoding the
characters <= 0X1F and > 0X7f remains, except for the '%'
character, which must always be encoded as it signifies a HEX
escaped character during the decode process. 2. URI Decoding:
Break up URI before decode. In chan_sip.c ast_uri_decode is
called on the entire URI instead of it's individual parts after
it is parsed. This is not good as ast_uri_decode can introduce
special characters back into the URI which can mess up parsing.
This patch resolves this by not decoding a URI until parsing is
completely done. There are many instances where we check to see
if pedantic checking is enabled before we decode a URI. In these
cases a new macro, SIP_PEDANTIC_DECODE, is used on the individual
parsed segments of the URI rather than constantly putting if
(pedantic) { decode() } checks everywhere in the code. In the
areas where ast_uri_decode is not dependent upon pedantic
checking this macro is not used, but decoding is still moved to
each individual part of the URI. The only behavior that should
change from this patch is the time at which decoding occurs.
Since I had to look over every place URI parsing occurs to create
this patch, I found several places where we use duplicate code
for parsing. To consolidate the code, those areas have updated to
use the parse_uri() function where possible. 3. SIP display-name
decoding according to RFC3261 section 25. To properly decode the
display-name portion of a FROM header, chan_sip's
get_calleridname() function required a complete re-write. More
information about this change can be found in the comments at the
beginning of this function. 4. Unit Tests. Unit tests for
ast_uri_encode, ast_uri_decode, and get_calleridname() have been
written. This involved the addition of the test_utils.c file for
testing the utils api. (closes issue #16299) Reported by: wdoekes
Patches: astsvn-16299-get_calleridname.diff uploaded by wdoekes
(license 717) get_calleridname_rewrite.diff uploaded by dvossel
(license 671) Tested by: wdoekes, dvossel, Nick_Lewis Review:
https://reviewboard.asterisk.org/r/469/
2010-01-26 15:46 +0000 [r243118-243158] Russell Bryant <russell@digium.com>
* tests/test_substitution.c: Log the variable name being tested.
* tests/test_substitution.c: Update test_substitution to show
failures in the test log.
* funcs/func_aes.c: Update func_aes to its pre-ast_str_substitution
state. This change makes the AES tests in test_substitution.c
pass. We still need to work through what's going wrong in the
ast_str version.
2010-01-26 01:56 +0000 [r242967-243077] Tilghman Lesher <tlesher@digium.com>
* tests/test_substitution.c: Fixing last errors in the conversion,
though it appears that the AES_* functions are still broken.
* tests/test_substitution.c: Using a dummy channel causes CDR()
testing to fail.
* tests/test_substitution.c: Wish I had gotten to the review before
this got submitted, because there's failures we need to address.
* /, main/Makefile, res/Makefile: Merged revisions 242969 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r242969 | tilghman | 2010-01-25 15:50:22 -0600 (Mon, 25 Jan 2010)
| 2 lines Err, and use the new menuselect define, too. ........
* build_tools/cflags.xml, /, build_tools/menuselect-deps.in,
configure, configure.ac: Merged revisions 242966 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r242966 | tilghman | 2010-01-25 15:36:33 -0600 (Mon, 25
Jan 2010) | 2 lines Only rebuild parsers by an option in
menuselect ........
2010-01-25 21:32 +0000 [r242954-242965] Russell Bryant <russell@digium.com>
* tests/test_substitution.c, tests/test_heap.c,
tests/test_ast_format_str_reduce.c, tests/test_skel.c,
tests/test_sched.c: Make unit test modules depend on
TEST_FRAMEWORK instead of off by default.
* tests/test_substitution.c: Convert test_substitution module to
the unit test API. Review:
https://reviewboard.asterisk.org/r/474/
2010-01-25 21:20 +0000 [r242933] Alexandr Anikin <may@telecom-service.ru>
* addons/ooh323c/src/ooh323.c, addons/ooh323c/src/oochannels.c,
addons/ooh323c/src/ooCalls.c: small corrections in call clearing
2010-01-25 21:13 +0000 [r242904-242919] Olle Johansson <oej@edvina.net>
* main/pbx.c, main/manager.c, include/asterisk/pbx.h: Change api
for pbx_builtin_setvar to actually return error code if a
function can't be written to. This patch removes code that was
duplicated from pbx.c to manager.c in order to prevent API change
in released versions of Asterisk. There are propably also other
places that would benefit from reading the return code and react
if a function returns error codes on writing a value into it.
* main/manager.c, /: Merged revisions 242850 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r242850 | oej | 2010-01-25 21:03:38 +0100 (Mån, 25 Jan 2010) | 2
lines Report error when writing to functions returns error in AMI
setvar action ........
2010-01-25 20:18 +0000 [r242857] Tilghman Lesher <tlesher@digium.com>
* /, configure, main/Makefile, configure.ac, res/Makefile: Merged
revisions 242852 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r242852 | tilghman | 2010-01-25 14:15:45 -0600 (Mon, 25 Jan 2010)
| 2 lines Restore FreeBSD to able-to-compile-ish-mode ........
2010-01-25 18:01 +0000 [r242812] Terry Wilson <twilson@digium.com>
* res/res_calendar.c: Fix INTERNAL_OBJ error on stop when
calendars.conf missing Initialize the calendars container before
calling load_config and return FAILURE on allocation failure.
Also, use the AST_MODULE_LOAD_* values for return values. Thanks
to rmudgett for pointing out the error and the need to use the
defined values for return
2010-01-25 05:45 +0000 [r242719-242729] Tilghman Lesher <tlesher@digium.com>
* /, main/Makefile, res/Makefile: Merged revisions 242728 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r242728 | tilghman | 2010-01-24 23:42:22 -0600 (Sun, 24 Jan 2010)
| 2 lines Buildbot pointed out an error (thanks, buildbot!)
........
* /, res/Makefile: Merged revisions 242723 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r242723 | tilghman | 2010-01-24 23:33:37 -0600 (Sun, 24 Jan 2010)
| 2 lines Oops, should have used CMD_PREFIX, not ECHO_PREFIX, for
the commands. ........
* /, main/Makefile: Merged revisions 242683 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r242683 | tilghman | 2010-01-24 23:13:28 -0600 (Sun, 24 Jan 2010)
| 2 lines Make the build of the Asterisk expression parser match
that of the AEL parser. ........
2010-01-24 22:42 +0000 [r242645] Alexandr Anikin <may@telecom-service.ru>
* addons/ooh323c/src/ooh323.c, addons/chan_ooh323.c,
addons/ooh323c/src/ooStackCmds.h,
addons/ooh323c/src/oochannels.c,
addons/ooh323c/src/ooCmdChannel.c,
addons/ooh323c/src/ooStackCmds.c: AST_CONTROL_CONNECTED_LINE
frame type processing added to setup DisplayIE field incorrect
q.931 message order filtered on incoming calls (first msg must be
setup, next must be not setup)
2010-01-24 21:49 +0000 [r242607] Sean Bright <sean@malleable.com>
* res/res_phoneprov.c: Instead of crashing, allocate our header
ast_str before we try to use it. (closes issue #16680) Reported
by: lmadsen Patches: issue16680_20100122.patch uploaded by
seanbright (license 71) Tested by: lmadsen
2010-01-24 06:40 +0000 [r242521] Tilghman Lesher <tlesher@digium.com>
* /, configure, include/asterisk/autoconfig.h.in, configure.ac,
pbx/Makefile, res/Makefile, makeopts.in: Merged revisions 242520
via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r242520 | tilghman | 2010-01-24 00:33:01 -0600 (Sun, 24 Jan 2010)
| 8 lines Only rebuild bison and flex source files on demand, if
bison and flex are detected by the configure script. Changed
after discussion on the -dev list about possible unnecessary
build failures, due to checkouts/untars causing these special
source files to possibly be newer than their resulting C files.
This should additionally ensure that nobody need learn about
extra Makefile arguments to ensure the proper files get rebuilt
when changes are made to these special source files. ........
2010-01-22 21:45 +0000 [r242424] Tilghman Lesher <tlesher@digium.com>
* /, res/Makefile: Merged revisions 242423 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r242423 | tilghman | 2010-01-22 15:44:18 -0600 (Fri, 22 Jan 2010)
| 7 lines Rebuild from flex, bison sources when necessary. (issue
#14629) Reported by: Marquis Patches:
20100121__issue14629.diff.txt uploaded by tilghman (license 14)
........
2010-01-22 16:20 +0000 [r242357] David Ruggles <thedavidfactor@gmail.com>
* apps/app_externalivr.c: Add send DTMF feature to ExternalIVR app
Implemented a new command 'D' that allows client IVRs to send
DTMF digits to the channel. (closes issue #16615) Reported by:
thedavidfactor Review: https://reviewboard.asterisk.org/r/465/
2010-01-22 15:09 +0000 [r242317] Tilghman Lesher <tlesher@digium.com>
* tests/test_sched.c: The irony of not compile-testing a test
program before committing is killing me.
2010-01-22 09:28 +0000 [r242227] Olle Johansson <oej@edvina.net>
* /, channels/chan_sip.c: Merged revisions 242226 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r242226 | oej | 2010-01-22 10:19:30 +0100 (Fre, 22 Jan 2010) | 3
lines Initialize notify_types to NULL ........
2010-01-22 04:57 +0000 [r242184-242186] Russell Bryant <russell@digium.com>
* main/test.c: Update the doxygenification of some comments.
* tests/test_sched.c: Convert scheduler API entry order test to the
test API. Review: https://reviewboard.asterisk.org/r/470/
* tests/test_skel.c: Add test API usage example to test_skel.c.
Review: https://reviewboard.asterisk.org/r/471/
2010-01-21 22:37 +0000 [r242092] Mark Michelson <mmichelson@digium.com>
* main/acl.c: Add missing argument to ast_calloc calls.
2010-01-21 21:05 +0000 [r242043] Olle Johansson <oej@edvina.net>
* main/acl.c: Make sure we initialize the ast_ha structure with
ast_calloc
2010-01-21 15:27 +0000 [r241938] Sean Bright <sean@malleable.com>
* /, configure, configure.ac: Merged revisions 241932 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r241932 | seanbright | 2010-01-21 10:25:46 -0500 (Thu,
21 Jan 2010) | 5 lines Fix configure check for PTHREAD_ONCE_INIT
when manually adding -Wall to CFLAGS. (closes issue #16666)
Reported by: romain_proformatique ........
2010-01-21 15:14 +0000 [r241896] Tilghman Lesher <tlesher@digium.com>
* channels/chan_vpb.cc: Formats are inconsistent between even
32-bit and 64-bit Linux. Use casts to ensure both compile.
2010-01-21 14:10 +0000 [r241855-241856] Russell Bryant <russell@digium.com>
* main/test.c: Point to a useful reference on the XML output
format.
* main/test.c: Modify test results XML format to match the JUnit
format. When this code was developed, we came up with our own XML
format for the test output. I have since started looking at
integration with other tools, namely continuous integration
frameworks, and this format seems to be supported across a number
of applications. With these changes in place, I was able to get
Atlassian Bamboo to interpret the test results.
2010-01-21 05:54 +0000 [r241766] Tilghman Lesher <tlesher@digium.com>
* /, funcs/func_math.c: Merged revisions 241765 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r241765 | tilghman | 2010-01-20 23:53:17 -0600 (Wed, 20 Jan 2010)
| 2 lines Guard against division by zero. ........
2010-01-20 21:14 +0000 [r241627-241714] David Vossel <dvossel@digium.com>
* res/res_rtp_asterisk.c: rtp timestamp to timeval calculation fix
The rtp timestamp to timeval calculation was only accurate for
8kHz audio. This patch corrects this. Review:
https://reviewboard.asterisk.org/r/468/ SWP-648
* Makefile, /: Merged revisions 241626 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r241626 | dvossel | 2010-01-20 14:00:04 -0600 (Wed, 20 Jan 2010)
| 6 lines fixes parsing error in Makefile. Some echo lines were
missing "; . Thanks to jparker for pointing out the problem.
........
2010-01-20 17:49 +0000 [r241581] Alec L Davis <sivad.a@paradise.net.nz>
* main/cdr.c: Add Calling and Called Subaddress to CDR record
Requires 'callingsubaddr' and 'calledsubaddr' fields in backend
cdr. (closes issue #16600) Reported by: alecdavis Patches:
cdr_subaddr.diff.txt uploaded by alecdavis (license 585) Tested
by: alecdavis Review: https://reviewboard.asterisk.org/r/460/
2010-01-20 13:01 +0000 [r241503] Kevin P. Fleming <kpfleming@digium.com>
* channels/chan_vpb.cc: Fix up compile breakage from
ast_tvdiff_ms() API change.
2010-01-20 08:18 +0000 [r241416] Alec L Davis <sivad.a@paradise.net.nz>
* main/pbx.c, channels/sig_pri.c: Update CDR variables as pbx
starts Allows CDR variables added in cdr.c:set_one_cid to become
visable during the call, by executing ast_cdr_update() early in
__ast_pbx run. Reverts sig_pri changes in trunk that are specific
to isdn technology only. (closes issue #16638) Reported by:
alecdavis Patches: cdr_update.diff3.txt uploaded by alecdavis
(license 585) Tested by: alecdavis
2010-01-19 22:59 +0000 [r241366] Jeff Peeler <jpeeler@digium.com>
* main/pbx.c: Initialize data on the stack so that Park doesn't
interpret random arguments. passdata was only being set in
pbx_substitue_variables when arguments were passed. (closes issue
#16406) (closes issue #16586) Reported by: DLNoah Patches:
bug16586v2.patch uploaded by jpeeler (license 325) Tested by:
DLNoah
2010-01-19 22:41 +0000 [r241364] Tilghman Lesher <tlesher@digium.com>
* doc/janitor-projects.txt, apps/app_sendtext.c: Enable SendText to
send strings in encoded format. See
http://lists.digium.com/pipermail/asterisk-users/2010-January/243462.html
2010-01-19 18:51 +0000 [r241314-241315] Jeff Peeler <jpeeler@digium.com>
* channels/chan_agent.c: small correction from 241314
* /, channels/chan_agent.c: Merged revisions 241227 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r241227 | jpeeler | 2010-01-19 11:22:18 -0600 (Tue, 19
Jan 2010) | 13 lines Fix deadlock in agent_read by removing call
to agent_logoff. One must always lock the agents list lock before
the agent private. agent_read locks the private immediately, so
locking the agents list lock is not an option (which is what
agent_logoff requires). Because agent_read already has access to
the agent private all that is necessary is to do the required
hanging up that agent_logoff performed. (closes issue #16321)
Reported by: valon24 Patches: bug16321.patch uploaded by jpeeler
(license 325) ........
2010-01-19 17:42 +0000 [r241230] Jason Parker <jparker@digium.com>
* Makefile: Allow parallel make (-j) to work properly. After some
back and forth with the reporter, we came up with the necessary
changes. (closes issue #16489) Reported by: Chainsaw Patches:
asterisk-1.6.2.1-parallel-make-minimal.patch uploaded by Chainsaw
(license 723) Tested by: Chainsaw, qwell
2010-01-19 00:28 +0000 [r241188] Tilghman Lesher <tlesher@digium.com>
* main/srv.c, res/res_agi.c, CHANGES, include/asterisk/srv.h:
Create iterative method for querying SRV results, and use that
for finding AGI servers. (closes issue #14775) Reported by:
_brent_ Patches: 20091215__issue14775.diff.txt uploaded by
tilghman (license 14) hagi-5.patch uploaded by brent (license
388) Tested by: _brent_ Reviewboard:
https://reviewboard.asterisk.org/r/378/
2010-01-19 00:24 +0000 [r241187] Alec L Davis <sivad.a@paradise.net.nz>
* channels/sig_pri.c: Update CDR variables before pbx starts
(overlap dial) Allows CDR variables added in cdr.c:set_one_cid to
become visable during the call. (issue #16638) Reported by:
alecdavis Patches: cdr_update.diff2.txt uploaded by alecdavis
(license 585) Tested by: alecdavis
2010-01-18 22:31 +0000 [r241143] Jeff Peeler <jpeeler@digium.com>
* main/channel.c, channels/chan_dahdi.c, channels/sig_analog.c,
main/features.c, pbx/pbx_dundi.c, main/enum.c,
include/asterisk/time.h, main/timing.c: Extend max call limit
duration from 24.8 days to 292+ million years. If the limit was
set past MAX_INT upon answering, the call was immediately hung up
due to overflow from the return of ast_tvdiff_ms (in
ast_check_hangup). The time calculation functions ast_tvdiff_sec
and ast_tvdiff_ms have been changed to return an int64_t to
prevent overflow. Also the reporter suggested adding a message
indicating the reason for the call hanging up. Given that the new
limit is so much higher, the message (which would only really be
useful in the overflow scenario) has been made a debug message
only. (closes issue #16006) Reported by: viraptor
2010-01-18 22:03 +0000 [r241098] Jason Parker <jparker@digium.com>
* main/rtp_engine.c: Fix an RTP instance allocation failure on
Solaris. (closes issue #16543) Reported by: crjw Patches:
rtp_sin_family.patch uploaded by crjw (license 963) Tested by:
crjw, qwell
2010-01-18 22:00 +0000 [r241097] Alec L Davis <sivad.a@paradise.net.nz>
* channels/sig_pri.c: Update CDR variables before pbx starts Allows
CDR variables added in cdr.c:set_one_cid to become visable during
the call. (closes issue #16638) Reported by: alecdavis Patches:
cdr_update.diff.txt uploaded by alecdavis (license 585)
2010-01-18 19:57 +0000 [r241016] Sean Bright <sean@malleable.com>
* /, main/config.c: Merged revisions 241015 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r241015 | seanbright | 2010-01-18 14:54:19 -0500 (Mon, 18 Jan
2010) | 12 lines Plug a memory leak when reading configs with
their comments. While reading through configuration files with
the intent of returning their full contents (comments
specifically) we allocated some memory and then forgot to free
it. This doesn't fix 16554 but clears up a leak I had in the lab.
(issue #16554) Reported by: mav3rick Patches:
issue16554_20100118.patch uploaded by seanbright (license 71)
Tested by: seanbright ........
2010-01-18 19:26 +0000 [r241012] Tilghman Lesher <tlesher@digium.com>
* funcs/func_strings.c, CHANGES: Make HASHes inheritable across
channel creation.
2010-01-18 18:00 +0000 [r240973-240974] David Ruggles <thedavidfactor@gmail.com>
* UPGRADE.txt: ExternalIVR information for UPGRADE.txt added a
paragraph about the fixes and changes to the ExternalIVR
application.
* doc/externalivr.txt: Updated ExternalIVR documentation Rewrote a
large portion of the existing documentation and added information
about the TCP/IP socket interface
2010-01-18 17:45 +0000 [r240971] David Vossel <dvossel@digium.com>
* Makefile, CHANGES: transmit_silence_during_record replaced by
transmit_silence In asterisk.conf, transmit_silence_during_record
has been removed in favor of using only the transmit_silence
option. The transmit_silence_during_record option remains a valid
option in asterisk.conf, but has been removed from the sample
config and noted in CHANGES.
2010-01-18 17:41 +0000 [r240969] David Ruggles <thedavidfactor@gmail.com>
* apps/app_externalivr.c: Add notification of interrupted file Add
file information to data element of T event so the file
information is sent to the client when it is interrupted.
Previously only notification of pending files that were dropped
was sent (closes issue #16147) Reported by: thedavidfactor Tested
by: thedavidfactor Review:
https://reviewboard.asterisk.org/r/449/
2010-01-18 16:45 +0000 [r240842-240887] David Vossel <dvossel@digium.com>
* Makefile: updated transmit_silence option documentation in
asterisk.conf This patch updates the transmit_silence option to
better document why the option exists, and what it affects.
Thanks to russell for providing the verbage for this update.
* apps/app_queue.c: fixes spelling error. s/memeber/member
2010-01-17 19:45 +0000 [r240717] Sean Bright <sean@malleable.com>
* main/pbx.c: Avoid a crash on Solaris when running 'core show
functions.' (closes issue #16309) Reported by: asgaroth
2010-01-16 00:54 +0000 [r240667] Sean Bright <sean@malleable.com>
* res/res_musiconhold.c: Get MoH building on OpenSolaris.
2010-01-15 23:50 +0000 [r240629] Tilghman Lesher <tlesher@digium.com>
* Makefile, main/asterisk.c: Err, oops, it was already the way I
intended.
2010-01-15 23:09 +0000 [r240548-240552] Russell Bryant <russell@digium.com>
* include/asterisk/doxygen/commits.h: Note where empty lines should
reside in commit messages.
* Makefile, /: Merged revisions 240547 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r240547 | russell | 2010-01-15 17:06:11 -0600 (Fri, 15 Jan 2010)
| 2 lines Fix a spelling error in the asterisk.conf sample.
........
2010-01-15 22:07 +0000 [r240505] Sean Bright <sean@malleable.com>
* res/res_timing_timerfd.c: Clarify error message in
res_timing_timerfd.
2010-01-15 21:42 +0000 [r240421-240500] Tilghman Lesher <tlesher@digium.com>
* utils/astcanary.c: Oops, missed an include
* utils/astcanary.c, main/asterisk.c: The previous attempt at using
a pipe to guarantee astcanary shutdown did not work. We're
revisiting the previous patch, albeit with a method that
overcomes the prior criticism that it was not POSIX-compliant.
(closes issue #16602) Reported by: frawd Patches:
20100114__issue16602.diff.txt uploaded by tilghman (license 14)
Tested by: frawd
* apps/app_directed_pickup.c, main/features.c,
include/asterisk/manager.h: Add pickup event to AMI. Also, fix
AMI documentation. (closes issue #16431) Reported by: syspert
Patches: 20100112__issue16431.diff.txt uploaded by tilghman
(license 14)
2010-01-15 20:58 +0000 [r240420] Mark Michelson <mmichelson@digium.com>
* main/utils.c: Make sure to set owner_line, ownder_func, and
owner_file in ast_calloc_with_stringfields. Asterisk would crash
on startup if MALLOC_DEBUG were set in menuselect. This is
because the manager action UpdateConfig had to resize its string
field allocation to set the description. When the resize
occurred, ast_copy_string would crash because we were attempting
to copy a string from a NULL pointer. Setting the strings
initially makes the code much less crashy.
2010-01-15 20:58 +0000 [r240415-240419] Tilghman Lesher <tlesher@digium.com>
* apps/app_voicemail.c: Make sure that the limit is N, not N - 1.
* /, apps/app_voicemail.c: Merged revisions 240414 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r240414 | tilghman | 2010-01-15 14:52:27 -0600 (Fri, 15
Jan 2010) | 15 lines Disallow leaving more than maxmsg
voicemails. This is a possibility because our previous method
assumed that no messages are left in parallel, which is not a
safe assumption. Due to the vmu structure duplication, it was
necessary to track in-process messages via a separate structure.
If at some point, we switch vmu to an ao2-reference-counted
structure, which would eliminate the prior noted duplication of
structures, then we could incorporate this new in-process
structure directly into vmu. (closes issue #16271) Reported by:
sohosys Patches: 20100108__issue16271.diff.txt uploaded by
tilghman (license 14) 20100108__issue16271__trunk.diff.txt
uploaded by tilghman (license 14)
20100108__issue16271__1.6.0.diff.txt uploaded by tilghman
(license 14) Tested by: jsutton ........
2010-01-15 20:41 +0000 [r240411] Russell Bryant <russell@digium.com>
* main/event.c: Ensure payload type is properly checked when
comparing against cached events. (closes issue #16607) Reported
by: ddv2005 Patches: event.patch uploaded by ddv2005 (license
769)
2010-01-15 18:21 +0000 [r240368] Sean Bright <sean@malleable.com>
* main/pbx.c, main/manager.c, res/res_smdi.c, apps/app_meetme.c,
channels/chan_sip.c, cel/cel_tds.c, main/features.c,
res/res_phoneprov.c, cdr/cdr_tds.c, apps/app_jack.c: Convert a
few places to use ast_calloc_with_stringfields where applicable.
2010-01-15 16:51 +0000 [r240329] Russell Bryant <russell@digium.com>
* configure: Update configure script for an OSP toolkit related
change.
2010-01-15 16:28 +0000 [r240328] Kevin P. Fleming <kpfleming@digium.com>
* configs/sip.conf.sample: Clarify RTP NAT handling a bit.
2010-01-14 23:13 +0000 [r240226-240271] Sean Bright <sean@malleable.com>
* res/res_config_ldap.c: Plug a memory leak in res_config_ldap.
(closes issue #16257) Reported by: nito Patches:
issue16257_20100111.diff uploaded by seanbright (license 71)
* res/res_timing_timerfd.c: If we aren't running on a machine that
support CLOCK_MONOTONIC, don't load. Group developed and tested
by seanbright, Corydon76, Kobaz, and Amorsen.
2010-01-14 18:03 +0000 [r240179] Jeff Peeler <jpeeler@digium.com>
* main/channel.c: Fix broken call pickup The problem was the
OUTGOING flag was not getting set properly on the channel,
resulting in pickup failing as ast_read thought the call was
inbound. Refer to 170393 for a more verbose description as this
is the same exact change. (closes issue #16539) Reported by:
syspert Patches: bug16539.patch uploaded by jpeeler (license 325)
Tested by: syspert
2010-01-14 17:34 +0000 [r240129-240175] Tilghman Lesher <tlesher@digium.com>
* main/pbx.c: Similarly, ensure that matchcid is duplicated
correctly when merging contexts.
* main/pbx.c: Ensure that the callerid is NULL when the parent is
effectively NULL. This applies only to pattern-match hints, which
create exact-match hints on the fly.
2010-01-14 16:14 +0000 [r240078] Matthew Nicholson <mnicholson@digium.com>
* main/udptl.c: This change fixes a few bugs in the way the far max
IFP was calculated that were introduced in r231692. (closes issue
#16497) Reported by: globalnetinc Patches:
udptl-max-ifp-fix1.diff uploaded by mnicholson (license 96)
Tested by: globalnetinc
2010-01-14 14:38 +0000 [r240039] Leif Madsen <lmadsen@digium.com>
* doc/building_queues.txt (added): Add documentation about how to
build queues. Add a how-to set of documentation about building
queues with Asterisk. This documentation is based on Asterisk
1.6.2 but should work on most versions with minor modifications.
(closes issue #16237) Reported by: lmadsen Patches: Building
Queues (FINAL).txt uploaded by lmadsen (license 10) Tested by:
pdhales, lmadsen, cmdrwalrus
2010-01-13 23:22 +0000 [r239920-239997] Tilghman Lesher <tlesher@digium.com>
* main/pbx.c: Oops, another tag error
* main/pbx.c: Oops, missed a closing tag
* main/pbx.c, include/asterisk/pbx.h: Add the TESTTIME() dialplan
function, which permits testing GotoIfTime. Specifically, by
setting TESTTIME() to a particular date and time, you can test
whether a dialplan correctly branches as was intended. This was
developed after recent questions on the -users list on how to
test their holiday dialplan logic. (closes issue #16464) Reported
by: tilghman Patches: 20100112__issue16464.diff.txt uploaded by
tilghman (license 14) Review:
https://reviewboard.asterisk.org/r/458/
* main/ast_expr2f.c, main/ast_expr2.fl: Flex uses fwrite
incorrectly, which breaks the build. Providing a workaround.
2010-01-13 19:48 +0000 [r239839] Jeff Peeler <jpeeler@digium.com>
* /, main/features.c: Merged revisions 239838 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r239838 | jpeeler | 2010-01-13 13:43:33 -0600 (Wed, 13 Jan 2010)
| 11 lines Fix regression for timed out parked call returning to
caller This issue seems to have been exposed by the fix in 160390
whereby using a masquerade prevented a crash. The new channel
used in the masquerade was not copying the macro information from
the old channel. (closes issue #15459) Reported by: djrodman
Patches: patch_15459.txt uploaded by mnick (license ) ........
2010-01-13 19:31 +0000 [r239834] Leif Madsen <lmadsen@digium.com>
* configs/extensions.conf.sample: Add more examples to
extensions.conf showing how to use various functionality and
provide commonly useful features. (closes issue #16090) Reported
by: pprindeville Patches: extensions.conf-bugid16090.patch#3
uploaded by pprindeville (license 347) Tested by: tzafrir,
pprindeville, lmadsen
2010-01-13 18:16 +0000 [r239797] Tilghman Lesher <tlesher@digium.com>
* main/Makefile, main/ast_expr2f.c, main/ast_expr2.fl: Code
previously added to ast_expr2f.c warranted a change in the source
file ast_expr2.fl. Also, made a Makefile change to ensure that
the expression parser C source files get regenerated correctly,
when we need that to happen.
2010-01-13 16:31 +0000 [r239712] David Vossel <dvossel@digium.com>
* Makefile, main/channel.c, apps/app_waitforring.c,
apps/app_waitforsilence.c: add silence gen to wait apps
asterisk.conf's 'transmit_silence' option existed before this
patch, but was limited to only generating silence while recording
and sending DTMF. Now enabling the transmit_silence option
generates silence during wait times as well. To achieve this,
ast_safe_sleep has been modified to generate silence anytime no
other generators are present and transmit_silence is enabled.
Wait apps not using ast_safe_sleep now generate silence when
transmit_silence is enabled as well. (closes issue #16524)
Reported by: kobaz (closes issue #16523) Reported by: kobaz
Tested by: dvossel Review:
https://reviewboard.asterisk.org/r/456/
2010-01-13 10:45 +0000 [r239663-239665] Olle Johansson <oej@edvina.net>
* main/poll.c: MAX() moved to utils.h
* channels/chan_sip.c: SIP Show channelstats fix - use float
division to show proper stats (closes issue #15819) Reported by:
klaus3000 Patches: asterisk-sip-show-channelstats-trunk.txt
uploaded by klaus3000 (license 65) Tested by: klaus3000, oej This
patch is for trunk only and will be blocked in 1.6.2
2010-01-13 07:02 +0000 [r239624-239625] TransNexus OSP Development <support@transnexus.com>
* doc/tex/channelvariables.tex: Updated channel variable list of
osplookup application.
* apps/app_osplookup.c: Updated XML doc for OSP.
2010-01-12 19:58 +0000 [r239571] Tilghman Lesher <tlesher@digium.com>
* main/pbx.c: Blank callerid and NULL callerid should not compare
equal. The second is the default state for matching CID in the
dialplan (no matching) while the first matches one particular
CallerID. This is a regression. (fixes AST-314, SWP-611)
2010-01-12 18:55 +0000 [r239525] Alec L Davis <sivad.a@paradise.net.nz>
* main/cdr.c: add Dialed Number Identifier (DNID) field to cdr
records. reviewboard link:
https://reviewboard.asterisk.org/r/455/ Reported by: alecdavis
Tested by: alecdavis Patch cdr_dnid.diff2.txt uploaded by
alecdavis (license 585)
2010-01-12 18:22 +0000 [r239520] Leif Madsen <lmadsen@digium.com>
* configs/sip.conf.sample: Note that direct T.38 is not supported.
(closes issue #16411) Reported by: stanusr Patches:
__20091210-sip.conf.sample-documentation.txt uploaded by lmadsen
(license 10)
2010-01-12 17:09 +0000 [r239473] Sean Bright <sean@malleable.com>
* res/res_config_ldap.c: Fix crash in res_config_ldap. We need to
allocate enough room for 2 pointers, not 2 characters. (closes
issue #16397) Reported by: bklang Patches: res_config_ldap.patch
uploaded by applsplatz (license 949) Tested by: applsplatz
2010-01-12 16:14 +0000 [r239427] David Vossel <dvossel@digium.com>
* channels/chan_sip.c: fixes text support in sdp answer The code
that handled setting 'm=text' in the sdp was not executing in the
correct order. The check to see if text was needed came after the
check to add 'm=text' to the sdp, this resulted in 'm=text'
always being set to 0 because it looked like text was never
required. (closes issue #16457) Reported by: peterj Patches:
textportinsdp.diff uploaded by peterj (license 951)
issue16457.diff uploaded by dvossel (license 671) Tested by:
peterj
2010-01-12 07:48 +0000 [r239389] Olle Johansson <oej@edvina.net>
* include/asterisk/astmm.h: Adding Tilghman's documentation from
asterisk-dev to the actual file.
2010-01-12 03:21 +0000 [r239152-239308] Tilghman Lesher <tlesher@digium.com>
* /, contrib/scripts/safe_asterisk: Merged revisions 239307 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r239307 | tilghman | 2010-01-11 21:18:36 -0600 (Mon, 11 Jan 2010)
| 8 lines Portability and other fixes for the safe_asterisk
script (closes issue #16416) Reported by: bklang Patches:
safe_asterisk-compat-1.patch uploaded by bklang (license 919)
20100106__issue16416__trunk.diff.txt uploaded by tilghman
(license 14) Tested by: bklang ........
* contrib/init.d/rc.mandriva.asterisk,
contrib/init.d/rc.debian.asterisk,
contrib/init.d/rc.redhat.asterisk,
contrib/init.d/rc.gentoo.asterisk,
contrib/init.d/rc.slackware.asterisk,
contrib/init.d/rc.archlinux.asterisk,
contrib/init.d/rc.suse.asterisk: Add LSB headers to init scripts.
(closes issue #14864) Reported by: lathama Patches:
lsb-init-info-debian.diff uploaded by pkempgen (license 169)
* res/res_pktccops.c: Socket level option is SOL_SOCKET, not
SO_SOCKET. (issue #16580)
* Makefile, contrib/init.d/rc.mandriva.asterisk,
contrib/init.d/rc.debian.asterisk,
contrib/init.d/rc.redhat.asterisk,
contrib/init.d/rc.suse.asterisk: Permit more options in the
Makefile as to startup options (closes issue #16454) Reported by:
syspert Patches: 20091228__issue16454__3.diff.txt uploaded by
tilghman (license 14) Tested by: syspert
* Makefile: Including bundle1.o breaks Tiger and Leopard (issue
#16449)
* addons/cdr_mysql.c, configs/cdr_mysql.conf.sample: Permit dates
and times to be stored in timezones other than the default
(typically, UTC) (closes issue #16401) Reported by: lordmortis
2010-01-11 16:41 +0000 [r239111-239114] Sean Bright <sean@malleable.com>
* res/res_calendar_exchange.c, res/res_calendar_icalendar.c,
res/res_calendar_caldav.c, res/res_clialiases.c: Pass NULL for
the ao2_callback function pointer instead of duplicating cb_true.
* main/astobj2.c: Fix ao2_callback when both OBJ_MULTIPLE and
OBJ_NODATA are passed. There is an issue which only affects trunk
and the new ao2_callback OBJ_MULTIPLE implementation. When both
OBJ_MULTIPLE and OBJ_NODATA are passed, only the first object is
visited, regardless of what is returned by the specified
callback. This causes a problem when we are clearing a container,
i.e.: ao2_callback(container, OBJ_UNLINK | OBJ_NODATA |
OBJ_MULTIPLE, NULL, NULL); Only unlinks the first object. This
patch resolves this. (closes issue #16564) Reported by: pj
Patches: issue16564_20100111.diff uploaded by seanbright (license
71) Tested by: pj, seanbright Review:
https://reviewboard.asterisk.org/r/457/
* main/test.c: Fix spelling of 'category.'
2010-01-10 19:37 +0000 [r239074] Tilghman Lesher <tlesher@digium.com>
* addons/chan_ooh323.c, main/frame.c, channels/chan_iax2.c:
According to POSIX, the capital L modifier applies only to
floating point types. Fixes a crash on Solaris. (closes issue
#16572) Reported by: crjw Patches: frame_changes.patch uploaded
by crjw (license 963) Plus several others found and fixed by me
2010-01-10 17:53 +0000 [r239037] Alexandr Anikin <may@telecom-service.ru>
* addons/ooh323c/src/ooq931.h, addons/ooh323c/src/oochannels.c,
addons/ooh323c/src/ooq931.c: add docallbacks flag in q931decode
function because when we decode received q931 packet we must do
callbacks and when we print sended q931 packet we must not.
2010-01-10 06:56 +0000 [r239000] Tilghman Lesher <tlesher@digium.com>
* Makefile, main/asterisk.c: It's been long enough -- make the
behavior introduced in 1.6 the default.
2010-01-09 01:08 +0000 [r238916] Tilghman Lesher <tlesher@digium.com>
* main/manager.c, /: Merged revisions 238915 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r238915 | tilghman | 2010-01-08 18:57:58 -0600 (Fri, 08 Jan 2010)
| 6 lines -1 is interpreted as an error, intead of the maximum
mask. (closes issue #16241) Reported by: vnovy Patches:
manager.c.patch uploaded by vnovy (license 922) ........
2010-01-08 23:30 +0000 [r238835] Jeff Peeler <jpeeler@digium.com>
* /, main/features.c: Merged revisions 238834 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r238834 | jpeeler | 2010-01-08 17:28:37 -0600 (Fri, 08 Jan 2010)
| 4 lines Stop a crash when no peer is passed to masq_park_call.
(distantly related to issue #16406) ........
2010-01-08 22:54 +0000 [r238754-238795] Tilghman Lesher <tlesher@digium.com>
* res/res_musiconhold.c: Add the class actually used in the
MusicOnHold start event. (closes issue #16499) Reported by:
syspert Patches: mohclass.patch uploaded by syspert (license 938)
* res/res_agi.c: Initialize variables that we attempt to free
later. (closes issue #16302) Reported by: yahsyn Patches:
20091124__issue16302.diff.txt uploaded by tilghman (license 14)
Tested by: yahsyn
2010-01-08 21:04 +0000 [r238716] Matthew Nicholson <mnicholson@digium.com>
* tests/test_ast_format_str_reduce.c (added): Added a test for
ast_format_reduce_str(). (related to issue #16560)
2010-01-08 19:39 +0000 [r238635] David Vossel <dvossel@digium.com>
* include/asterisk/audiohook.h, main/audiohook.c: fixes
AUDIOHOOK_INHERIT regression During the process of removing an
audiohook from one channel and attaching it to another the
audiohook's status is updated to DONE and then back to whatever
it was previously. Typically updating the status after setting it
to DONE is not a good idea because DONE can trigger unrecoverable
audiohook destruction events... because of this a conditional
check was added to audiohook_update_status to explicitly prevent
the audiohook from ever changing after being set to DONE. It was
this check that prevented audiohook inherit from work properly
though. Now ast_audiohook_move_by_source is treated as a special
exception, as the audiohook must be returned to its previous
status after attaching it to the new channel. This is only a safe
operation because the audiohook's lock is held the entire time,
otherwise this could cause trouble. (closes issue #16522)
Reported by: corruptor
2010-01-08 19:32 +0000 [r238630] Matthew Nicholson <mnicholson@digium.com>
* /, main/file.c: Merged revisions 238629 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r238629 | mnicholson | 2010-01-08 13:20:44 -0600 (Fri, 08 Jan
2010) | 5 lines Properly calculate the remaining space in the
output string when reducing format strings. (closes issue #16560)
Reported by: goldwein ........
2010-01-08 17:18 +0000 [r238583] Jeff Peeler <jpeeler@digium.com>
* main/features.c: Stop trying to find a parking space after
traversing the parkinglot one time. (closes issue #16428)
Reported by: Yasuhiro Konishi
2010-01-07 21:24 +0000 [r238527] Richard Mudgett <rmudgett@digium.com>
* channels/sig_pri.c: Fix using the wrong pointer type in
do_idle_thread().
2010-01-07 20:42 +0000 [r238361-238492] David Vossel <dvossel@digium.com>
* main/channel.c: fixes ast_transfer stall until hangup if called
with a channel that doesn't support transfers ast_transfer sets
res to 0 if there is no technology transfer function, but then
tests for it to be negative before deciding to do an early exit.
As a result, it will will wait for an AST_CONTROL_TRANSFER
message that will never come. (closes issue #16424) Reported by:
davidw Patches: Issue_16424_trunk_234134.patch uploaded by davidw
(license 780)
* /, channels/chan_iax2.c: Merged revisions 238411 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r238411 | dvossel | 2010-01-07 14:14:25 -0600 (Thu, 07
Jan 2010) | 10 lines fixes crash in "scheduled_destroy" in
chan_iax A signed short was used to represent a callnumber. This
is makes it possible to attempt to access the iaxs array with a
negative index. (closes issue #16565) Reported by: jensvb
........
* channels/chan_sip.c: Change in sip show channels display format
allowing more digits for CID (closes issue #16459) Reported by:
Rzadzins Patches: chan_sip_longer_cid.patch uploaded by Rzadzins
(license 953)
* apps/app_queue.c: cli 'queue show' formatting fix. queue name was
truncated over 12 characters (closes issue #16078) Reported by:
RoadKill Patches: quequename_limit.patch uploaded by ppyy
(license 906) Tested by: dvossel
2010-01-07 09:14 +0000 [r238313] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
* configs/sip.conf.sample: Document the usefulness of explicit
udp:// in the register string
2010-01-06 21:45 +0000 [r238231] Tilghman Lesher <tlesher@digium.com>
* /, funcs/func_cdr.c: Merged revisions 238230 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r238230 | tilghman | 2010-01-06 15:41:55 -0600 (Wed, 06 Jan 2010)
| 4 lines Revise documentation on disposition values to the
actual values used. (closes issue #16289) Reported by: wdoekes
........
2010-01-06 20:37 +0000 [r238134-238181] Jeff Peeler <jpeeler@digium.com>
* apps/app_meetme.c: Fix misreverting from 177158. (closes issue
#15725) Reported by: shanermn Patches: v1-15725.patch uploaded by
dimas (license 88) Tested by: shanermn
* main/features.c: Fix channel name comparison for bridge
application. The channel name comparison was not comparing the
whole string and therefore if one channel name was a substring of
the other, the bridge would fail. (closes issue #16528) Reported
by: telecos82 Patches: res_features_r236843.diff uploaded by
telecos82 (license 687)
2010-01-06 16:36 +0000 [r238091] David Vossel <dvossel@digium.com>
* include/asterisk/test.h: fixes test.c compile issue when
TEST_FRAMEWORK is not enabled The ast_test_status_update()
function is defined in test.h. When TEST_FRAMEWORK is not enabled
a macro is defined as a no-op place holder for this function. The
macro did not contain the correct number of arguments. This
caused a compile error. Much thanks to wdoekes for reporting the
issue and supplying the patch!
2010-01-06 15:35 +0000 [r238014] Sean Bright <sean@malleable.com>
* addons/format_mp3.c: Fix reading samples from format_mp3 after
ast_seekstream/ast_tellstream. There is a bug when using
ast_seekstream/ast_tellstream with format_mp3 in that the file
read position is not reset before attempting to read samples. So
when we seek to determine the maximum size of the file (as in
res_agi's STREAM FILE) we weren't then resetting the file pointer
so that we could properly read samples. This patch addresses that
(in a similar manner to format_wav.c). (closes issue #15224)
Reported by: rbd Patches: 20091230_addons_1.4_issue15224.diff
uploaded by seanbright (license 71) Tested by: rbd, seanbright
Review: https://reviewboard.asterisk.org/r/453
2010-01-06 15:19 +0000 [r238010] Russell Bryant <russell@digium.com>
* /, apps/app_mp3.c: Merged revisions 238009 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r238009 | russell | 2010-01-06 09:18:22 -0600 (Wed, 06 Jan 2010)
| 7 lines Resolve a crash due to an ast_frame not being fully
initialized. (closes issue #16531) Reported by: john8675309
(closes SWP-615) ........
2010-01-06 06:53 +0000 [r237968] Tilghman Lesher <tlesher@digium.com>
* channels/chan_sip.c: Whoa, duplicate setting (dead code).
2010-01-05 23:08 +0000 [r237920] David Vossel <dvossel@digium.com>
* apps/app_queue.c: fixes holdtime playback issue in app_queue When
reporting hold time, the number of seconds should be mod 60.
Otherwise audio playback could be something like "2 minutes 123
seconds" rather than "2 minutes 3 seconds". Also, the "minute"
sound file is missing, so for the moment until that file can be
created the "minutes" file is used instead. (closes issue #16168)
Reported by: nickilo Patches: patch-unified-trunk-rev-222176
uploaded by nickilo (license ) Tested by: nickilo, wonderg
2010-01-05 20:56 +0000 [r237882] Mark Michelson <mmichelson@digium.com>
* apps/app_dial.c: Mismerged a bit.
2010-01-05 19:29 +0000 [r237839] David Vossel <dvossel@digium.com>
* main/pbx.c: fixes subscriptions being lost after 'module reload'
During a module reload if multiple extension configs are present,
such as both extensions.conf and extensions.ael, watchers for one
config's hints will be lost during the merging of the other
config. This happens because hint watchers are only preserved for
the current config being merged. The old context list is
destroyed after the merging takes place, meaning any watchers
that were not perserved will be removed. Now all hints are
preserved during merging regardless of what config file is being
merged. These hints are only restored if they are present within
the new context list. (closes issue #16093) Reported by: jlaroff
2010-01-05 18:57 +0000 [r237804] Richard Mudgett <rmudgett@digium.com>
* channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_analog.c,
channels/sig_analog.h, channels/sig_pri.c: Removed unused
parameters from analog_available() and sig_pri_available().
2010-01-05 18:46 +0000 [r237802-237803] Mark Michelson <mmichelson@digium.com>
* apps/app_dial.c, CHANGES: Add a missing part of the connected
line work into trunk. Part of the work done for connected line
was to add an optional argument to the 'f' option to allow for
the connected party information of the outgoing channel to be set
to the argument provided. This was overlooked during the merge of
the work to trunk and is being added back now. The CHANGES file
has also been updated to note this change.
* CHANGES: Spell "aficionado" like someone who isn't stupid.
2010-01-05 17:26 +0000 [r237699-237749] Russell Bryant <russell@digium.com>
* main/utils.c: Fix build of utility apps that include utils.c.
* /, main/utils.c: Merged revisions 237697 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r237697 | russell | 2010-01-05 11:13:28 -0600 (Tue, 05 Jan 2010)
| 7 lines Change a NOTICE log message to DEBUG where it belongs.
(closes issue #16479) Reported by: alexrecarey (closes SWP-577)
........
2010-01-05 16:08 +0000 [r237656] Michiel van Baak <michiel@vanbaak.info>
* apps/app_mixmonitor.c: Make CLI command 'mixmonitor start|stop
<channel> work again. (closes issue #16534) Reported by:
jlaguilar Fix as suggested by jlaguilar in the bugreport
2010-01-04 21:48 +0000 [r237406-237574] Tilghman Lesher <tlesher@digium.com>
* /, main/say.c: Merged revisions 237573 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r237573 | tilghman | 2010-01-04 15:45:46 -0600 (Mon, 04 Jan 2010)
| 6 lines Bounds checking for input string (closes issue #16407)
Reported by: qwell Patches: 20100104__issue16407.diff.txt
uploaded by tilghman (license 14) ........
* main/pbx.c, /: Merged revisions 237493 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r237493 | tilghman | 2010-01-04 14:57:35 -0600 (Mon, 04 Jan 2010)
| 8 lines Regression in issue #15421 - Pattern matching (closes
issue #16482) Reported by: wdoekes Patches:
astsvn-16482-betterfix.diff uploaded by wdoekes (license 717)
20091223__issue16482.diff.txt uploaded by tilghman (license 14)
Tested by: wdoekes, tilghman ........
* main/config.c: Oops, didn't compile (thanks, kpfleming)
* main/config.c: Further reduce the encoded blank values back to
blank in the realtime API. (closes issue #16533) Reported by:
sergee Patches: 200100104__issue16533.diff.txt uploaded by
tilghman (license 14) Tested by: sergee
* main/pbx.c, /, res/res_agi.c, include/asterisk/channel.h: Merged
revisions 237405 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r237405 | tilghman | 2010-01-04 12:19:00 -0600 (Mon, 04 Jan 2010)
| 16 lines Add a flag to disable the Background behavior, for AGI
users. This is in a section of code that relates to two other
issues, namely issue #14011 and issue #14940), one of which was
the behavior of Background when called with a context argument
that matched the current context. This fix broke FreePBX,
however, in a post-Dial situation. Needless to say, this is an
extremely difficult collision of several different issues. While
the use of an exception flag is ugly, fixing all of the issues
linked is rather difficult (although if someone would like to
propose a better solution, we're happy to entertain that
suggestion). (closes issue #16434) Reported by: rickead2000
Patches: 20091217__issue16434.diff.txt uploaded by tilghman
(license 14) 20091222__issue16434__1.6.1.diff.txt uploaded by
tilghman (license 14) Tested by: rickead2000 ........
2010-01-04 16:39 +0000 [r237327] David Vossel <dvossel@digium.com>
* apps/app_queue.c: app_queue segfaults if realtime field uniqueid
is NULL (closes issue #16385) Reported by: haakon Patches:
app_queue.c.patch uploaded by haakon (license 880)
app_queue.c.patch_v2 uploaded by dvossel (license 671) Tested by:
haakon
2010-01-04 16:24 +0000 [r237323] Jeff Peeler <jpeeler@digium.com>
* res/res_agi.c: Fix timeout for AGI command speech recognize.
(closes issue #16297) Reported by: semond
2010-01-04 16:20 +0000 [r237319] Tilghman Lesher <tlesher@digium.com>
* channels/chan_local.c, /: Merged revisions 237318 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r237318 | tilghman | 2010-01-04 10:18:59 -0600 (Mon, 04
Jan 2010) | 3 lines It's also possible for the Local channel to
directly execute an Application. Reviewboard:
https://reviewboard.asterisk.org/r/452/ ........
2010-01-04 07:55 +0000 [r237284] Olle Johansson <oej@edvina.net>
* res/res_pktccops.c, channels/chan_mgcp.c: - Disable res_pktccops
by default - Add dependency in chan_mgcp that was missing - Add a
small amount of doc to the source code
2010-01-04 03:38 +0000 [r237250] TransNexus OSP Development <support@transnexus.com>
* apps/app_osplookup.c: 1. Added reporting operator names in
AuthReq. 2. Added retrieving operator names from AuthRsp and
exporting them.
2010-01-02 16:35 +0000 [r237213] Tilghman Lesher <tlesher@digium.com>
* channels/chan_sip.c: global_contact_ha was renamed in trunk
2010-01-02 09:54 +0000 [r237136] Olle Johansson <oej@edvina.net>
* /, channels/chan_sip.c: Merged revisions 237135 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r237135 | oej | 2010-01-02 10:52:30 +0100 (Lör, 02 Jan 2010) | 2
lines Release memory of the contact acl before unloading module
........
2009-12-30 23:51 +0000 [r237098] Alexandr Anikin <may@telecom-service.ru>
* addons/ooh323c/src/ooh323.c, addons/ooh323c/src/ooq931.c,
addons/ooh323c/src/ooCalls.c: small q931 processing and
signalling corrections don't decode UUIE from Q931StatusMessage
clean call without callIdentifier data don't start tcs/msd
exchange procedure after call proceeding received (closes issue
#16365) Reported by: benngard2 Tested by: may213, benngard2
2009-12-30 22:30 +0000 [r237050] Jason Parker <jparker@digium.com>
* main/say.c, doc/lang/vietnamese.ods (added),
apps/app_voicemail.c: Add app_voicemail and say.c support for
Vietnamese. Also add an XXX comment that I'm baffled nobody has
ever complained about. We say "first message", and then we go
into language-specific stuff where we proceed to say..."first
message". (closes issue #15053) Reported by: dinhtrung Patches:
vietnamese.ods uploaded by dinhtrung (license 776)
app_voicemail.c.diff uploaded by dinhtrung (license 776) (closes
issue #15626) Reported by: dinhtrung Patches: say.c.diff uploaded
by dinhtrung (license 776)
2009-12-30 21:59 +0000 [r236982] Tilghman Lesher <tlesher@digium.com>
* channels/chan_local.c, /: Merged revisions 236981 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r236981 | tilghman | 2009-12-30 15:57:10 -0600 (Wed, 30
Dec 2009) | 9 lines Don't queue frames to channels that have no
means to process them. (closes issue #15609) Reported by: aragon
Patches: 20091230__issue16521__1.4__chan_local_only.diff.txt
uploaded by tilghman (license 14) Tested by: aragon Review:
https://reviewboard.asterisk.org/r/452/ ........
2009-12-30 21:09 +0000 [r236893-236902] Jeff Peeler <jpeeler@digium.com>
* utils/ael_main.c: One more LOW_MEMORY compile fix.
* channels/chan_sip.c, main/cli.c: Fix compiling with LOW_MEMORY.
Modified handle_verbose to be LOW_MEMORY aware, removed old RTP
related code in chan_sip. (closes issue #16381) Reported by:
michael_iedema Patches: ast_complete_source_filename.patch
uploaded by michael iedema (license 942) modified by me
2009-12-30 17:53 +0000 [r236802-236847] Tilghman Lesher <tlesher@digium.com>
* cdr/cdr_adaptive_odbc.c, cel/cel_adaptive_odbc.c: When the field
is blank, don't warn about the field being unable to be coerced,
just skip the column. (closes
http://lists.digium.com/pipermail/asterisk-dev/2009-December/041362.html)
Reported by Nic Colledge on the -dev list, fixed by me.
* channels/chan_sip.c: Shut down the SIP session timers more
gracefully, in order to prevent a possible crash. (closes issue
#16452) Reported by: corruptor Patches:
20091221__issue16452.diff.txt uploaded by tilghman (license 14)
Tested by: corruptor
2009-12-29 10:59 +0000 [r236756] TransNexus OSP Development <support@transnexus.com>
* configs/osp.conf.sample, apps/app_osplookup.c, configure.ac: 1.
Updated for OSP Toolkit 3.6.0. 2. Added service type ported
number query. 3. Formated code.
2009-12-28 22:09 +0000 [r236713] Jason Parker <jparker@digium.com>
* main/ast_expr2.y, main/ast_expr2.c: Allow "REMAINDER" to function
properly in expressions. (closes issue #16427) Reported by:
wdoekes Patches: ast16-reminder-remainder.patch uploaded by
wdoekes (license 717) Tested by: wdoekes
2009-12-28 17:37 +0000 [r236667] Tilghman Lesher <tlesher@digium.com>
* apps/app_voicemail.c: Use recommended option, not deprecated
option. (closes issue #16515) Reported by: ManChicken
2009-12-28 15:22 +0000 [r236510-236613] Sean Bright <sean@malleable.com>
* /, configure, include/asterisk/autoconfig.h.in, configure.ac,
include/asterisk/threadstorage.h: Merged revisions 236585 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r236585 | seanbright | 2009-12-28 10:12:08 -0500 (Mon, 28 Dec
2009) | 7 lines Try a test compile to see if PTHREAD_ONCE_INIT
requires extra braces. There was conditional code (based on build
platform) to optioinally wrap PTHREAD_ONCE_INIT in braces that
was removed since it is fixed in newer versions of
Solaris/OpenSolaris, but I am still running into it on Solaris 10
x86 so add a configure-time check for it. ........
* /, apps/app_meetme.c: Merged revisions 236509 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r236509 | seanbright | 2009-12-28 07:43:36 -0500 (Mon, 28 Dec
2009) | 12 lines Avoid a crash with large numbers of MeetMe
conferences. Similar to changes made to Queue(), when we have
large numbers of conferences in meetme.conf (1000s) and we use
alloca()/strdupa(), we can blow out the stack and crash, so
instead just use a single fixed buffer. (closes issue #16509)
Reported by: Kashif Raza Patches: 20091223_16509.patch uploaded
by seanbright (license 71) Tested by: seanbright ........
2009-12-27 18:20 +0000 [r236434] Tilghman Lesher <tlesher@digium.com>
* contrib/init.d/rc.debian.asterisk, /: Merged revisions 236433 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r236433 | tilghman | 2009-12-27 12:19:38 -0600 (Sun, 27 Dec 2009)
| 2 lines Turn on colors in the daemon, since there's many
requests for it on Ubuntu. ........
2009-12-26 15:27 +0000 [r236358] Kevin P. Fleming <kpfleming@digium.com>
* /, sounds/Makefile: Merged revisions 236357 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r236357 | kpfleming | 2009-12-26 09:26:17 -0600 (Sat, 26 Dec
2009) | 1 line update to latest releases with zero uid/gid
........
2009-12-23 19:17 +0000 [r236304-236312] David Vossel <dvossel@digium.com>
* CHANGES: Update CHANGES to reflect new QUEUE_MEMBER option,
"ready"
* apps/app_queue.c: QUEUE_MEMBER(..., ready) counts only ready
agents, not free agents wrapping up The QUEUE_MEMBER dialplan
function can return total members, logged-in members and "free"
members count. A member is counted as "free" immediately after
his call ends, even though its wrap-up time, if specified in
queues.conf, has not yet expired, and the queue will not actually
route a call to it. This Patch introduces a new "ready" option
that only counts free agents no longer in the wrap up time
period. (closes issue #16240) Reported by: kkm Patches:
appqueue-memberfun-readyoption-trunk.diff uploaded by kkm
(license 888) Tested by: kkm, dvossel
* CHANGES, apps/app_queue.c: update CHANGES to reflect new 'R'
app_queue option plus a minor optimization to the feature patch
(issue #16384)
* apps/app_queue.c: new parameter 'R' to the Queue application The
'R' argument stops moh and indicates ringing once the agent is
ringing. This allows the person in the queue to know their call
is potentially about to be answered. (closes issue #16384)
Reported by: haakon Patches: new_app_queue.c.patch uploaded by
haakon (license 880) Tested by: haakon, loloski, dvossel
2009-12-23 18:25 +0000 [r236183-236300] Tilghman Lesher <tlesher@digium.com>
* apps/app_stack.c: AGI may be invoked from outside the dialplan
(closes issue #16510) Reported by: atis Patches:
20091223__issue16510.diff.txt uploaded by tilghman (license 14)
Tested by: atis
* /, res/res_agi.c: Merged revisions 236184 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r236184 | tilghman | 2009-12-22 20:55:24 -0600 (Tue, 22 Dec 2009)
| 4 lines If EXEC only gets a single argument, don't crash when
the second is used. (closes issue #16504) Reported by: bklang
........
* include/asterisk/test.h: Allow test_heap.c to compile when
AST_DEVMODE is true, but TEST_FRAMEWORK is false
* apps/app_voicemail.c: Actually use tmp for something (brings
trunk back into sync with 1.6 branches).
2009-12-22 21:53 +0000 [r236027-236144] David Vossel <dvossel@digium.com>
* channels/chan_iax2.c: fixes iax "can't compress subclass
4294967295" error (closes issue #16456) Reported by: dvossel
Tested by: dvossel
* /, channels/chan_sip.c: Merged revisions 236062 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r236062 | dvossel | 2009-12-22 10:58:19 -0600 (Tue, 22 Dec 2009)
| 11 lines fixes issue with p->method incorrectly set to ACK It
is possible for a second ACK to come in for a retransmitted
message. If an ack does not match an unacked message in our
queue, restore the previous p->method as this ACK is completely
ignored. (closes issue #16295) Reported by: omolenkamp Patches:
issue16295_v2.diff uploaded by dvossel (license 671) ........
* CHANGES: update CHANGES to reflect the addition of the test
framework
* include/asterisk/test.h (added), build_tools/cflags-devmode.xml,
tests/test_heap.c, main/test.c (added),
include/asterisk/_private.h, main/asterisk.c: Unit Test Framework
API The Unit Test Framework is a new API that manages
registration and execution of unit tests in Asterisk with the
purpose of verifying the operation of C functions. The Framework
consists of a single test manager accompanied by a list of
registered test functions defined within the code. A test is
defined, registered, and unregistered from the framework using a
set of macros which allow the test code to only be compiled
within asterisk when the TEST_FRAMEWORK flag is enabled in
menuselect. This allows the test code to exist in the same file
as the C functions it intends to verify. Registered tests may be
viewed and executed via a set of new CLI commands. CLI commands
are also present for generating and exporting test results into
xml and txt formats. For more information and use cases please
refer to the documentation provided at the beginning of the
test.h file. Review: https://reviewboard.asterisk.org/r/447/
2009-12-21 19:54 +0000 [r235941] Jeff Peeler <jpeeler@digium.com>
* /, res/res_monitor.c: Merged revisions 235940 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r235940 | jpeeler | 2009-12-21 13:43:41 -0600 (Mon, 21 Dec 2009)
| 13 lines Change Monitor to not assume file to write to does not
contain pathing. 227944 changed the fname_base argument to always
append the configured monitor path. This change was necessary to
properly compare files for uniqueness. If a full path is given
though, nothing needs to be appended and that is handled
correctly now. (closes issue #16377) (closes issue #16376)
Reported by: bcnit Patches: res_monitor.c-issue16376-1.patch
uploaded by dant (license 670) ........
2009-12-21 18:51 +0000 [r235904] Kevin P. Fleming <kpfleming@digium.com>
* contrib/upstart/asterisk.upstart-0.3.9, include/asterisk/cel.h,
main/say.c, include/asterisk/channel.h,
include/asterisk/manager.h, channels/sig_pri.c,
include/asterisk/logger.h, include/asterisk/http.h,
include/asterisk/callerid.h, include/asterisk/syslog.h,
channels/chan_dahdi.c, include/asterisk/app.h,
include/asterisk/doxyref.h, include/asterisk/event.h,
channels/sig_analog.c, channels/chan_misdn.c,
contrib/upstart/asterisk.user.conf,
include/asterisk/rtp_engine.h,
include/asterisk/security_events.h,
include/asterisk/stringfields.h: Change all refererences to 1.6.3
to be 1.8, since that will be the next feature release
2009-12-21 17:00 +0000 [r235822] Tilghman Lesher <tlesher@digium.com>
* /, main/features.c: Merged revisions 235821 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r235821 | tilghman | 2009-12-21 10:45:03 -0600 (Mon, 21 Dec 2009)
| 8 lines Send parking lot announcement to the channel which
parked the call, not the park-ee. (closes issue #16234) Reported
by: yeshuawatso Patches: 20091210__issue16234.diff.txt uploaded
by tilghman (license 14) 20091221__issue16234__1.4.diff.txt
uploaded by tilghman (license 14) Tested by: yeshuawatso ........
2009-12-20 08:22 +0000 [r235740-235774] Alec L Davis <sivad.a@paradise.net.nz>
* main/dsp.c: restarts busydetector (if enabled) when DTMF is
received after call is bridged. (closes issue 0016389) Reported
by: alecdavis Tested by: alecdavis Patch
dtmf_busydetector.diff2.txt uploaded by alecdavis (license 585)
* apps/app_dial.c, CHANGES: app_dial optional parameter to option
'r' to allow play indication from indications.conf (closes issue
#14504) Reported by: alecdavis Tested by: alecdavis,jsmith Patch
app_dial.play_ring_indications.diff7.txt uploaded by alecdavis
(license 585)
2009-12-18 22:51 +0000 [r235660] Jeff Peeler <jpeeler@digium.com>
* main/channel.c, /, include/asterisk/cdr.h: Merged revisions
235635 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r235635 | jpeeler | 2009-12-18 16:29:51 -0600 (Fri, 18 Dec 2009)
| 48 lines Correct CDR dispositions for BUSY/FAILED This patch is
simple in that it reorders the disposition defines so that the
fix for issue 12946 works properly (the default CDR disposition
was changed to AST_CDR_NOANSWER). Also, the
AST_CDR_FLAG_ORIGINATED flag was set in ast_call to ensure all
CDR records are written. The side effects of CDR changes are
scary, so I'm documenting the test cases performed to attempt to
catch any regressions. The following tests were all performed
using 1.4 rev 195881 vs head (235571) + patch: A calls B C calls
B (busy) Hangup C Hangup A (Both SIP and features) A calls B A
blind transfers to C Hangup C (Both SIP and features) A calls B A
attended transfers to C Hangup C A calls B A attended transfers
to C (SIP) C blind transfers to A (features) Hangup A All of the
test scenario CDRs matched. The following tests were performed
just with the patch to ensure proper operation (with
unanswered=yes): exten =>s,1,Answer exten =>s,n,ResetCDR(w) exten
=>s,n,ResetCDR(w) exten =>s,1,ResetCDR(w) exten =>s,n,ResetCDR(w)
(closes issue #16180) Reported by: aatef Patches: bug16180.patch
uploaded by jpeeler (license 325) ........
2009-12-18 22:40 +0000 [r235573-235656] Tilghman Lesher <tlesher@digium.com>
* /, configure, configure.ac: Merged revisions 235652 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r235652 | tilghman | 2009-12-18 16:39:30 -0600 (Fri, 18
Dec 2009) | 2 lines Revise verbiage, per #asterisk-dev discussion
........
* /, configure, configure.ac: Merged revisions 235572 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r235572 | tilghman | 2009-12-18 15:18:16 -0600 (Fri, 18
Dec 2009) | 2 lines Point to the typical missing package, not the
cryptic "termcap support". ........
2009-12-17 23:21 +0000 [r235521] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: Remove some old code for going to the 'fax'
extension when a T.38 switchover occurs. This would have already
happened when we detected the CNG tone so this was basically a
noop.
2009-12-17 17:19 +0000 [r235422] Tilghman Lesher <tlesher@digium.com>
* main/pbx.c, /: Merged revisions 235421 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r235421 | tilghman | 2009-12-17 11:17:51 -0600 (Thu, 17 Dec 2009)
| 8 lines Use context from which Macro is executed, not macro
context, if applicable. Also, ensure that the extension COULD
match, not just that it won't match more. (closes issue #16113)
Reported by: OrNix Patches: 20091216__issue16113.diff.txt
uploaded by tilghman (license 14) Tested by: OrNix ........
2009-12-17 00:52 +0000 [r235342-235382] Jeff Peeler <jpeeler@digium.com>
* channels/chan_dahdi.c, channels/sig_analog.c: Fix call forwarding
for analog phones. (closes issue #16440) Reported by: mmichelson
* configs/jabber.conf.sample, include/asterisk/jabber.h, CHANGES,
res/res_jabber.c: Add auth_policy option to jabber.conf for auto
user registration. The option is global and currently the
acceptable values as noted in the sample config are accept or
deny. (closes issue #15228) Reported by: lp0
2009-12-16 05:24 +0000 [r235298] Jared Smith <jaredsmith@jaredsmith.net>
* /, configs/sip.conf.sample: Merged revisions 235181 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r235181 | jsmith | 2009-12-15 15:07:55 -0600 (Tue, 15
Dec 2009) | 4 lines Add a line showing that we can use CIDR
notation. patch by jsmith, after discussion with jtodd ........
2009-12-16 00:31 +0000 [r235265] Jeff Peeler <jpeeler@digium.com>
* main/manager.c, CHANGES: Enhance AMI redirect to allow channels
to be redirected to different places. New parameters
ExtraContext, ExtraExtension, and ExtraPriority have been added
to redirect the second channel to a different location.
Previously, it was only possible to redirect both channels to the
same place. (closes issue #15853) Reported by: haakon Patches:
trunk-manager.c.patch uploaded by haakon (license 880) Tested by:
jpeeler
2009-12-15 23:51 +0000 [r235229] Tilghman Lesher <tlesher@digium.com>
* include/asterisk/strings.h: Is it Friday yet?
2009-12-15 23:41 +0000 [r235226] Jeff Peeler <jpeeler@digium.com>
* main/channel.c: Change match criteria existence in
ast_channel_cmp_cb to use ast_strlen_zero. (closes issue #16161)
Reported by: may213 Patches: core-show-channel.patch uploaded by
may213 (license 454)
2009-12-15 18:43 +0000 [r235132] David Vossel <dvossel@digium.com>
* channels/chan_sip.c: reverse minor sip registration regression A
registration regression caused by a code tweak in (issue #14331)
and a bug fix in (issue #15539) caused some sip registration
config entries to be constructed incorrectly. Origially issue
#14331 contained the code tweak as well as a bug fix, but since
the issue was reported as a tweak the bug fix portion was moved
into issue #15539. Both the tweak and the bug fix contained minor
incorrect logic that resulted in some SIP registrations to fail.
(issue #14331) (issue #15539)
2009-12-15 15:33 +0000 [r235053] Tilghman Lesher <tlesher@digium.com>
* /, res/res_agi.c: Merged revisions 235052 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r235052 | tilghman | 2009-12-15 09:29:24 -0600 (Tue, 15 Dec 2009)
| 4 lines Mandatory argument checking (closes issue #16446)
Reported by: nicchap ........
2009-12-15 14:35 +0000 [r235010] Kevin P. Fleming <kpfleming@digium.com>
* apps/app_fax.c: spandsp does in fact support V.17 modulation at
14.4 kilobits per second, so we should generate T38MaxBitRate of
14400 (even though that doesn't really affect the FAX
transmission much at all)
2009-12-15 07:18 +0000 [r234855-234976] Alec L Davis <sivad.a@paradise.net.nz>
* apps/app_directory.c: Support option 'n', as applications like
Playback, Background etc. Suggested on asterisk-dev as trivial
application change. Reported by: alecdavis Tested by: alecdavis
* main/dsp.c: Whitespace.
* main/dsp.c: restarts busydetector (if enabled) when DTMF is
received. (closes issue #16389) Reported by: alecdavis Tested by:
alecdavis Patch dtmf_busydetector.diff.txt uploaded by alecdavis
(license 585)
* apps/app_directory.c: fixes escape to extensions 'o' and 'a', for
digits '0' and '*' (closes issue #16437) Reported by: alecdavis
Tested by: alecdavis Patch extension_o_a_fix.diff.txt uploaded by
alecdavis (license 585)
* apps/app_directory.c: ast_stream_and_wait(chan,dir-usingkeypad)
didn't capture the dialled DTMF. (closes issue #16409) Reported
by: alecdavis Tested by: alecdavis Patch bug_16409.diff.txt
uploaded by alecdavis (license 585)
2009-12-14 23:16 +0000 [r234820] Tilghman Lesher <tlesher@digium.com>
* configs/voicemail.conf.sample, CHANGES, apps/app_voicemail.c:
Allow greetings-only mailboxes for Voicemail. (closes issue
#15132) Reported by: floletarmo Patches: voicemail_changes.patch
uploaded by floletarmo (license 784) (with some additional
changes by me)
2009-12-14 21:32 +0000 [r234776] Jason Parker <jparker@digium.com>
* apps/app_readexten.c: Allow tonelist as argument to ReadExten.
ReadExten already supported playing a tonezone from
indications.conf. It now has the ability to use a tonelist like
440+480/2000|0/4000 (closes issue #15185) Reported by: jcovert
Patches: app_readexten.c.patch uploaded by jcovert (license 551)
Tested by: qwell Patch modified by me, to maintain backwards
compatibility.
2009-12-14 21:13 +0000 [r234700] Tilghman Lesher <tlesher@digium.com>
* /, build_tools/make_version_c, build_tools/make_version_h: Merged
revisions 234699 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r234699 | tilghman | 2009-12-14 15:09:56 -0600 (Mon, 14 Dec 2009)
| 5 lines Deal with the situation where .flavor exists but
.version does not. Also make the script slightly more portable,
in keeping with autoconf syntax. (closes issue #14737) Reported
by: davidw ........
2009-12-14 17:19 +0000 [r234631] Leif Madsen <lmadsen@digium.com>
* doc/tex/imapstorage.tex, /: Update IMAP build documentation.
Update the IMAP build documentation to show how to build on
64-bit platforms. (issue #16433) Reported by: shrift Tested by:
lmadsen
2009-12-14 16:08 +0000 [r234572] Sean Bright <sean@malleable.com>
* main/timing.c: The default rate for 'timing test' is actually
50/sec, not 100/sec as advertised.
2009-12-14 10:46 +0000 [r234526] Olle Johansson <oej@edvina.net>
* /, channels/chan_sip.c: Merged revisions 234492 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r234492 | oej | 2009-12-14 11:16:00 +0100 (Mån, 14 Dec 2009) | 8
lines Stop sending 183's after call hangup. There where still
cases where the 183 keep-alive mechanism would not stop sending
183's even though the Asterisk server had sent a final reply to
the invite. EDVX-28 ........
2009-12-13 09:41 +0000 [r234458] Tilghman Lesher <tlesher@digium.com>
* main/pbx.c: Trim leading/trailing spaces from the filename, to
deal with common user error.
2009-12-11 23:17 +0000 [r234380] Jeff Peeler <jpeeler@digium.com>
* /, apps/app_meetme.c: Merged revisions 234379 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r234379 | jpeeler | 2009-12-11 16:37:21 -0600 (Fri, 11 Dec 2009)
| 11 lines Fix talking detection status after conference user is
muted. This patch ensures that when a conference user is muted
that the accompanying AMI Meetme talking off event is sent. Also,
the meetme list output is updated to show the muted user as
unmonitored. (closes issue #16247) Reported by: dimas Patches:
v3-16247.patch uploaded by dimas (license 88) ........
2009-12-10 21:01 +0000 [r234256] Jason Parker <jparker@digium.com>
* Makefile, /: Merged revisions 234255 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r234255 | qwell | 2009-12-10 14:58:09 -0600 (Thu, 10 Dec 2009) |
9 lines Fix unselecting of menuselect options via GLOBAL_MAKEOPTS
and USER_MAKEOPTS. (closes issue #16296) Reported by: abelbeck
Patches: issue16296-20091210.diff uploaded by qwell (license 4)
(abelbeck described a fix, which I expanded upon) Tested by:
abelbeck, qwell, lmadsen ........
2009-12-10 18:56 +0000 [r234210] Tilghman Lesher <tlesher@digium.com>
* res/res_musiconhold.c: Missed a case that emits a WARNING where
none is warranted.
2009-12-10 17:31 +0000 [r234173] Jeff Peeler <jpeeler@digium.com>
* apps/app_meetme.c, apps/app_page.c, main/app.c, CHANGES: Add
audio announcement option to app_page As described in the CHANGES
file: * MeetMe has a new option 'G' to play an announcement
before joining a conference. * Page has a new option 'A(x)' which
will playback an announcement simultaneously to all paged phones
(and optionally excluding the caller's one using the new option
'n') before the call is bridged. To add the new option to meetme,
the conference flag options had to be extended to 64 bits.
(closes issue #14365) Reported by: dferrer Patches:
page_announce.patch uploaded by dferrer (license 525) modified by
me Review: https://reviewboard.asterisk.org/r/188/
2009-12-10 16:24 +0000 [r234129] Tilghman Lesher <tlesher@digium.com>
* /, channels/chan_sip.c: Merged revisions 234095 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r234095 | tilghman | 2009-12-10 10:08:20 -0600 (Thu, 10 Dec 2009)
| 9 lines When we receive no response at all to our INVITE, allow
the channel to be destroyed. (closes issue #15627) Reported by:
falves11 Patches: 20091209__issue15627__1.6.0.diff.txt uploaded
by tilghman (license 14) 20091209__issue15627__1.4.diff.txt
uploaded by tilghman (license 14) Tested by: falves11 Review:
https://reviewboard.asterisk.org/r/446/ (closes issue #15716)
Reported by: dant (closes issue #16270) Reported by: corruptor
(closes issue #15356) Reported by: falves11 (issue #16382)
Reported by: lftsy ........
2009-12-09 23:35 +0000 [r233967-234055] Russell Bryant <russell@digium.com>
* UPGRADE.txt, CHANGES: Move an entry from CHANGES to UPGRADE.txt.
* UPGRADE.txt, CHANGES: Move an entry from CHANGES that should be
in UPGRADE.txt.
* CHANGES: Provide a real description of LOCAL_PEEK().
* CHANGES: Remove a feature from CHANGES that was listed twice for
1.6.2.
* CHANGES: Fix up the faxdetect entry in CHANGES. This feature was
listed as a 1.6.2 feature, even though it's in all 1.6.X
versions. The description of the feature was also no longer
accurate.
* CHANGES: Remove an entry from CHANGES that is already in
UPGRADE.txt (where it should be).
2009-12-08 18:40 +0000 [r233718-233732] Tilghman Lesher <tlesher@digium.com>
* addons/res_config_mysql.c: Typo pointed out on #asterisk-dev (by
atis_work)
* res/res_musiconhold.c: Find another ref leak and change how we
manage module references. (closes issue #16388, closes issue
#16279, closes issue #16390) Reported by: parisioa Patches:
20091208__issue16388.diff.txt uploaded by tilghman (license 14)
Tested by: parisioa, tilghman Review:
https://reviewboard.asterisk.org/r/442/
2009-12-08 18:00 +0000 [r233692] Russell Bryant <russell@digium.com>
* formats/format_sln.c, formats/format_wav.c,
formats/format_ogg_vorbis.c, formats/format_sln16.c,
formats/format_wav_gsm.c, formats/format_siren7.c,
formats/format_ilbc.c, formats/format_vox.c,
formats/format_pcm.c, formats/format_h263.c,
formats/format_g723.c, formats/format_h264.c,
formats/format_g726.c, formats/format_siren14.c,
formats/format_jpeg.c, formats/format_gsm.c,
formats/format_g729.c: Set a module load priority for format
modules. A recent change to app_voicemail made it such that the
module now assumes that all format modules are available while
processing voicemail configuration. However, when autoloading
modules, it was possible that app_voicemail was loaded before the
format modules. Since format modules don't depend on anything,
set a module load priority on them to ensure that they get loaded
first when autoloading. This fix applies to trunk, 1.6.1, and
1.6.2. The fix for 1.4 and 1.6.0 will require a different
approach since the module load priority functionality is not
present in the module API. (issue #16412) Reported by: jiddings
2009-12-07 23:28 +0000 [r233611] David Vossel <dvossel@digium.com>
* main/utils.c: fixes incorrect logic in ast_uri_encode issue
#16299
2009-12-07 23:10 +0000 [r233577] Atis Lezdins <atis@iq-labs.net>
* contrib/valgrind.supp: Fix compatibility with valgrind 3.3 and
older. (noticed in issue #16388) Reported by: parisioa Patches:
valgrind.supp uloaded by atis (license 242) Tested by: atis,
parisioa
2009-12-07 19:48 +0000 [r233545] David Ruggles <thedavidfactor@gmail.com>
* apps/app_externalivr.c: Fix TCP Client interface Fix a couple of
very minor bugs that prevent the socket client from working. The
wrong set of properties were used in one place and the size of
the address variable isn't set if the host name is an ip address.
Also includes a fix for a bug that was introduced previously.
(closes issue #16121) Reported by: thedavidfactor Tested by:
thedavidfactor Review: https://reviewboard.asterisk.org/r/439/
2009-12-07 18:08 +0000 [r233472] David Vossel <dvossel@digium.com>
* /, channels/chan_sip.c: Merged revisions 233471 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r233471 | dvossel | 2009-12-07 12:07:38 -0600 (Mon, 07 Dec 2009)
| 9 lines fixes missing Contact header angle brackets (closes
issue #16298) Reported by: mgernoth Patches:
reg_parse_issue_1.4.diff uploaded by dvossel (license 671) Tested
by: dvossel ........
2009-12-07 17:59 +0000 [r233468] Jeff Peeler <jpeeler@digium.com>
* include/asterisk/jabber.h, CHANGES, res/res_jabber.c: Add
applications JabberJoin, JabberLeave, JabberSendGroup for XMPP
groupchat (closes issue #14352) Reported by: fiddur Patches:
trunk-14352-2.diff uploaded by phsultan (license 73) Tested by:
fiddur
2009-12-07 16:14 +0000 [r233394] Matthew Nicholson <mnicholson@digium.com>
* channels/chan_sip.c: Do not reject SDP packets describing only
non audio streams. (closes issue #16387) Reported by: zalex1953
Patches: media-level-c-fix1.diff uploaded by mnicholson (license
96) Tested by: mnicholson, zalex1953
2009-12-06 07:01 +0000 [r233358] Tilghman Lesher <tlesher@digium.com>
* include/asterisk/compat.h, main/strcompat.c, main/app.c: Move
implementation of closefrom(3) from app.c to strcompat.c
2009-12-04 21:54 +0000 [r233280] David Vossel <dvossel@digium.com>
* configs/iax.conf.sample, /: Merged revisions 233279 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r233279 | dvossel | 2009-12-04 15:54:01 -0600 (Fri, 04
Dec 2009) | 7 lines clarify requirecalltoken option in
iax.sample.conf (closes issue #16223) Reported by: bklang
Patches: clarify-iax-requirecalltoken.patch uploaded by bklang
(license 919) ........
2009-12-04 21:06 +0000 [r233239] Tilghman Lesher <tlesher@digium.com>
* main/translate.c: Using the builtin function breaks OpenBSD 4.2
(closes issue #16395) Reported by: jtodd
2009-12-04 20:21 +0000 [r233121-233235] David Vossel <dvossel@digium.com>
* CHANGES: update CHANGES file for .m3u support in Mp3Player
application
* apps/app_mp3.c: .m3u support for Mp3Player app (closes issue
#14823) Reported by: macli Patches: app_mp3.diff1 uploaded by
macli (license ) Tested by: macli, dvossel
* CHANGES: update CHANGES for new queue option,
penaltymemberslimit.
* apps/app_queue.c: changes penaltymemberslimit to use scanf for
config value parsing
* configs/queues.conf.sample, apps/app_queue.c: new queue option,
penaltymemberslimit, disregards penalty on too few queue members
when enabled (closes issue #14559) Reported by: fiddur Patches:
trunk-199584-1.diff uploaded by fiddur (license 678) Tested by:
fiddur, dvossel
* /, apps/app_voicemail.c: Merged revisions 233116 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r233116 | dvossel | 2009-12-04 11:21:34 -0600 (Fri, 04
Dec 2009) | 6 lines document and rename strip_control() in
app_voicemail (closes issue #16291) Reported by: wdoekes ........
2009-12-04 17:18 +0000 [r233100] Russell Bryant <russell@digium.com>
* main/channel.c, /: Merged revisions 233092 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r233092 | russell | 2009-12-04 11:12:47 -0600 (Fri, 04 Dec 2009)
| 7 lines Only do frame payload check for HOLD frames. This code
was added for helping to debug the source of invalid HOLD frames.
However, a side effect of this is that it will incorrectly report
errors for frames that have an integer payload. Make the check
for this block specific to the HOLD frame case. ........
2009-12-04 17:15 +0000 [r233093] Matthias Nick <mnick@digium.com>
* pbx/pbx_config.c: Parse global variables or expressions in hint
extensions Parse global variables or expressions in hint
extensions. Like: exten => 400,hint,DAHDI/i2/${GLOBAL(var)}
(closes issue #16166) Reported by: rmudgett Tested by: mnick,
rmudgett
2009-12-04 16:55 +0000 [r233059-233089] Michiel van Baak <michiel@vanbaak.info>
* channels/chan_skinny.c: Let's unlock the lines list after the
AST_LIST_TRAVERSE instead of inside it.
* channels/chan_skinny.c: Only assign line and device in
handle_transfer_button when we have a subchannel. (closes issue
#16040) Reported by: ebroad
2009-12-04 16:08 +0000 [r233050] Tilghman Lesher <tlesher@digium.com>
* addons/res_config_mysql.c: Update the mysql driver to always
return NULL columns, as this is needed for the realtime API to
work correctly. (closes issue #16138) Reported by: sohosys
Patches: 20091029__issue16138.diff.txt uploaded by tilghman
(license 14) Tested by: sohosys
2009-12-04 15:38 +0000 [r233046] Matthias Nick <mnick@digium.com>
* /, main/dsp.c: Merged revisions 233014 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r233014 | mnick | 2009-12-04 09:17:03 -0600 (Fri, 04 Dec 2009) |
11 lines Warning message gets displayed only once Added
additional field 'int display_inband_dtmf_warning', which when
set to '1' displays the warning ('Inband DTMF is not supported on
codec %s. Use RFC2833'), and when set to '0' doesn't display the
warning. Otherwise you would get hundreds of warnings every
second. (closes issue #15769) Reported by: falves11 Patches:
patch_15769_14.txt uploaded by mnick (license 874) Tested by:
mnick, falves11 ........
2009-12-04 05:26 +0000 [r232854-232982] Tilghman Lesher <tlesher@digium.com>
* res/res_pktccops.c: Buildbot complained
* configure, include/asterisk/autoconfig.h.in, configure.ac,
res/res_pktccops.c: OS X does not define MSG_NOSIGNAL, but it
does have a socket option SO_NOSIGPIPE. (closes issue #16178)
Reported by: oej
* configs/voicemail.conf.sample, CHANGES, apps/app_voicemail.c: Add
pagerdateformat, to allow shorter dates for SMS messages. (closes
issue #16263) Reported by: andrew Patches: pagerdate.patch
uploaded by andrew (license 240) (with a slight modification by
me)
* /, apps/app_voicemail.c: Merged revisions 232820 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r232820 | tilghman | 2009-12-03 14:10:19 -0600 (Thu, 03
Dec 2009) | 8 lines Deprecate "cz" in favor of "cs". Also, change
the use of language codes so that language registers as a prefix,
rather than an exact match. (closes issue #16272) Reported by:
patrol-cz Patches: 20091203__issue16272.diff.txt uploaded by
tilghman (license 14) ........
2009-12-03 20:26 +0000 [r232853] Alexandr Anikin <may@telecom-service.ru>
* addons/ooh323c/src/ooh323.c, addons/chan_ooh323.c,
addons/ooh323c/src/ooh245.c: jitterbuffer setup correction
correction of double pointer references from previous rev
2009-12-03 08:47 +0000 [r232738-232771] TransNexus OSP Development <support@transnexus.com>
* apps/app_osplookup.c: Replaced two deprecated functions of OSP
Toolkit.
* apps/app_osplookup.c: Added custom info support.
2009-12-03 00:38 +0000 [r232700] Jeff Peeler <jpeeler@digium.com>
* configs/voicemail.conf.sample, CHANGES, apps/app_voicemail.c:
Extend voicemail to allow IMAP folders to be specified per
mailbox. Previously only possible per context, new option called
imapfolder. (closes issue #14298) Reported by: jablko Patches:
patch-200906202 uploaded by jablko (license 675)
2009-12-03 00:09 +0000 [r232660-232661] Tilghman Lesher <tlesher@digium.com>
* res/res_musiconhold.c: Remove debugging line
* include/asterisk/astobj2.h, res/res_musiconhold.c: Fix multiple
issues with musiconhold, which led to classes not getting
destroyed properly. * Classes are now tracked past removal from
the core container, and module removal is actively prevented
until all references are freed. * A hanging reference stored in
the channel has been removed. This could have caused a mismatch
and the music state not properly cleared, if two or more reloads
occurred between MOH being stopped and MOH being restarted. * In
certain circumstances, duplicate classes were possible. * A race
existed at reload time between a process being killed and the
thread responsible for reading from the related pipe respawning
that process. * Several reference counts have also been
corrected. At least one could have caused deleted classes to
stick around forever, consuming resources. This originally
manifested as MOH external processes that were not killed at
reload time. (closes issue #16279, closes issue #16207) Reported
by: parisioa, dcabot Patches: 20091202__issue16279__2.diff.txt
uploaded by tilghman (license 14) Tested by: parisioa, tilghman
2009-12-02 23:27 +0000 [r232657] David Vossel <dvossel@digium.com>
* UPGRADE.txt, CHANGES: update CHANGES and UPGRADE.txt for early
media behavior change between 1.6.1 and 1.6.2 (closes issue
#16212) Reported by: miki
2009-12-02 22:17 +0000 [r232587] David Ruggles <thedavidfactor@gmail.com>
* apps/app_externalivr.c: Prevent double closing of FDs by EIVR
This caused a problem when asterisk was under heavy load and
running both AGI and EIVR applications. EIVR would close an FD at
which point it would be considered freed and be used by a new AGI
instance the second close would then close the FD now in use by
AGI. (closes issue #16305) Reported by: diLLec Tested by:
thedavidfactor, diLLec Review:
https://reviewboard.asterisk.org/r/436/
2009-12-02 22:02 +0000 [r232582] Jeff Peeler <jpeeler@digium.com>
* main/manager.c, /: Merged revisions 232581 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r232581 | jpeeler | 2009-12-02 15:57:42 -0600 (Wed, 02 Dec 2009)
| 7 lines Send ack (response/message) after receiving manager
action userevent (closes issue #16264) Reported by: dimas
Patches: event-ack.patch uploaded by dimas (license 88) ........
2009-12-02 21:37 +0000 [r232580] Matthew Nicholson <mnicholson@digium.com>
* addons/chan_mobile.c: Fix support for multiline SMS messages in
chan_mobile. (closes issue #16278) Reported by: Artem Patches:
multiline-sms-fix2.diff uploaded by mnicholson (license 96)
Tested by: Artem
2009-12-02 21:32 +0000 [r232576] Jeff Peeler <jpeeler@digium.com>
* main/manager.c: Make manager response to "Action: events" finish
with empty line (closes issue #16275) Reported by: vnovy Patches:
manager.c.diff uploaded by vnovy (license 922)
2009-12-02 21:13 +0000 [r232544] Matthew Nicholson <mnicholson@digium.com>
* addons/chan_mobile.c: Do something with the service indicator so
that asterisk does not attempt to use a chan_mobile endpoint that
does not have service. (closes issue #16132) Reported by: nikkk
Patches: service-indicator2.diff uploaded by mnicholson (license
96) Tested by: nikkk
2009-12-02 20:10 +0000 [r232442-232510] Joshua Colp <jcolp@digium.com>
* CHANGES, main/asterisk.c, doc/asterisk.sgml: Add an 'X' option to
the asterisk application which enables #exec for configuration
files. This option can be used to enable #exec support in the
asterisk.conf configuration file. (closes issue #16260) Reported
by: atis Patches: exec_includes.patch uploaded by atis (license
242)
* apps/app_record.c, CHANGES: Add an option to Record which enables
a mode where any DTMF digit will terminate recording. (closes
issue #15436) Reported by: Vince Patches: app_record.diff
uploaded by Vince (license 823) Tested by: dbrooks
2009-12-02 17:18 +0000 [r232365] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Do not change the exten string field or
rebuild the contact header on an inbound sip_pvt if the outbound
call is redirected.
2009-12-02 17:06 +0000 [r232356] Joshua Colp <jcolp@digium.com>
* /, apps/app_amd.c: Merged revisions 232355 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r232355 | file | 2009-12-02 13:04:52 -0400 (Wed, 02 Dec 2009) | 5
lines Fix a bug where if you hung up very quickly after calling
AMD it would overwrite the AMDSTATUS of HANGUP with TOOLONG.
(closes issue #16239) Reported by: CGMChris ........
2009-12-02 17:00 +0000 [r232351] David Vossel <dvossel@digium.com>
* /, main/acl.c: Merged revisions 232350 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r232350 | dvossel | 2009-12-02 10:59:18 -0600 (Wed, 02 Dec 2009)
| 6 lines ast_outaddrfor doesn't do htons() on port, looks odd in
strace. (closes issue #16290) Reported by: wdoekes ........
2009-12-02 16:40 +0000 [r232345] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: Add support for handling the 415 Unsupported
media type response like we do for a 488 Not acceptable here
response. (closes issue #16186) Reported by: atis Patches:
sip_t38_response_415.patch uploaded by atis (license 242)
2009-12-02 15:42 +0000 [r232269] David Vossel <dvossel@digium.com>
* funcs/func_groupcount.c, /: Merged revisions 232268 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r232268 | dvossel | 2009-12-02 09:41:36 -0600 (Wed, 02
Dec 2009) | 9 lines fixes segfault in func_groupcount closes
issue #16337) Reported by: Parantido Patches: issue_16337.diff
uploaded by dvossel (license 671) Tested by: Parantido, dvossel
........
2009-12-02 14:54 +0000 [r232230] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: Fix a bug where a scheduled item ID would
get retained on registrations in a certain scenario causing code
to execute during reload that should not. (issue AST-263)
2009-12-02 03:26 +0000 [r232164] Tilghman Lesher <tlesher@digium.com>
* configure, include/asterisk/autoconfig.h.in,
include/asterisk/compat.h, main/strcompat.c, configure.ac: So
apparently, some platforms don't have ffsll(3). The manpage lies;
it says that the function is in POSIX, but that's only for
ffs(3), not ffsll(3).
2009-12-02 00:45 +0000 [r232091] Jeff Peeler <jpeeler@digium.com>
* channels/chan_dahdi.c, /: Merged revisions 232090 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r232090 | jpeeler | 2009-12-01 18:42:58 -0600 (Tue, 01
Dec 2009) | 10 lines Do not modify the gain settings on data
calls. (The digital flag actually represents a data call.)
(closes issue #15972) Reported by: udosw Patches:
transcap_digital_fix.diff.txt uploaded by alecdavis (license 585)
Tested by: alecdavis ........
2009-12-01 23:56 +0000 [r232008-232017] Russell Bryant <russell@digium.com>
* main/translate.c: Use __builtin_ffsll() from gcc instead of
ffssll() to fix a FreeBSD build error.
* funcs/func_lock.c: Fix a build error on FreeBSD.
* /, main/file.c: Merged revisions 232007 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r232007 | russell | 2009-12-01 17:25:36 -0600 (Tue, 01 Dec 2009)
| 2 lines Fix a warning pointed out by buildbot. ........
2009-12-01 21:54 +0000 [r231927] Jeff Peeler <jpeeler@digium.com>
* main/channel.c, /: Merged revisions 231911 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r231911 | jpeeler | 2009-12-01 15:29:31 -0600 (Tue, 01 Dec 2009)
| 12 lines Fix crash with invalid frame data The crash was
happening as a result of a frame containing an invalid data
pointer, but was set with data length of zero. The few times the
issue was reproduced it _seemed_ that the frame was queued
properly, that is the data pointer was set to NULL. I never could
reproduce the crash so as a last resort the crash has been fixed,
but a check in __ast_read has been added to give as much
information about the source of problematic frames in the future.
(closes issue #16058) Reported by: atis ........
2009-12-01 21:20 +0000 [r231867] David Vossel <dvossel@digium.com>
* main/pbx.c, /: Merged revisions 231853 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r231853 | dvossel | 2009-12-01 15:14:31 -0600 (Tue, 01 Dec 2009)
| 3 lines WaitExten m option with no parameters generates frame
with zero datalen but non-null data ptr ........
2009-12-01 20:27 +0000 [r231814-231850] Tilghman Lesher <tlesher@digium.com>
* res/res_rtp_asterisk.c, channels/chan_unistim.c,
main/rtp_engine.c, addons/chan_ooh323.c, channels/chan_sip.c,
res/res_adsi.c, addons/chan_ooh323.h,
include/asterisk/callerid.h, channels/chan_phone.c,
channels/chan_dahdi.c, channels/chan_skinny.c, main/callerid.c,
channels/chan_h323.c, addons/ooh323cDriver.c,
include/asterisk/rtp_engine.h, addons/ooh323cDriver.h: More
32->64 bit codec conversions. In the process of swapping ULAW to
a place in the extended codec space, we found several unhandled
cases, where a 32-bit integer was still being used to handle a
codec field. Most of these have been fixed with this commit,
although there is at least one case (codec_dahdi) which depends
upon outside headers to be altered before a conversion can be
made. (Fixes AST-278, SWP-459)
* include/asterisk/mod_format.h: Formats need to be able to
represent all 64 codec bits.
2009-12-01 15:47 +0000 [r231741] Matthew Nicholson <mnicholson@digium.com>
* /, main/file.c: Merged revisions 231740 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r231740 | mnicholson | 2009-12-01 09:34:57 -0600 (Tue, 01 Dec
2009) | 2 lines Ignore unknown formats in ast_format_str_reduce()
and return an error if no know formats are found. ........
2009-11-30 21:47 +0000 [r231692] Kevin P. Fleming <kpfleming@digium.com>
* main/udptl.c, channels/chan_sip.c, include/asterisk/udptl.h:
Another round of UDPTL stack fixes/improvements: 1) Allow users
of UDPTL stack to associate a character-string tag with a UDPTL
session, so that log/error/debug messages generated by the UDPTL
stack can be 'connected' to the endpoint that caused them to be
generated. 2) Improve comments (and process) of calculating the
far end's maximum IFP size when redundancy mode is in use for
error correction. 3) When an IFP larger than the calculated 'far
max IFP' size is presented for writing, truncate it rather than
putting in the buffer and allowing the buffer to overflow; this
will cause the ends to retrain to a lower bit rate that produces
IFPs of an appropriate size if possible, and if not possible, the
FAX transfer will fail completely. In these cases, it is due to
the one endpoint supplying a T38FaxMaxDatagram value that is
improperly calculated and is too low to be of use; we have
configuration options available to override this behavior. 4)
Eliminate use of T38FaxMaxDatagram value in udptl.conf; it is no
longer needed.
2009-11-30 21:31 +0000 [r231616-231688] Matthew Nicholson <mnicholson@digium.com>
* include/asterisk/file.h, /, main/file.c, main/app.c,
apps/app_voicemail.c: Merged revisions 231614 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r231614 | mnicholson | 2009-11-30 15:11:44 -0600 (Mon, 30 Nov
2009) | 8 lines Remove duplicate entries from voicemail format
lists. This prevents app_voicemail from entering an infinite loop
when the same format is specified twice in the format list.
(closes issue #15625) Reported by: Shagg63 Tested by: mnicholson
Review: https://reviewboard.asterisk.org/r/429/ ........
* include/asterisk/file.h, /, main/app.c, apps/app_voicemail.c:
Reverted 231616
* include/asterisk/file.h, /, main/app.c, apps/app_voicemail.c:
Merged revisions 231614 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r231614 | mnicholson | 2009-11-30 15:11:44 -0600 (Mon, 30 Nov
2009) | 8 lines Remove duplicate entries from voicemail format
lists. This prevents app_voicemail from entering an infinite loop
when the same format is specified twice in the format list.
(closes issue #15625) Reported by: Shagg63 Tested by: mnicholson
Review: https://reviewboard.asterisk.org/r/429/ ........
2009-11-30 20:44 +0000 [r231602] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: When receiving SDP that matches the version
of the last one do not treat it as a fatal error. (closes issue
#16238) Reported by: seandarcy
2009-11-30 18:55 +0000 [r231491-231556] David Vossel <dvossel@digium.com>
* apps/app_queue.c: app_queue crashes randomly, often during
call-transfers This patch adds a ref to the queue_ent object's
parent call_queue in queue_exec() so the call_queue won't be
destroyed while the the queue_ent still holds a pointer to it.
(closes issue 0015686) Tested by: dvossel, aragon
* res/res_rtp_asterisk.c, /: Merged revisions 231441 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r231441 | dvossel | 2009-11-30 11:14:08 -0600 (Mon, 30
Nov 2009) | 11 lines fixes crash caused by RTP comfort noise
payload greater than 24 bytes AST-2009-010 (closes issue #16242)
Reported by: amorsen Patches: issue16242.diff uploaded by oej
(license 306) Tested by: amorsen, oej, dvossel ........
2009-11-30 16:53 +0000 [r231439] Tilghman Lesher <tlesher@digium.com>
* main/asterisk.dynamics (added), Makefile.rules: Export dynamic
(weak-linked) symbols correctly. (closes issue #15193) Reported
by: eliel Patches: 20091111__issue15193.diff.txt uploaded by
tilghman (license 14)
2009-11-30 16:29 +0000 [r231436] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: Fix a bug where an immediate masquerade
would cause a queued unhold frame to get lost. Now we just
indicate unhold directly after the masquerade is complete. (issue
ABE-2011)
2009-11-27 08:47 +0000 [r231401] TransNexus OSP Development <support@transnexus.com>
* apps/app_osplookup.c: 1. Modified exported variable names. 2.
Added destination port support. 3. Added new protocols. 4. Added
QoS.
2009-11-26 02:09 +0000 [r231299-231369] Tilghman Lesher <tlesher@digium.com>
* doc/CODING-GUIDELINES, main/asterisk.c: Reorder option flags.
Change guidelines so that example code is consistent with
guidelines
* main/channel.c, /: Merged revisions 231298 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r231298 | tilghman | 2009-11-25 16:31:57 -0600 (Wed, 25 Nov 2009)
| 2 lines After a frame duplication failure, unlock the channel
before returning. ........
2009-11-25 15:42 +0000 [r231189] Matthew Nicholson <mnicholson@digium.com>
* pbx/pbx_lua.c: Load pbx_lua with global symbols to allow linking
with other lua libraries. Found by Maxim Litnitskiy.
2009-11-24 20:31 +0000 [r231134] Tilghman Lesher <tlesher@digium.com>
* apps/app_queue.c: Found a few places where queue refcounts were
counted incorrectly. Also add debug statements. (closes issue
#15982, closes issue #15984) Reported by: atis Patches:
20091111__issue15982.diff.txt uploaded by tilghman (license 14)
Tested by: atis
2009-11-24 18:50 +0000 [r231058-231095] Jeff Peeler <jpeeler@digium.com>
* main/features.c: Fix erroneous hangup extension execution
ast_spawn_extension behaves differently from 1.4 in that hangups
and extensions that do not exist do not return an error, whereas
in 1.6 it does. This is now taken into account so that the
AST_FLAG_BRIDGE_HANGUP_RUN flag gets set properly. (closes issue
#16106) Reported by: ajohnson Tested by: ajohnson
* channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_pri.c:
Fix problem on digital channels due to digital flag not getting
set Changed areas in sig_pri to set the digital flag using a
callback that will also set the corresponding flag in chan_dahdi.
Modified dahdi_request slightly so that if a bearer is marked as
digital, that information is available when creating the new
channel. (closes issue #16151) Reported by: alecdavis Patch based
on bug_16151.diff.txt uploaded by alecdavis (license 585)
2009-11-24 13:52 +0000 [r231025] Matthew Nicholson <mnicholson@digium.com>
* CHANGES: Updated CHANGES file to describe the new 'd' option to
app_followme added in r230964 (related to issue #14155) Reported
by: junky
2009-11-24 04:58 +0000 [r230994] Tilghman Lesher <tlesher@digium.com>
* include/asterisk/app.h, funcs/func_strings.c, CHANGES: Add
REPLACE & PASSTHRU functions, overhaul of func_strings, fix API
docs for the ast_get_encoded_* functions. * Add REPLACE function,
which searches a given variable for a set of characters and
replaces each with a given character. * Add PASSTHRU function,
which passes a literal string back, like a NoOp for functions.
Intent is to be able to specify a literal string to another
function that takes a variable name as an argument. * Let the
array manipulation functions work with dialplan functions, in
addition to variables. This allows the array manipulation
functions to modify ASTDB and ODBC backends, assuming the
func_odbc configuration has both read and write functions.
(closes issue #15223) Reported by: ajohnson Patches:
20091112__issue15223.diff.txt uploaded by tilghman (license 14)
Tested by: lmadsen, tilghman
2009-11-23 22:37 +0000 [r230964] Matthew Nicholson <mnicholson@digium.com>
* apps/app_followme.c: Add an option to app_followme to disable the
"please hold" announcement. (closes issue #14155) Reported by:
junky Patches: M14555-trunk.diff uploaded by junky (license 177)
(modified) Tested by: junky
2009-11-23 15:45 +0000 [r230881] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c, configs/sip.conf.sample: Change fax
detection in chan_sip so it behaves as one would expect.
Internally the way T.38 is negotiated has changed and the option
no longer reflects a behavior that is valid. It will now look for
a CNG tone on received calls and if present send the call to the
'fax' extension. It is then up to the application or channel to
request the switch over to T.38.
2009-11-23 15:34 +0000 [r230773-230877] Kevin P. Fleming <kpfleming@digium.com>
* /, channels/chan_sip.c: Merged revisions 230839 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r230839 | kpfleming | 2009-11-23 09:09:24 -0600 (Mon, 23 Nov
2009) | 1 line Correct fix for issue #16268... the reporter's
original patch was very close to correct. ........
* /, channels/chan_sip.c: Merged revisions 230772 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r230772 | kpfleming | 2009-11-23 08:13:56 -0600 (Mon, 23 Nov
2009) | 5 lines Ensure that SDP parsing does not ignore the last
line of the SDP. (closes issue #16268) Reported by: sgimeno
........
2009-11-20 22:35 +0000 [r230726] David Vossel <dvossel@digium.com>
* channels/chan_iax2.c: fixes iax2 show cache locking error, thanks
alecdavis! (closes issue #16094) Reported by: alecdavis Patches:
bug16094.diff.txt uploaded by alecdavis (license 585) Tested by:
alecdavis, dvossel
2009-11-20 21:47 +0000 [r230697] Tilghman Lesher <tlesher@digium.com>
* include/asterisk/unaligned.h: Revert code in error and include
the gcc suggested workaround for the original problem, while gcc
investigates.
2009-11-20 21:01 +0000 [r230628] Matthew Nicholson <mnicholson@digium.com>
* /, main/features.c: Merged revisions 230627 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r230627 | mnicholson | 2009-11-20 14:53:06 -0600 (Fri, 20 Nov
2009) | 8 lines Copy the peer CDR's userfield to the bridge CDR
if it exists. This is necessary for the recordagentcalls option
in chan_agent to store the recorded file name in the bridge CDR.
(closes issue #14590) Reported by: msetim Patches:
queue_agent_userfield.patch uploaded by Laureano (license 265)
Tested by: Laureano, mnicholson ........
2009-11-20 17:28 +0000 [r230584] David Ruggles <thedavidfactor@gmail.com>
* doc/externalivr.txt, apps/app_externalivr.c: Fix/Implement error
events for non-existing files also include a better cmd define
for S command Review: https://reviewboard.asterisk.org/r/430/
2009-11-20 17:26 +0000 [r230509-230583] David Vossel <dvossel@digium.com>
* include/asterisk/audiohook.h, main/audiohook.c: audiohook signal
trigger on every status change (issue #14618) Review:
https://reviewboard.asterisk.org/r/434/
* /, apps/app_mixmonitor.c: Merged revisions 230508 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r230508 | dvossel | 2009-11-19 15:22:46 -0600 (Thu, 19
Nov 2009) | 10 lines fixes MixMonitor thread not exiting when
StopMixMonitor is used (closes issue #16152) Reported by: AlexMS
Patches: stopmixmonitor_1.4.diff uploaded by dvossel (license
671) Tested by: dvossel, AlexMS Review:
https://reviewboard.asterisk.org/r/424/ ........
2009-11-19 14:53 +0000 [r230438] David Ruggles <thedavidfactor@gmail.com>
* apps/app_externalivr.c: Basic cleanup of ExternalIVR: cleaned up
argument parsing; implemented good coding practices where
applicable; replaced most notice level logging with verbose
logging; replaced warning messages that terminated with error
messages; fixed memory leak identified by russellb
2009-11-16 16:40 +0000 [r230343-230381] Kevin P. Fleming <kpfleming@digium.com>
* apps/app_fax.c: Fix another buglet in T.38 session teardown at
the end of FAX sessions.
* apps/app_fax.c: Ensure that only one end of a T.38 session
initiates teardown at completion.
2009-11-16 01:49 +0000 [r230314] TransNexus OSP Development <support@transnexus.com>
* apps/app_osplookup.c: 1. Added SIP Diversion support. 2. Fixed
compile warning for UUID.
2009-11-15 17:23 +0000 [r230247] Kevin P. Fleming <kpfleming@digium.com>
* /, channels/chan_iax2.c: Merged revisions 230246 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r230246 | kpfleming | 2009-11-15 11:19:06 -0600 (Sun, 15
Nov 2009) | 6 lines Correct mistaken option name in error
message. The configuration option for allowing hosts to make
non-token-based calls is 'calltokenoptional', not
'calltokenignore'. (reported on asterisk-users) ........
2009-11-15 07:53 +0000 [r230217] Tilghman Lesher <tlesher@digium.com>
* include/asterisk/channel.h: Increase maximum length of language
buffers (closes issue #16217) Reported by: dsessions
2009-11-13 22:00 +0000 [r230145] Joshua Colp <jcolp@digium.com>
* /, channels/chan_sip.c: Merged revisions 230144 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r230144 | file | 2009-11-13 16:00:19 -0600 (Fri, 13 Nov 2009) | 8
lines Respect the maddr parameter in the Via header. (closes
issue #14446) Reported by: frawd Patches: via_maddr.patch
uploaded by frawd (license 610) Tested by: frawd ........
2009-11-13 20:42 +0000 [r230111] Tilghman Lesher <tlesher@digium.com>
* apps/app_dial.c, channels/chan_sip.c, apps/app_meetme.c,
apps/app_fax.c, configs/manager.conf.sample,
res/res_musiconhold.c, include/asterisk/manager.h,
channels/chan_iax2.c, apps/app_queue.c, CHANGES,
res/res_monitor.c, main/cdr.c, main/channel.c, main/manager.c,
main/features.c, apps/app_minivm.c, apps/app_chanspy.c,
apps/app_voicemail.c: Display a list of channel variables in each
channel-oriented event. (Closes AST-33) Reviewboard:
https://reviewboard.asterisk.org/r/368/
2009-11-13 19:44 +0000 [r229912-230039] Joshua Colp <jcolp@digium.com>
* channels/chan_local.c, /: Merged revisions 230038 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r230038 | file | 2009-11-13 13:44:07 -0600 (Fri, 13 Nov
2009) | 9 lines Fix a crash caused by two threads thinking they
should both free the chan_local private structure when only one
should. (closes issue #15314) Reported by: sroberts Patches:
Issue15314_Move_Nulling_owner.patch uploaded by davidw (license
780) Tested by: davidw, lottc ........
* UPGRADE.txt, apps/app_chanisavail.c, CHANGES: Store the cause
code that is returned when trying to create a channel in
ChanIsAvail in the AVAILCAUSECODE dialplan variable instead of
overwriting the device state in AVAILSTATUS. (closes issue
#14426) Reported by: macli
* /: Merged revisions 229965 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r229965 | file | 2009-11-13 11:19:59 -0600 (Fri, 13 Nov 2009) | 6
lines Document a limitation in the AVAILSTATUS variable from
ChanIsAvail and provide a workaround for it that does not change
existing behavior. (closes issue #14426) Reported by: macli
........
* channels/chan_sip.c: Fix T.38 negotiation regression introduced
with the SDP parser changes.
2009-11-13 10:53 +0000 [r229819-229871] Olle Johansson <oej@edvina.net>
* main/loader.c: Fixing trunk in a way so that it compiles again.
Thanks, Philippe :-)
* addons/cdr_mysql.c: If CDR logging is disabled, it's considered a
FAILURE
* configs/modules.conf.sample, CHANGES, main/asterisk.c,
main/loader.c: Add the capability to require a module to be
loaded, or else Asterisk exits. Review:
https://reviewboard.asterisk.org/r/426/
2009-11-13 03:16 +0000 [r229788] TransNexus OSP Development <support@transnexus.com>
* apps/app_osplookup.c: Added full number portability parameter
support.
2009-11-12 23:43 +0000 [r229750-229754] Jason Parker <jparker@digium.com>
* configs/alsa.conf.sample: Update sample config for ALSA mute and
noaudiocapture
* channels/chan_alsa.c: Add mute functionality. Add config option
to not try to open capture device. Adds "console {mute|unmute}"
CLI command. Adds mute and noaudiocapture config options (will
update sample configs shortly). (closes issue #14673) Reported
by: Nick_Lewis Patches: chan_alsa.c-oneway3.patch uploaded by
Nick Lewis (license 657) Tested by: qwell
* channels/chan_oss.c: Fix mute toggling on OSS channels.
2009-11-12 16:44 +0000 [r229670] David Vossel <dvossel@digium.com>
* funcs/func_audiohookinherit.c, /: Merged revisions 229669 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r229669 | dvossel | 2009-11-12 10:41:49 -0600 (Thu, 12 Nov 2009)
| 6 lines fixes merging error, datastore was being freed in the
wrong function. (closes issue #16219) Reported by: aragon
........
2009-11-12 13:54 +0000 [r229639] Leif Madsen <lmadsen@digium.com>
* configs/sip.conf.sample: Update sip.conf.sample. Just updating a
spelling error and some capitalization in a documentation update
that Olle added. May the Swenglish be with you.
2009-11-12 10:24 +0000 [r229606-229607] Olle Johansson <oej@edvina.net>
* configs/sip.conf.sample: Clarification
* configs/sip.conf.sample: Clarify some security issues early in
the sample configuration
2009-11-11 20:47 +0000 [r229568] David Ruggles <thedavidfactor@gmail.com>
* doc/externalivr.txt: Remove non-functional feature from
ExternalIVR documentation Remove non-functional socket
implementation of ExternalIVR from documentation (closes issue
#16225) Reported by: thedavidfactor Patches:
externalivr.txt.20091111.1542.patch uploaded by thedavidfactor
(license 903)
2009-11-11 19:48 +0000 [r229460-229499] David Brooks <dbrooks@digium.com>
* main/pbx.c, /: Merged revisions 229498 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r229498 | dbrooks | 2009-11-11 13:46:19 -0600 (Wed, 11 Nov 2009)
| 8 lines Solaris doesn't like NULL going to ast_log Solaris will
crash if NULL is passed to ast_log. This simple patch simply uses
S_OR to get around this. (closes issue #15392) Reported by:
yrashk ........
* apps/app_softhangup.c: Flags not initialized in app_softhangup.c,
causing undefined behavior Trivial patch [kobaz] to initialize an
ast_flags = {0} (closes issue #16129) Reported by: kobaz
2009-11-11 14:30 +0000 [r229431] Leif Madsen <lmadsen@digium.com>
* CHANGES: Update CHANGES file. Updating the CHANGES file after
noticing an email on the asterisk-dev mailing list from Russell.
(issue #15874)
2009-11-10 22:14 +0000 [r229361] Tilghman Lesher <tlesher@digium.com>
* main/pbx.c, /: Merged revisions 229360 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r229360 | tilghman | 2009-11-10 16:09:16 -0600 (Tue, 10 Nov 2009)
| 12 lines If two pattern classes start with the same digit and
have the same number of characters, they will compare equal. The
example given in the issue report is that of [234] and [246],
which have these characteristics, yet they are clearly not
equivalent. The code still uses these two characteristics, yet
when the two scores compare equal, an additional check will be
done to compare all characters within the class to verify
equality. (closes issue #15421) Reported by: jsmith Patches:
20091109__issue15421__2.diff.txt uploaded by tilghman (license
14) Tested by: jsmith, thedavidfactor ........
2009-11-10 22:01 +0000 [r229356] David Ruggles <thedavidfactor@gmail.com>
* doc/externalivr.txt: Merged revisions 229355 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r229355 | diruggles | 2009-11-10 16:45:15 -0500 (Tue, 10 Nov
2009) | 9 lines Fix ExternalIVR Documentation Remove
documentation for event that doesn't function (closes issue
#16220) Reported by: thedavidfactor Patches:
externalivr.txt.20091110.1622.patch uploaded by thedavidfactor
(license 903) ........
2009-11-10 21:22 +0000 [r229351] Tilghman Lesher <tlesher@digium.com>
* apps/app_stack.c: When GOSUB is invoked within an AGI, it may not
exit correctly. (closes issue #16216) Reported by: atis Patches:
20091110__atis_work.diff.txt uploaded by tilghman (license 14)
Tested by: atis
2009-11-10 20:06 +0000 [r229282] Joshua Colp <jcolp@digium.com>
* /, codecs/codec_g726.c: Merged revisions 229281 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r229281 | file | 2009-11-10 16:03:14 -0400 (Tue, 10 Nov 2009) | 8
lines Remove broken support for direct transcoding between G.726
RFC3551 and G.726 AAL2. On some systems the translation core
would actually consider g726aal2 -> g726 -> signed linear to be a
quicker path then g726aal2 -> signed linear which exposed this
problem. (closes issue #15504) Reported by: globalnetinc ........
2009-11-10 17:33 +0000 [r229228] David Ruggles <thedavidfactor@gmail.com>
* /, doc/externalivr.txt: Merged revisions 229191 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r229191 | diruggles | 2009-11-10 12:23:59 -0500 (Tue, 10 Nov
2009) | 11 lines Document ExternalIVR event tag collision
ExternalIVR uses the D tag for two different event types. This
documents that behavior and how to differentiate between the two
cases. Also includes a minor spelling fix and clarification
(closes issue #16211) Reported by: thedavidfactor Patches:
externalivr.txt.20091109.1507.patch uploaded by thedavidfactor
(license 903) ........
2009-11-10 17:16 +0000 [r229168] David Vossel <dvossel@digium.com>
* /, channels/chan_iax2.c: Merged revisions 229167 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r229167 | dvossel | 2009-11-10 11:15:57 -0600 (Tue, 10
Nov 2009) | 9 lines don't crash on log message in solaris
AST-2009-006 (closes issue #16206) Reported by: bklang Tested by:
bklang ........
2009-11-10 15:53 +0000 [r229102] Matthew Nicholson <mnicholson@digium.com>
* channels/chan_sip.c: Reverted revision 201717. (closes issue
0016175) Reported by: paul-tg
2009-11-10 15:27 +0000 [r229093] David Vossel <dvossel@digium.com>
* res/res_config_pgsql.c: fixes pgsql double free of threadstorage
A thread storage variable was being freed incorrectly, which
resulted in a double free if two queries were made in the same
thread. (closes issue #16011) Reported by: cristiandimache
Patches: issue16011.diff uploaded by dvossel (license 671)
2009-11-10 11:16 +0000 [r229050] Gavin Henry <ghenry@suretecsystems.com>
* contrib/scripts/asterisk.ldap-schema: Schema file additions *
Added AsteriskDialplan, AsteriskAccount and AsteriskMailbox
objectClasses to allow standalone dialplan, account and mailbox
entries (STRUCTURAL) * Added new Fields: - AstAccountLanguage,
AstAccountTransport, AstAccountPromiscRedir, -
AstAccountAccountCode, AstAccountSetVar, AstAccountAllowOverlap,
- AstAccountVideoSupport, AstAccountIgnoreSDPVersion * Removed
redundant IPaddr (there's already IPAddress) - Gives more
configuration Flags for SIP-Users available (tested) - Allows to
create Asterisk Attributes in defined Asterisk ObjectClasses
without extensibleObject (which really should be the last
resort); gives also additional possibilities for LDAP-filter
(closes issue #15874) Reported by: Medozas Patches:
asterisk.ldap-schema.patch uploaded by Medozas (license 41)
Tested by: Medozas, suretec
2009-11-09 22:50 +0000 [r229015] Terry Wilson <twilson@digium.com>
* channels/chan_local.c: Don't crash when bridge->tech_pvt == NULL
This is a similar solution to what is in place for chan_agent
(closes issue #16003) Reported by: atis Tested by: twilson
2009-11-09 17:17 +0000 [r228979] Tilghman Lesher <tlesher@digium.com>
* channels/iax2-parser.c: Don't try to convert a 64-bit integer,
where only a 32-bit integer is stored. (closes issue #16194)
Reported by: habile
2009-11-09 16:28 +0000 [r228947] Matthew Nicholson <mnicholson@digium.com>
* configs/queues.conf.sample, CHANGES, apps/app_queue.c: Add the
'relative-periodic-announce' option to app_queue to allow for
calculating the time of announcments from the end of the previous
announcment rather than from the beginning. (closes issue #15260)
Reported by: tonils
2009-11-09 15:38 +0000 [r228897] Leif Madsen <lmadsen@digium.com>
* main/channel.c, /: Merged revisions 228896 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r228896 | lmadsen | 2009-11-09 09:37:43 -0600 (Mon, 09 Nov 2009)
| 6 lines Update WARNING message. Update a WARNING message to
give a suggested fix when encountered. (closes issue #16198)
Reported by: atis Tested by: atis ........
2009-11-09 14:37 +0000 [r228858] Matthew Nicholson <mnicholson@digium.com>
* /, include/asterisk/lock.h: Merged revisions 228827 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r228827 | mnicholson | 2009-11-09 08:16:03 -0600 (Mon,
09 Nov 2009) | 8 lines Perform limited bounds checking when
destroying ast_mutex_t structures to make sure we don't try to
use negative indices. (closes issue #15588) Reported by: zerohalo
Patches: 20090820__issue15588.diff.txt uploaded by tilghman
(license 14) Tested by: zerohalo ........
2009-11-09 07:37 +0000 [r228798] Tilghman Lesher <tlesher@digium.com>
* addons/cdr_mysql.c, main/event.c, channels/chan_console.c,
res/res_pktccops.c, main/loader.c: Fix various problems detected
with Valgrind. * chan_console accessed pvts after deallocation. *
cdr_mysql stored a pointer that was freed by realloc() * The
module loader did not check usecount on shutdown, which led to
chan_iax2 reading a timer that was already unloaded. * The event
subsystem sometimes creates an event with no IEs. Due to a corner
condition, the code would read beyond the memory boundary. *
res_pktccops did not correctly check whether its monitor thread
was started. (closes issue #16062) Reported by: alexanderheinz
Patches: 20091109__issue16062.diff.txt uploaded by tilghman
(license 14) Tested by: tilghman
2009-11-07 17:02 +0000 [r228766] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
* contrib/init.d/rc.debian.asterisk: Add LSB headers to the Debian
init.d script See also issue #14864 .
2009-11-06 22:35 +0000 [r228693] David Vossel <dvossel@digium.com>
* main/channel.c, /: Merged revisions 228692 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r228692 | dvossel | 2009-11-06 16:33:27 -0600 (Fri, 06 Nov 2009)
| 9 lines fixes audiohook write crash occuring in chan_spy
whisper mode. After writing to the audiohook list in ast_write(),
frames were being freed incorrectly. Under certain conditions
this resulted in a double free crash. (closes issue #16133)
Reported by: wetwired (closes issue #16045) Reported by:
bluecrow76 Patches: issue16045.diff uploaded by dvossel (license
671) Tested by: bluecrow76, dvossel, habile ........
2009-11-06 22:32 +0000 [r228691] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c, CHANGES, channels/sig_pri.c: Created
standard location to add options to chan_dahdi for ISDN dialing.
Dial(DAHDI/g1[/extension[/options]]) Current options:
K(<keypad_digits>) R Reverse charging indication (Collect calls)
The earlier Dial(DAHDI/g1[/K<keypad_digits>][/extension] format
was variable and did not allow for the easy addition of more
options. The earlier 'C' prefix character for reverse charge
indiation would conflict with the a-d DTMF digits if ISDN uses
them.
2009-11-06 22:07 +0000 [r228661] David Brooks <dbrooks@digium.com>
* tests/test_amihooks.c: ami_testhooks.c automatically registers
hook ami_testhooks.c was registering for AMI events upon module
load. Moved the registration to its own CLI command. Added CLI
command for unregistering the hook. Changed some of the wording,
removed unnecessary arguments/parameters. Reported by: rmudgett
2009-11-06 22:02 +0000 [r228658-228659] Mark Michelson <mmichelson@digium.com>
* addons/chan_ooh323.c: Make compilation of chan_ooh323 disabled by
default. All addons modules should be disabled by default,
requiring the user to turn them on if desired. After all, these
are addons we're talking about here.
* addons/ooh323c/src/ooh323.c, addons/ooh323c/src/ooh245.c: Get
chan_ooh323 to compile with gcc 4.2. For some reason, the code
compiles just fine with later versions of GCC, but this one
requires some weird double casting in order to get rid of all
warnings. Whatever.
2009-11-06 19:53 +0000 [r228621] Richard Mudgett <rmudgett@digium.com>
* main/frame.c: Fix compiler warning gcc 4.2.4 found
2009-11-06 19:47 +0000 [r228620] Matthew Nicholson <mnicholson@digium.com>
* funcs/func_base64.c, /, main/utils.c: Merged revisions 228378 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r228378 | mnicholson | 2009-11-06 10:26:59 -0600 (Fri, 06 Nov
2009) | 8 lines Properly handle '=' while decoding base64
messages and null terminate strings returned from BASE64_DECODE.
(closes issue #15271) Reported by: chappell Patches:
base64_fix.patch uploaded by chappell (license 8) Tested by:
kobaz ........
2009-11-06 19:38 +0000 [r228616] Tilghman Lesher <tlesher@digium.com>
* channels/chan_nbs.c, addons/chan_mobile.c: Missed these two
channel drivers on the codec_bits merge
2009-11-06 18:37 +0000 [r228499-228548] Joshua Colp <jcolp@digium.com>
* /, channels/chan_sip.c: Merged revisions 228547 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r228547 | file | 2009-11-06 14:32:58 -0400 (Fri, 06 Nov 2009) | 4
lines Don't overwrite caller ID name on a trunk with the
configured fullname when using users.conf (issue ABE-1989)
........
* doc/tex/localchannel.tex: Fix the localchannel.tex file.
2009-11-06 17:22 +0000 [r228420-228441] David Vossel <dvossel@digium.com>
* codecs/codec_ilbc.c: Fixes merging issue from 1.4, frame data is
held in data.ptr in trunk
* /, codecs/codec_ilbc.c: Merged revisions 228418 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r228418 | dvossel | 2009-11-06 11:07:13 -0600 (Fri, 06 Nov 2009)
| 13 lines fixes segfault in iLBC For reasons not yet known, it
appears possible for an ast_frame to have a datalen greater than
zero while the actual data is NULL during Packet Loss
Concealment. Most codecs don't support PLC so this doesn't affect
them. This patch catches the malformed frame and prevents the
crash from occuring. Additional efforts to determine why it is
possible for a frame to look like this are still being
investigated. (issue #16979) ........
2009-11-06 16:42 +0000 [r228410] Joshua Colp <jcolp@digium.com>
* /, main/abstract_jb.c: Merged revisions 228409 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r228409 | file | 2009-11-06 12:41:20 -0400 (Fri, 06 Nov 2009) | 7
lines Fix a bug caused by a partially invalid frame (from the
jitterbuffer) passing through the Asterisk core. (closes issue
#15560) Reported by: jvandal (closes issue #15709) Reported by:
covici ........
2009-11-06 15:42 +0000 [r228268-228339] David Vossel <dvossel@digium.com>
* /, main/astfd.c: Merged revisions 228338 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r228338 | dvossel | 2009-11-06 09:41:41 -0600 (Fri, 06 Nov 2009)
| 5 lines fixes crash in astfd.c (closes issue #15981) Reported
by: slavon ........
* funcs/func_audiohookinherit.c: fixes memory leak in
func_audiohookinherit.c (closes issue #15394) Reported by: boroda
Patches: bug15394_memoryleak_diff2.txt uploaded by dbrooks
(license 790) Tested by: dbrooks, boroda
2009-11-05 22:59 +0000 [r228233] Mark Michelson <mmichelson@digium.com>
* funcs/func_cdr.c: Fix XML in func_cdr.c
2009-11-05 22:12 +0000 [r228191-228196] Tilghman Lesher <tlesher@digium.com>
* apps/app_meetme.c: Yet another error message in the dialplan
(thanks, rmudgett/russellb)
* apps/app_meetme.c: MEETME_INFO should not return a literal error
message to the dialplan. (closes issue #15450) Reported by:
JimVanM Patches: meetmeinfopatch.diff.txt uploaded by dbrooks
(license 790) Tested by: JimVanM
2009-11-05 21:23 +0000 [r228189] Jeff Peeler <jpeeler@digium.com>
* apps/app_chanspy.c: Fix the fix for chanspy option o In 224178, I
assumed the uploaded patch was correct as it had received
positive feedback. The flags were being checked in the incorrect
location. Upon testing the fix this time it was also found that
the flags from the dialplan weren't being copied to the
chanspy_translation_helper. (closes issue #16167) Reported by:
marhbere
2009-11-05 19:34 +0000 [r228145] David Brooks <dbrooks@digium.com>
* channels/chan_misdn.c, /: Merged revisions 228078 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r228078 | dbrooks | 2009-11-05 12:59:41 -0600 (Thu, 05
Nov 2009) | 9 lines chan_misdn Asterisk 1.4.27-rc2 crash Crash
related to chan_misdn connection. Patch submitted by
gknispel_proformatique, tested by francesco_r. "I have many crash
since i have upgraded to Asterisk 1.4.27-rc2. Attached a full
bt." This patch zeros out an ast_frame. (closes issue #16041)
Reported by: francesco_r ........
2009-11-05 19:16 +0000 [r228080] Jason Parker <jparker@digium.com>
* channels/chan_vpb.cc, /: Merged revisions 228079 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r228079 | qwell | 2009-11-05 13:14:25 -0600 (Thu, 05 Nov
2009) | 8 lines Fix crash on VPB exception when no hardware is
present. (closes issue #14970) Reported by: tzafrir Patches:
vpb_exception.diff uploaded by tzafrir (license 46) Tested by:
markwaters ........
2009-11-05 17:26 +0000 [r228015-228049] Tilghman Lesher <tlesher@digium.com>
* main/frame.c: Rework codecs command to comply with the 64-bit
scheme
* apps/app_externalivr.c: Don't crash if no arguments are passed.
(closes issue #16119) Reported by: thedavidfactor
2009-11-04 23:50 +0000 [r227914-227945] Jeff Peeler <jpeeler@digium.com>
* /, res/res_monitor.c: Merged revisions 227944 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r227944 | jpeeler | 2009-11-04 17:47:08 -0600 (Wed, 04 Nov 2009)
| 14 lines Fix incorrect filename comparsion after monitor file
change The logic to detect if a requested file is indeed a
different file from the current file was incorrect. The main
issue being confusion of the use of filename_base which was
previously set without pathing information and then compared to
another full path. Robust file comparison logic has been added to
properly check if two files are the same even if symlinks are
used. (closes issue #15313) Reported by: caspy Patches:
20091103__issue15313__1.4.diff.txt uploaded by jpeeler (license
325) but mostly tilghman's work ........
* addons/chan_ooh323.c: Update chan_ooh323 to support the expanded
codec bitfield from 227580.
2009-11-04 22:10 +0000 [r227898] Alexandr Anikin <may@telecom-service.ru>
* addons/ooh323c/src/oochannels.h,
addons/ooh323c/src/ooCmdChannel.h, addons/chan_ooh323.c,
addons/ooh323c/src/printHandler.h, addons/ooh323c/src/ooq931.h,
addons/ooh323c/src/ootrace.h, addons/chan_ooh323.h,
addons/ooh323c/src/ooasn1.h, addons/ooh323c/src/ootypes.h,
addons/ooh323c/src/ooLogChan.c, addons/ooh323c/src/ooStackCmds.c,
addons/ooh323c/src/errmgmt.c, addons/ooh323c/src/ooTimer.c,
addons/ooh323c/src/ooLogChan.h,
addons/ooh323c/src/ooCapability.c,
addons/ooh323c/src/ooStackCmds.h, addons/ooh323c/src/dlist.c,
addons/ooh323c/src/eventHandler.c,
addons/ooh323c/src/ooCapability.h,
addons/ooh323c/src/eventHandler.h, addons/Makefile,
addons/ooh323cDriver.c, addons/ooh323c/src/ooDateTime.c,
addons/ooh323c/src/rtctype.c, addons/ooh323cDriver.h,
addons/ooh323c/src/ooCalls.c, addons/ooh323c/src/encode.c,
addons/ooh323c/src/ooUtils.c, addons/ooh323c/src/ooGkClient.c,
addons/ooh323c/src/ooDateTime.h, addons/ooh323c/src/ooCalls.h,
addons/ooh323c/src/ooh323ep.c, addons/ooh323c/src/ooGkClient.h,
addons/ooh323c/src/ooports.c, addons/ooh323c/src/ooh323ep.h,
addons/ooh323c/src/memheap.c, addons/ooh323c/src/ooh323.c,
addons/ooh323c/src/h323/H323-MESSAGESDec.c,
addons/ooh323c/src/ooh245.c, addons/ooh323c/src/memheap.h,
addons/ooh323c/src/ooh323.h, addons/ooh323c/src/decode.c,
addons/ooh323c/src/context.c, addons/ooh323c/src/perutil.c,
addons/ooh323c/src/h323/MULTIMEDIA-SYSTEM-CONTROLDec.c,
addons/ooh323c/src/ooh245.h, addons/ooh323c/src/ooSocket.c,
addons/ooh323c/src/h323/H235-SECURITY-MESSAGESDec.c,
addons/ooh323c/src/oochannels.c,
addons/ooh323c/src/ooCmdChannel.c,
addons/ooh323c/src/printHandler.c, addons/ooh323c/src/ooSocket.h,
addons/ooh323c/src/ooCommon.h, addons/ooh323c/src/ooq931.c,
addons/ooh323c/src/ootrace.c: Reworked chan_ooh323 channel
module. Many architectural and functional changes. Main changes
are threading model chanes (many thread in ooh323 stack instead
of one), modifications and improvements in signalling part,
additional codecs support (726, speex), t38 mode support. This
module tested and used in production environment. (closes issue
#15285) Reported by: may213 Tested by: sles, c0w, OrNix Review:
https://reviewboard.asterisk.org/r/324/
2009-11-04 21:39 +0000 [r227829-227897] Matthew Nicholson <mnicholson@digium.com>
* apps/app_dial.c, CHANGES: Added the 'a' option to app dial and
modified app_dial to set the answertime when the called channel
answers. This change causes answertime to be correct even if the
called channel hangs up during an announcement triggered by the
A() option. (closes issue #15936) Reported by: falves11 Patches:
dial-macro-billsec-fix1.diff uploaded by mnicholson (license 96)
dial-caller-answer1.diff uploaded by mnicholson (license 96)
Tested by: falves11, mnicholson
* apps/app_dial.c, /: Merged revisions 227827 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r227827 | mnicholson | 2009-11-04 14:52:27 -0600 (Wed, 04 Nov
2009) | 10 lines This patch modifies the Dial application to
monitor the calling channel for hangups while playing back
announcements. (closes issue #16005) Reported by: falves11
Patches: dial-announce-hangup-fix1.diff uploaded by mnicholson
(license 96) Tested by: mnicholson, falves11 Review:
https://reviewboard.asterisk.org/r/407/ ........
2009-11-04 20:35 +0000 [r227824] Tilghman Lesher <tlesher@digium.com>
* include/asterisk/unaligned.h: Fixes for gcc 4.4
2009-11-04 20:13 +0000 [r227759] Matthew Nicholson <mnicholson@digium.com>
* channels/chan_sip.c: Modify the SDP parsing code to parse session
and media level items separately. With the new code, media level
proprieties should no longer be confused with session level
proprieties. This change also reorganizes some of the SDP parsing
code which should make it easier to manage in the future. (closes
issue #14994) Reported by: frawd Tested by: frawd, mnicholson,
file Review: https://reviewboard.asterisk.org/r/414/
2009-11-04 19:26 +0000 [r227712-227739] Joshua Colp <jcolp@digium.com>
* /, static-http/prototype.js: Merged revisions 227735 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r227735 | file | 2009-11-04 15:25:37 -0400 (Wed, 04 Nov
2009) | 5 lines Fix a security issue where it may be possible for
someone to execute a cross-site AJAX request exploit.
(AST-2009-009) ........
* /, channels/chan_sip.c: Merged revisions 227700 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r227700 | file | 2009-11-04 15:17:39 -0400 (Wed, 04 Nov 2009) | 5
lines Fix a security issue where sending a REGISTER with a
differing username in the From URI and Authorization header would
reveal whether it was valid or not. (AST-2009-008) ........
2009-11-04 16:41 +0000 [r227646] Mark Michelson <mmichelson@digium.com>
* main/frame.c: Add a couple more casts so that code compiles
correctly.
2009-11-04 16:35 +0000 [r227645] Tilghman Lesher <tlesher@digium.com>
* include/asterisk/pbx.h: mmichelson reported a compilation error
related to codec bit expansion that should be resolved with a
simple include of frame_defs.h
2009-11-04 16:25 +0000 [r227643] Jeff Peeler <jpeeler@digium.com>
* channels/chan_dahdi.c: fix trunk building
2009-11-04 16:17 +0000 [r227579-227615] Tilghman Lesher <tlesher@digium.com>
* channels/chan_sip.c, channels/chan_iax2.c: Two other trunk build
fixes (reported by seanbright on #asterisk-dev)
* addons/format_mp3.c: Fix trunk building
* main/udptl.c, main/autoservice.c, apps/app_dahdibarge.c,
main/frame.c, channels/chan_local.c, main/rtp_engine.c,
include/asterisk/autoconfig.h.in, apps/app_record.c,
apps/app_test.c, bridges/bridge_softmix.c,
apps/app_alarmreceiver.c, codecs/ex_alaw.h, codecs/ex_adpcm.h,
formats/format_wav_gsm.c, formats/format_sln16.c,
codecs/ex_gsm.h, channels/chan_iax2.c, main/indications.c,
res/res_rtp_multicast.c, channels/chan_dahdi.c,
include/asterisk/bridging_technology.h, pbx/pbx_spool.c,
channels/sig_analog.c, include/asterisk/audiohook.h,
channels/chan_skinny.c, configure, main/strcompat.c,
include/asterisk/compat.h, formats/format_pcm.c, main/features.c,
channels/chan_alsa.c, apps/app_amd.c, formats/format_h263.c,
apps/app_url.c, apps/app_externalivr.c, formats/format_jpeg.c,
main/bridging.c, codecs/ex_ulaw.h, apps/app_milliwatt.c,
formats/format_gsm.c, apps/app_dial.c, main/pbx.c,
formats/format_wav.c, channels/chan_bridge.c, apps/app_echo.c,
apps/app_fax.c, include/asterisk/slin.h, channels/chan_agent.c,
configure.ac, formats/format_ogg_vorbis.c, apps/app_disa.c,
include/asterisk/unaligned.h, codecs/ex_speex.h,
include/asterisk/channel.h, apps/app_talkdetect.c,
channels/iax2-parser.c, apps/app_speech_utils.c,
channels/iax2-parser.h, channels/chan_misdn.c,
apps/app_waitforring.c, channels/iax2.h, codecs/codec_dahdi.c,
main/audiohook.c, apps/app_chanspy.c, formats/format_g726.c,
include/asterisk/frame_defs.h (added),
include/asterisk/translate.h, include/asterisk/slinfactory.h,
channels/chan_unistim.c, channels/chan_vpb.cc,
channels/chan_multicast_rtp.c, formats/format_sln.c,
apps/app_meetme.c, apps/app_dictate.c, codecs/ex_g722.h,
codecs/ex_g726.h, channels/chan_gtalk.c, res/res_musiconhold.c,
apps/app_followme.c, formats/format_siren7.c,
include/asterisk/abstract_jb.h, main/asterisk.exports,
main/channel.c, formats/format_ilbc.c, channels/chan_phone.c,
main/dial.c, main/manager.c, funcs/func_volume.c, res/res_agi.c,
apps/app_mp3.c, main/app.c, doc/codec-64bit.txt (added),
formats/format_h264.c, include/asterisk/rtp_engine.h,
include/asterisk/frame.h, formats/format_siren14.c,
codecs/ex_ilbc.h, channels/chan_mgcp.c, apps/app_jack.c,
res/res_rtp_asterisk.c, apps/app_nbscat.c, channels/chan_sip.c,
codecs/ex_lpc10.h, apps/app_festival.c, main/slinfactory.c,
main/translate.c, res/res_adsi.c, channels/chan_console.c,
channels/h323/chan_h323.h, channels/sig_pri.c, apps/app_queue.c,
channels/chan_oss.c, channels/chan_jingle.c,
formats/format_vox.c, include/asterisk/bridging.h,
main/abstract_jb.c, main/file.c, channels/chan_h323.c,
formats/format_g723.c, codecs/codec_ulaw.c, apps/app_sms.c,
include/asterisk/pbx.h, main/dsp.c, formats/format_g729.c: Expand
codec bitfield from 32 bits to 64 bits. Reviewboard:
https://reviewboard.asterisk.org/r/416/
* configure, include/asterisk/autoconfig.h.in, configure.ac:
chan_misdn will fail to compile if the redirect_dn member is
missing
2009-11-04 08:22 +0000 [r227545] Olle Johansson <oej@edvina.net>
* main/manager.c: Add destruction of iterators to avoid problems
with refcounters (per Russell's review of another patch)
2009-11-04 03:15 +0000 [r227509] Tilghman Lesher <tlesher@digium.com>
* apps/app_queue.c: Don't crash when state_interface is NULL.
2009-11-03 22:13 +0000 [r227462-227464] Russell Bryant <russell@digium.com>
* res/res_pktccops.c: Resolve another warning.
* main/manager.c, pbx/pbx_config.c: Resolve a warning from gcc
4.4.1.
* channels/chan_mgcp.c: Resolve some dev-mode warnings.
2009-11-03 21:26 +0000 [r227448] David Brooks <dbrooks@digium.com>
* main/manager.c, include/asterisk/manager.h, tests/test_amihooks.c
(added): AMI hook interface This patch, originally submitted by
jozza, enables custom modules to send actions to AMI and receive
messages from AMI via a hook interface. Included is a simple test
module to illustrate the interface. (closes issue #14635)
Reported by: jozza Review:
https://reviewboard.asterisk.org/r/412/
2009-11-03 21:21 +0000 [r227435] Matthew Nicholson <mnicholson@digium.com>
* main/cdr.c, apps/app_forkcdr.c, configs/cdr_custom.conf.sample,
funcs/func_cdr.c, main/features.c, include/asterisk/cdr.h,
CHANGES: This patch adds a sequence field to CDRs that can be
combined with the linkedid or uniqueid field to uniquely identify
a CDR. (closes issue #15180) Reported by: Nick_Lewis Patches:
cdr-sequence10.diff uploaded by mnicholson (license 96) Tested
by: mnicholson
2009-11-03 21:16 +0000 [r227424] Joshua Colp <jcolp@digium.com>
* configs/queues.conf.sample, apps/app_queue.c: Add support for
using a hint when configuring a state interface using the format
hint:<extension>@<context>. (closes issue #15168) Reported by:
p_lindheimer Patches: queue_extenstate5_1.4.svn.patch uploaded by
GameGamer43 (license 894)
2009-11-03 19:59 +0000 [r227372] Jason Parker <jparker@digium.com>
* Makefile, main/Makefile: Fix some build issues on Solaris.
(closes issue #14517) (SWP-109) Reported by: asgaroth Patches:
bug_14517.diff uploaded by snuffy (license 35) Tested by:
asgaroth, snuffy, dougm, qwell
2009-11-03 19:48 +0000 [r227361-227368] Leif Madsen <lmadsen@digium.com>
* apps/app_controlplayback.c: Change warning message to debug
message. app_controlplayback outputs a warning, when in fact it
is normal. (closes issue #16071) Reported by: atis Patches:
controlplayback_warning.patch uploaded by atis (license 242)
* configs/extensions.conf.sample: Additional fixes to the
extensions.conf.sample file. Update the extensions.conf.sample
[stdexten] context so that we use the variable instead of
requiring it to be passed explicitly. Also updated uses of the
[stdexten] context throughout. (closes issue #15858) Reported by:
pprindeville Patches: stdexten-context-update.txt uploaded by
lmadsen (license 10) Tested by: pprindeville
2009-11-03 18:22 +0000 [r227298] Matthew Nicholson <mnicholson@digium.com>
* channels/chan_sip.c: Fixed a spelling error in the q850 reason
header option in the output of sip show settings.
2009-11-03 17:58 +0000 [r227277] Richard Mudgett <rmudgett@digium.com>
* /: Recorded merge of revisions 227275 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r227275 | rmudgett | 2009-11-03 11:55:47 -0600 (Tue, 03 Nov 2009)
| 4 lines Make sure the outgoing flag is cleared if a new channel
fails to get created for outgoing calls. This is the relevant
portion of asterisk/trunk -r226648 ........
2009-11-03 17:56 +0000 [r227276] Tilghman Lesher <tlesher@digium.com>
* channels/chan_mgcp.c: Code guidelines fixes only
2009-11-03 17:12 +0000 [r227238] David Vossel <dvossel@digium.com>
* channels/chan_sip.c: user.conf entries in SIP were not having
their peer type set. (closes issue #16120) Reported by: jsmith
2009-11-03 16:56 +0000 [r227237] Olle Johansson <oej@edvina.net>
* funcs/func_speex.c: Adding some clarifications to func_speex
doxygen docs. The functions needed doesn't exist in Speex 1.05
which is what a lot of distros use. 1.2 seems to have been in
beta status for years, and does include the sexy functions needed
for func_speex to work.
2009-11-03 15:37 +0000 [r227167] Joshua Colp <jcolp@digium.com>
* /: Merged revisions 227166 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r227166 | file | 2009-11-03 11:36:16 -0400 (Tue, 03 Nov 2009) | 5
lines Fix a bug where an RPID header could be generated with a
blank username in the URI. (closes issue #15909) Reported by:
kobaz ........
2009-11-03 15:19 +0000 [r227162] Leif Madsen <lmadsen@digium.com>
* configs/extensions.conf.sample: Update extensions.conf.sample
file to fix incorrect extensions. (closes issue #15857) Reported
by: pprindeville Patches: stdexten.patch#2 uploaded by
pprindeville (license 347) Tested by: pprindeville
2009-11-03 11:11 +0000 [r227091] Olle Johansson <oej@edvina.net>
* Makefile, /, channels/chan_sip.c: Merged revisions 227088 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r227088 | oej | 2009-11-03 11:29:59 +0100 (Tis, 03 Nov 2009) | 7
lines Use proper response code when violating Contact ACL's.
https://reviewboard.asterisk.org/r/415/ Thanks kpfleming for a
quick review. (EDVX-003) ........
2009-11-02 22:29 +0000 [r227049] Tilghman Lesher <tlesher@digium.com>
* configs/mgcp.conf.sample, include/asterisk/pktccops.h (added),
CHANGES, res/res_pktccops.c (added), channels/chan_mgcp.c,
configs/res_pktccops.conf.sample (added): Add PacketCable NCS 1.0
support for Docsis/Eurodocsis networks (closes issue #12950)
Reported by: alea-soluciones Patches:
ncs-pktccops-12950-r206803.patch uploaded by alea-soluciones
(license 514) Tested by: alea-soluciones, adomjan, urtho,
nahuelgreco
2009-11-02 20:59 +0000 [r226973-226974] David Brooks <dbrooks@digium.com>
* channels/chan_sip.c: SIP channel name uniqueness SIP channel
names were supposed to be unique by way of a name suffix derived
from the pointer to the channel's private data. Uniqueness was
preserved on 32-bit systems, but not on 64-bit systems. This
patch, as suggested by kpfleming, replaces this suffix with a
simple incremented unsigned int. (closes issue #15152) Reported
by: palbrecht Review: https://reviewboard.asterisk.org/r/420/
* /: SIP channel name uniqueness SIP channel names were supposed to
be unique by way of a name suffix derived from the pointer to the
channel's private data. Uniqueness was preserved on 32-bit
systems, but not on 64-bit systems. This patch, as suggested by
kpfleming, replaces this suffix with a simple incremented
unsigned int. (closes issue #15152) Reported by: palbrecht
Review: https://reviewboard.asterisk.org/r/420/
2009-11-02 20:43 +0000 [r226970] Olle Johansson <oej@edvina.net>
* main/http.c: Adding external reference for doxygen
2009-11-02 18:08 +0000 [r226890] Joshua Colp <jcolp@digium.com>
* apps/app_dial.c, /: Merged revisions 226889 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r226889 | file | 2009-11-02 14:08:11 -0400 (Mon, 02 Nov 2009) |
11 lines Fix a bug where the recorded privacy introduction file
would not get removed if the caller hung up while the called
party had not yet answered. This was fixed by introducing an
argument to the 'n' option which, when enabled, removes the
introduction file under all scenarios. This was done to preserve
the behavior that has existed for quite some time. (closes issue
#14674) Reported by: ulogic Patches: bug14674.patch uploaded by
jpeeler (license 325) ........
2009-11-02 17:34 +0000 [r226882] Richard Mudgett <rmudgett@digium.com>
* channels/sig_pri.h, channels/chan_dahdi.c, UPGRADE.txt,
channels/sig_pri.c: DAHDI ISDN channel names will not allow
device state to work. (Interim solution.) Since ISDN works like
SIP and not analog ports in regard to devices, the device state
based on the ISDN channel number could not work. This has not
been an issue until the advent of PTMP NT mode. Previously, ISDN
lines were used as trunks and did not have to keep track of
specific devices. As an interim solution until device states are
properly implemented, the channel name is being changed to the
following format to use the generic device state support:
DAHDI/i<span>/<number>[:<subaddress>]-<sequence-number> Dialplan
hints would thus be: exten => xxx,hint,DAHDI/i2/5551212 This will
work with the following restrictions: * The number of
devices/phones cannot exceed the number of B channels. (i.e., BRI
has 2) * Each device/phone can only have one number. No shared
MSN's. * The phones/devices probably should not use
subaddressing.
2009-11-02 17:15 +0000 [r226812] Tilghman Lesher <tlesher@digium.com>
* /, contrib/init.d/rc.redhat.asterisk: Merged revisions 226811 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r226811 | tilghman | 2009-11-02 11:14:20 -0600 (Mon, 02 Nov 2009)
| 8 lines Don't allow two separate instances of safe_asterisk
when restarting from the init script. (closes issue #14562)
Reported by: davidw Patches: Initially
20091022__issue14562.diff.txt uploaded by tilghman (license 14)
Modified to 20091030__Issue14562_diff.txt uploaded by davidw
(license 780) Tested by: davidw ........
2009-11-02 14:57 +0000 [r226687] Matthew Nicholson <mnicholson@digium.com>
* channels/chan_sip.c, configs/sip.conf.sample, CHANGES: This patch
adds support for a draft proposal for adding Q.850 reason headers
to sip messages. (closes issue #13385) Reported by: adomjan
Patches: sip.conf.sample-trunk20090929-reason_q850.patch uploaded
by adomjan (license 487) CHANGES-trunk20090929-reason_q850.patch
uploaded by adomjan (license 487)
chan_sip.c-trunk20090929-reason_q850_atoi_fix.patch uploaded by
adomjan (license 487) sip-q850-hangupcause1.diff uploaded by
mnicholson (license 96) Tested by: adomjan
2009-10-30 23:26 +0000 [r226648] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c, channels/sig_pri.c: Cleanup some flags on
DAHDI PRI channel hangup. * Cleanup some flags on DAHDI PRI
channel hangup. (sig_pri split) * Make sure the outgoing flag is
cleared if a new channel fails to get created for outgoing calls.
* Remove some unused flags since sig_pri was split.
2009-10-30 04:08 +0000 [r226606] Russell Bryant <russell@digium.com>
* include/asterisk/doxygen/architecture.h (added),
res/res_rtp_asterisk.c, res/res_rtp_multicast.c,
include/asterisk/doxyref.h, contrib/asterisk-ng-doxygen,
main/asterisk.c: Add an "Asterisk Architecture Overview" section
to the doxygen documentation. This is a side project I've been
poking at this week. The intent is to discuss Asterisk
architecture in a top down fashion to help new developers
understand how Asterisk is put together. There is a ton of stuff
to write about, so this will just continue to evolve over time.
2009-10-29 18:13 +0000 [r226532] Joshua Colp <jcolp@digium.com>
* channels/chan_local.c, /, doc/tex/localchannel.tex: Merged
revisions 226531 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r226531 | file | 2009-10-29 15:11:26 -0300 (Thu, 29 Oct 2009) | 6
lines Add an option to enabling passing music on hold start and
stop requests through instead of acting on them in chan_local.
(closes issue #14709) Reported by: dimas ........
2009-10-29 12:20 +0000 [r226490] Olle Johansson <oej@edvina.net>
* channels/chan_local.c: Doxygen documentation update
2009-10-28 20:50 +0000 [r226453] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
* build_tools/get_documentation: remove empty awk pattern (//)
Solaris 10 nawk doesn't lthe empty pattern ike '//' for 'always'.
Just remove that. No pattern at all always matches.
2009-10-28 20:11 +0000 [r226378-226384] Leif Madsen <lmadsen@digium.com>
* /, configs/sip.conf.sample: Merged revisions 226382 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r226382 | lmadsen | 2009-10-28 15:06:13 -0500 (Wed, 28
Oct 2009) | 9 lines Update documentation in sip.conf.sample.
Update the documentation in sip.conf.sample in order to make it
more clear that directmedia/canreinvite do not cause Asterisk to
ignore reINVITEs. It is only used to stop Asterisk from
generating a reINVITE, but does not stop it from accepting them
if necessary. (closes issue #15644) Reported by: lmadsen ........
* doc/tex/channelvariables.tex: Merged revisions 226377 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r226377 | lmadsen | 2009-10-28 14:48:29 -0500 (Wed, 28 Oct 2009)
| 7 lines Update CALLINGSUBADDR channel variable documentation.
(closes issue #15734) Reported by: alecdavis Patches:
channelvariables.tex.diff.txt uploaded by alecdavis (license 585)
Tested by: alecdavis ........
2009-10-28 18:04 +0000 [r226305] Tilghman Lesher <tlesher@digium.com>
* /, include/asterisk/linkedlists.h: Merged revisions 226304 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r226304 | tilghman | 2009-10-28 13:02:25 -0500 (Wed, 28 Oct 2009)
| 2 lines Fix documentation (pointed out by TheDavidFactor on
#-dev) ........
2009-10-28 08:47 +0000 [r226227-226270] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
* contrib/upstart/asterisk.user.conf: Remove extra cleanup in case
we have more than one Asterisk. /var/run would be cleaned on
startup on most systems anyway.
* contrib/upstart/asterisk.user.conf (added): another variation of
the upstart script
2009-10-27 21:03 +0000 [r226184] Olle Johansson <oej@edvina.net>
* Makefile: Adding compile time flags for Snow Leopard, Leopard and
some other animals
2009-10-27 20:22 +0000 [r226159] Tilghman Lesher <tlesher@digium.com>
* main/manager.c, /: Merged revisions 226138 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r226138 | tilghman | 2009-10-27 15:16:49 -0500 (Tue, 27 Oct 2009)
| 7 lines Manager output is not always NULL-terminated, so force
a NULL at the end of the filestream. (closes issue #15495)
Reported by: pdf Patches: 20090916__issue15495.diff.txt uploaded
by tilghman (license 14) Tested by: pdf ........
2009-10-27 16:48 +0000 [r226099] Terry Wilson <twilson@digium.com>
* res/res_http_post.c: Don't prepend the URI prefix to the post
directory
2009-10-27 13:30 +0000 [r226060] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c, configs/sip.conf.sample, CHANGES: Add
support for receiving unsolicited MWI NOTIFY messages. This
change adds a configuration option to SIP peers,
unsolicited_mailbox, which configures a virtual mailbox to use
for received new/old MWI information. This virtual mailbox can
then be used by any device supporting MWI. (closes issue #13028)
Reported by: AsteriskRocks Patches:
bug_13028_chan_sip_external_mwi_20090707.patch uploaded by cmaj
(license 830)
2009-10-26 22:46 +0000 [r226018] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
* /, configure, configure.ac: detect ARM Linux EABI OSARCH as
linux-gnu instead of linux-gnueabi * Set OSARCH to linux-gnu even
if host_os is linux-gnueabi * When checking if we are Linux,
check OSARCH rather than host_os The newer ARM ABI ("EABI") shows
the OS name 'linux-gnueabi' rather than 'linux-gnu' . This patch
sets OSARCH to be 'linux-gnu' even in such a case. OSARCH is
tested for the value of 'linux-gnu' in one or two places in the
tree. This patch also fixes the check libcap to check for $OSARCH
rather than $host_os . See also:
http://wiki.debian.org/ArmEabiPort Merged revisions 225957 via
svnmerge from http://svn.digium.com/svn/asterisk/branches/1.4
2009-10-26 22:04 +0000 [r225955-225956] Kevin P. Fleming <kpfleming@digium.com>
* main/editline/makelist.in, channels/chan_sip.c, UPGRADE.txt,
UPGRADE-1.6.txt, doc/lang/language-criteria.txt: Fix building in
REF_DEBUG mode.
* main/astobj2.c: Correct broken logic from revision 225405. The
code committed in revision 225405 was broken; instead of removing
the unreference code, the logic used to decide when to do it
should have been reversed. This patch corrects the situation, and
makes reference counting work properly again.
2009-10-26 19:40 +0000 [r225912] Jeff Peeler <jpeeler@digium.com>
* channels/chan_sip.c: ACL check not present for verifying SIP
INVITEs The ACL check in check_peer_ok was missing and has now
been restored. The missing check allowed for calls to be made on
prohibited networks where an ACL was defined in sip.conf and the
allowguest option was set to off. See the AST security advisory
below for more information. Merge code associated with
AST-2009-007. (closes issue #16091) Reported by: thom4fun
2009-10-26 16:07 +0000 [r225872] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c: Make conditionals create previous code
when libpri/ss7 are present.
2009-10-26 13:29 +0000 [r225767-225836] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
* channels/chan_dahdi.c: span numbers in pri debug / error messages
Prefix PRI trace messages with the span number. This makes the
trace readable even when you have a multi-port device. (closes
issue #15054) Reported by: tzafrir Patches:
dahdi_pri_debug_spannum.diff uploaded by tzafrir (license 46)
* channels/chan_dahdi.c: Re-arange code a bit to build in dev-mode
without ss7 No change of functionality here. Just localized a
variable and indented code into blocks.
* channels/chan_dahdi.c: Make chan_dahdi build even without PRI /
SS7 (Note: still some strange build warnings without SS7 in
dev-mode)
2009-10-24 14:40 +0000 [r225727] Kevin P. Fleming <kpfleming@digium.com>
* channels/chan_sip.c: Improve performance of pedantic mode dialog
searching in chan_sip. This patch changes chan_sip to use the new
astobj2 OBJ_MULTIPLE iterator support to make pedantic mode
dialog searching in find_call() not require a linear search of
all dialogs in the list of dialogs. This patch does *not* change
the dialog matching logic (more on that later), just improves the
searching performance.
2009-10-23 16:57 +0000 [r225692] Richard Mudgett <rmudgett@digium.com>
* channels/sig_pri.h, channels/chan_dahdi.c,
configs/chan_dahdi.conf.sample, configure,
include/asterisk/autoconfig.h.in, configure.ac, CHANGES,
channels/sig_pri.c: Add to chan_dahdi ISDN HOLD, Call deflection,
and keypad facility support. * Added handling of received
HOLD/RETRIEVE messages and the optional ability to transfer a
held call on disconnect similar to an analog phone. * Added
CallRerouting/CallDeflection support for Q.SIG, ETSI PTP, ETSI
PTMP. Will reroute/deflect an outgoing call when receive the
message. Can use the DAHDISendCallreroutingFacility to send the
message for the supported switches. * Added ability to
send/receive keypad digits in the SETUP message. Send keypad
digits in SETUP message:
Dial(DAHDI/g1[/K<keypad_digits>][/extension]) Access any received
keypad digits in SETUP message by: ${CHANNEL(keypad_digits)} *
Added support for BRI PTMP NT mode.
2009-10-23 16:40 +0000 [r225690] Sean Bright <sean@malleable.com>
* Makefile, agi/Makefile, agi/agi.xml (added): Optionally build and
install the sample AGIs in the agi/ directory.
2009-10-23 14:41 +0000 [r225650] David Vossel <dvossel@digium.com>
* channels/chan_sip.c: Fixes an iterator memory leak and
uninitialized memory
2009-10-23 14:02 +0000 [r225582] Kevin P. Fleming <kpfleming@digium.com>
* Makefile, /: Merged revisions 225581 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r225581 | kpfleming | 2009-10-23 09:00:01 -0500 (Fri, 23 Oct
2009) | 10 lines Don't force menuselect.makeopts to be rebuilt on
every build. For some reason the menuselect.makeopts file was
listed as PHONY in the Makefile, resulting in 'make' needing to
rebuild it for every build. This then resulted in the embedded
module rules being rebuilt on every build, which can be slow and
is unnecessary. This patch fixes the problem by properly allowing
'make' to know when the menuselect.makeopts file needs to be
rebuilt (defining the proper dependencies). ........
2009-10-22 22:24 +0000 [r225483-225515] Leif Madsen <lmadsen@digium.com>
* README: Update README documentation. Update the README
documentation to correctly describe which CLI command you should
use when attempting to get help from the CLI. (closes issue
#16064) Reported by: thedavidfactor Patches: readme.patch
uploaded by thedavidfactor (license 903)
* /, doc/valgrind.txt, contrib/valgrind.supp (added): Merged
revisions 225484 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r225484 | lmadsen | 2009-10-22 16:51:52 -0500 (Thu, 22 Oct 2009)
| 11 lines Clean valgrind output by suppressing false errors.
Update valgrind.txt documentation and add valgrind.supp file in
order to allow those who are creating valgrind output to have
less false errors in the logfile. (closes issue #16007) Reported
by: atis Patches: valgrind.txt.diff uploaded by atis (license
242) asterisk2.supp uploaded by atis (license 242) Tested by:
atis, amorsen ........
* include/asterisk/doxyref.h,
include/asterisk/doxygen/asterisk-git-howto.h (added): Add
Asterisk Git HowTo documentation. Added documentation on how to
create a local git repository from SVN. This documentation was
added via doxygen. (closes issue #15814) Reported by: tzafrir
Patches: git-asterisk-howto uploaded by tzafrir (license 46)
2009-10-22 20:07 +0000 [r225446] Richard Mudgett <rmudgett@digium.com>
* channels/sig_pri.c: Search for the subaddress only within the
extension section of the dial string.
Dial(DAHDI/(g|G|r|R)<group#(0-63)>[c|r<cadance#>|d][/extension])
2009-10-22 19:55 +0000 [r225445] David Vossel <dvossel@digium.com>
* main/tcptls.c, channels/chan_sip.c, apps/app_externalivr.c,
include/asterisk/tcptls.h: SIP TCP/TLS: move client connection
setup/write into tcp helper thread, various related
locking/memory fixes. What this patch fixes 1.Moves sip TCP/TLS
connection setup into the TCP helper thread: Connection setup
takes awhile and before this it was being done while holding the
monitor lock. 2.Moves TCP/TLS writing to the TCP helper thread:
Through the use of a packet queue and an alert pipe, the TCP
helper thread can now be woken up to write data as well as read
data. 3.Locking error: sip_xmit returned an XMIT_ERROR without
giving up the tcptls_session lock. This lock has been completely
removed from sip_xmit and placed in the new sip_tcptls_write()
function. 4.Memory leak: When creating a tcptls_client the
tls_cfg was alloced but never freed unless the tcptls_session
failed to start. Now the session_args for a sip client are an ao2
object which frees the tls_cfg on destruction. 5.Pointer to stack
variable: During sip_prepare_socket the creation of a client's
ast_tcptls_session_args was done on the stack and stored as a
pointer in the newly created tcptls_session. Depending on the
events that followed, there was a slight possibility that pointer
could have been accessed after the stack returned. Given the new
changes, it is always accessed after the stack returns which is
why I found it. Notable code changes 1.I broke tcptls.c's
ast_tcptls_client_start() function into two functions. One for
creating and allocating the new tcptls_session, and a separate
one for starting and handling the new connection. This allowed me
to create the tcptls_session, launch the helper thread, and then
establish the connection within the helper thread. 2.Writes to a
tcptls_session are now done within the helper thread. This is
done by using an alert pipe to wake up the thread if new data
needs to be sent. The thread's sip_threadinfo object contains the
alert pipe as well as the packet queue. 3.Since the threadinfo
object contains the alert pipe, it must now be accessed outside
of the helper thread for every write (queuing of a packet). For
easy lookup, I moved the threadinfo objects from a linked list to
an ao2_container. (closes issue #13136) Reported by: pabelanger
Tested by: dvossel, whys (closes issue #15894) Reported by:
dvossel Tested by: dvossel Review:
https://reviewboard.asterisk.org/r/380/
2009-10-22 19:33 +0000 [r225440] Sean Bright <sean@malleable.com>
* Makefile, utils/Makefile, utils/utils.xml (added),
doc/janitor-projects.txt: Add the programs in utils/ to
menuselect. Nothing in utils/ is now built by default except for
astcanary. Review: https://reviewboard.asterisk.org/r/353/
2009-10-22 19:10 +0000 [r225406] Tilghman Lesher <tlesher@digium.com>
* configs/voicemail.conf.sample, CHANGES, apps/app_voicemail.c:
Permit storage of voicemail secrets in a separate file, located
within the spool directory. (closes issue #14276) Reported by:
klaus3000 Patches: app_voicemail.c-svn-trunk-r214898.txt uploaded
by klaus3000 (license 65) Tested by: jamesgolovich
2009-10-22 18:41 +0000 [r225405] Kevin P. Fleming <kpfleming@digium.com>
* main/astobj2.c: Fix a refcount error introduced by yesterday's
OBJ_MULTIPLE commit. When an object is being unlinked from its
container *and* being returned to the caller, we do not want to
decrement the reference count after unlinking it from the
container, as the reference that the container held is what we
are returning to the caller... and if it was the only remaining
reference to the object, that could result in the object being
destroyed.
2009-10-22 17:11 +0000 [r225360] Tilghman Lesher <tlesher@digium.com>
* main/pbx.c, /, apps/app_meetme.c, include/asterisk/channel.h:
Merged revisions 225105 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r225105 | tilghman | 2009-10-21 11:02:12 -0500 (Wed, 21 Oct 2009)
| 4 lines Fix documentation for ast_softhangup() and correct the
misuse thereof. (closes issue #16103) Reported by: majorbloodnok
........
2009-10-22 16:33 +0000 [r225357] Richard Mudgett <rmudgett@digium.com>
* main/channel.c, configure, include/asterisk/autoconfig.h.in,
configure.ac, funcs/func_connectedline.c,
include/asterisk/channel.h, CHANGES, channels/sig_pri.c,
funcs/func_callerid.c: Add support for calling and called
subaddress. Partial support for COLP subaddress. The Telecom
Specs in NZ suggests that SUB ADDRESS is always on, so doing
"desk to desk" between offices each with an asterisk box over the
ISDN should then be possible, without a whole load of DDI numbers
required. (closes issue #15604) Reported by: alecdavis Patches:
asterisk_subaddr_trunk.diff11.txt uploaded by alecdavis (license
585) Some minor modificatons were made. Tested by: alecdavis,
rmudgett Review: https://reviewboard.asterisk.org/r/405/
2009-10-21 21:58 +0000 [r225307] David Vossel <dvossel@digium.com>
* /, channels/chan_iax2.c: Merged revisions 225243 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r225243 | dvossel | 2009-10-21 15:58:08 -0500 (Wed, 21
Oct 2009) | 13 lines IAX2: VNAK loop caused by signaling frames
with no destination call number It is possible for the PBX thread
to queue up signaling frames before a destination call number is
received. This can result in signaling frames being sent out with
no destination call number. Since recent versions of Asterisk
require accurate destination callnumbers for all Full Frames,
this can cause a VNAK loop to occur. To resolve this no signaling
frames are sent until a destination callnumber is received, and
destination call numbers are now only required for iax_pvt
matching when the frame is an ACK. Review:
https://reviewboard.asterisk.org/r/413/ ........
2009-10-21 21:15 +0000 [r225244-225245] Kevin P. Fleming <kpfleming@digium.com>
* doc/tex/manager.tex, channels/chan_sip.c: Add 'mohsuggest'
configuration option to 'sip show peer' CLI command and
SIPShowPeer AMI action. (closes issue #15990) Reported by:
_brent_ Patches: sip_peer_info_mohsuggest-r3.patch uploaded by
brent (license 388) Review:
https://reviewboard.asterisk.org/r/381/
* main/channel.c, main/manager.c, apps/app_directed_pickup.c,
apps/app_softhangup.c, funcs/func_channel.c,
include/asterisk/astobj2.h, res/snmp/agent.c,
include/asterisk/channel.h, include/asterisk/lock.h,
apps/app_chanspy.c, main/astobj2.c, main/cli.c: Finish
implementaton of astobj2 OBJ_MULTIPLE, and convert
ast_channel_iterator to use it. This patch finishes the
implementation of OBJ_MULTIPLE in astobj2 (the case where
multiple results need to be returned; OBJ_NODATA mode already was
supported). In addition, it converts ast_channel_iterators (only
the targeted versions, not the ones that iterate over all
channels) to use this method. During this work, I removed the
'ao2_flags' arguments to the ast_channel_iterator constructor
functions; there were no uses of that argument yet, there is only
one possible flag to pass, and it made the iterators less
'opaque'. If at some point in the future someone really needs an
ast_channel_iterator that does not lock the container, we can
provide constructor(s) for that purpose. Review:
https://reviewboard.asterisk.org/r/379/
2009-10-21 16:46 +0000 [r225170-225172] Russell Bryant <russell@digium.com>
* /, main/translate.c: Merged revisions 225171 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r225171 | russell | 2009-10-21 11:44:49 -0500 (Wed, 21 Oct 2009)
| 2 lines Revert 225169, as this doesn't account for the
possibility of a list of frames. ........
* /, main/translate.c: Merged revisions 225169 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r225169 | russell | 2009-10-21 11:39:20 -0500 (Wed, 21 Oct 2009)
| 2 lines Isolate the frame returned from ast_translate().
........
2009-10-21 15:42 +0000 [r225102] Tilghman Lesher <tlesher@digium.com>
* apps/app_meetme.c: Apparently, I don't need to specify the ".so"
suffix to get a match
2009-10-21 15:35 +0000 [r225089] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c, configs/sip.conf.sample, CHANGES: Add
support for specifying the IP address to use for media streams in
sip.conf This is the second commit for this and documents the
text stream using the configured IP address and fixes a bug in
the original patch where the UDPTL stream would also use the
different IP address. (closes issue #14729) Reported by: _brent_
Patches: media_address.patch uploaded by brent (license 388)
2009-10-21 15:21 +0000 [r225048] Tilghman Lesher <tlesher@digium.com>
* apps/app_meetme.c, CHANGES: Turn on DENOISE filter for all
conference participants. (Fixes SWP-238)
2009-10-21 15:04 +0000 [r225034] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c, configs/sip.conf.sample, CHANGES: Revert
media_address commit, I'm going to roll a fix to the SDP
generation in the next version.
2009-10-21 14:39 +0000 [r225033] David Vossel <dvossel@digium.com>
* configs/iax.conf.sample, /, channels/chan_sip.c,
configs/sip.conf.sample, channels/chan_iax2.c: Merged revisions
225032 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r225032 | dvossel | 2009-10-21 09:37:04 -0500 (Wed, 21 Oct 2009)
| 20 lines IAX/SIP shrinkcallerid option The shrinking of caller
id removes '(', ' ', ')', non-trailing '.', and '-' from the
string. This means values such as 555.5555 and test-test result
in 555555 and testtest. There are instances, such as Skype
integration, where a specific value is passed via caller id that
must be preserved unmodified. This patch makes the shrinking of
caller id optional in chan_sip and chan_iax in order to support
such cases. By default this option is on to preserve previous
expected behavior. (closes issue #15940) Reported by: dimas
Patches: v2-15940.patch uploaded by dimas (license 88)
15940_shrinkcallerid_trunk.c uploaded by dvossel (license 671)
Tested by: dvossel Review:
https://reviewboard.asterisk.org/r/408/ ........
2009-10-21 13:34 +0000 [r225003] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c, configs/sip.conf.sample, CHANGES: Add
support for specifying the IP address to use for media streams in
sip.conf (closes issue #14729) Reported by: _brent_ Patches:
media_address.patch uploaded by brent (license 388)
2009-10-21 03:09 +0000 [r224932] Russell Bryant <russell@digium.com>
* main/frame.c, /, main/translate.c, include/asterisk/dsp.h,
codecs/codec_dahdi.c, include/asterisk/frame.h,
include/asterisk/translate.h, main/dsp.c: Merged revisions 224931
via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r224931 | russell | 2009-10-20 21:59:54 -0500 (Tue, 20 Oct 2009)
| 5 lines Isolate frames returned from a DSP instance or codec
translator. The reasoning for these changes are the same as what
I wrote in the commit message for rev 222878. ........
2009-10-21 02:43 +0000 [r224930] Richard Mudgett <rmudgett@digium.com>
* channels/sig_pri.c: Make PRI_SUBCMD_xxx handling subaddress
friendly.
2009-10-20 22:09 +0000 [r224856] Tilghman Lesher <tlesher@digium.com>
* funcs/func_speex.c, /, main/audiohook.c: Merged revisions 224855
via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r224855 | tilghman | 2009-10-20 17:07:11 -0500 (Tue, 20 Oct 2009)
| 5 lines Pay attention to the return value of the manipulate
function. While this looks like an optimization, it prevents a
crash from occurring when used with certain audiohook callbacks
(diagnosed with SVN trunk, backported to 1.4 to keep the source
consistent across versions). ........
2009-10-20 17:47 +0000 [r224774] Joshua Colp <jcolp@digium.com>
* /, main/features.c: Merged revisions 224773 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r224773 | file | 2009-10-20 14:46:37 -0300 (Tue, 20 Oct 2009) | 5
lines Add support for relaying early media in the features
attended transfer option. (closes issue #14828) Reported by:
licedey ........
2009-10-20 12:44 +0000 [r224738] Matthew Nicholson <mnicholson@digium.com>
* CHANGES: Added information to CHANGES about the dynamic range
compression feature added to dahdi.
2009-10-19 23:47 +0000 [r224671] Kevin P. Fleming <kpfleming@digium.com>
* res/res_rtp_asterisk.c, /: Merged revisions 224670 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r224670 | kpfleming | 2009-10-19 18:44:07 -0500 (Mon, 19
Oct 2009) | 7 lines Correct timestamp calculations when RTP
sample rates over 8kHz are used. While testing some endpoints
that support 16kHz and 32kHz sample rates, some log messages were
generated due to calc_rxstamp() computing timestamps in a way
that produced odd results, so this patch sanitizes the result of
the computations. ........
2009-10-19 22:02 +0000 [r224637] Matthew Nicholson <mnicholson@digium.com>
* channels/chan_dahdi.c, configs/chan_dahdi.conf.sample: Add
dynamic range compression support for analog channels. (closes
issue AST-29)
2009-10-19 19:49 +0000 [r224567] Joshua Colp <jcolp@digium.com>
* apps/app_dial.c, /: Merged revisions 224565 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r224565 | file | 2009-10-19 16:47:50 -0300 (Mon, 19 Oct 2009) | 5
lines Do not attempt early media bridging (ie: direct RTP setup)
if options are enabled that should prevent it. (closes issue
#14763) Reported by: cupotka ........
2009-10-19 19:40 +0000 [r224562] Kevin P. Fleming <kpfleming@digium.com>
* formats/format_siren14.c: Remove useless debugging message.
2009-10-19 15:50 +0000 [r224527] Tilghman Lesher <tlesher@digium.com>
* doc/janitor-projects.txt: Remove a completed project and add
another
2009-10-19 14:32 +0000 [r224491] Joshua Colp <jcolp@digium.com>
* channels/sig_pri.h, channels/sig_pri.c: Add a callback to sig_pri
which is called when sig_pri is going to queue a control frame on
a channel.
2009-10-19 00:05 +0000 [r224446-224448] Tilghman Lesher <tlesher@digium.com>
* apps/app_voicemail.c: Allow ODBC storage to be queried with
multiple mailboxes, and remove multiple goto's. This corrects an
issue reported on the -users list.
* configs/res_odbc.conf.sample: Clarify that "forcecommit" is NOT
an alias for "autocommit", but instead controls the default
disposition of uncommitted transactions.
2009-10-17 16:39 +0000 [r224403] Tilghman Lesher <tlesher@digium.com>
* include/asterisk/app.h, main/app.c: Remove unnecessary typedef
2009-10-17 02:01 +0000 [r224331-224335] Jeff Peeler <jpeeler@digium.com>
* channels/chan_dahdi.c: fix typo, sorry
* channels/chan_dahdi.c, /, channels/sig_pri.c: Merged revisions
224330 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r224330 | jpeeler | 2009-10-16 20:32:47 -0500 (Fri, 16 Oct 2009)
| 13 lines Fix stale caller id data from being reported in AMI
NewChannel event The problem here is that chan_dahdi is designed
in such a way to set certain values in the dahdi_pvt only once.
One of those such values is the configured caller id data in
chan_dahdi.conf. For PRI, the configured caller id data could be
overwritten during a call. Instead of saving the data and
restoring, it was decided that for all non-analog channels it was
simply best to not set the configured caller id in the first
place and also clear it at the end of the call. (closes issue
#15883) Reported by: jsmith ........
2009-10-16 20:40 +0000 [r224261] Richard Mudgett <rmudgett@digium.com>
* /, channels/sig_pri.c: Merged revisions 224260 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r224260 | rmudgett | 2009-10-16 15:25:23 -0500 (Fri, 16 Oct 2009)
| 18 lines Never released PRI channels when using Busy() or
Congestion() dialplan apps. When the Busy() or Congestion()
application is used towards ISDN (an ISDN progress is sent), the
responding ISDN Disconnect or Release may contain the ISDN cause
user busy or one of the congestion causes. In chan_dahdi.c these
causes will only set the needbusy or needcongestion flags and not
activate the softhangup procedure. Unfortunately only the latter
can interrupt the endless wait loop of Busy()/Congestion().
Result: PRI channels staying in state busy for the rest of
asterisk life or until the other end times out and forces the
call to clear. (issue #14292) Reported by: tomaso Patches:
disc_rel_userbusy.patch uploaded by tomaso (license 564) (This
patch is unrelated to the issue.) ........
2009-10-15 22:33 +0000 [r224225] Tilghman Lesher <tlesher@digium.com>
* include/asterisk/app.h, main/pbx.c, main/app.c: Create an API for
adding an optional time unit onto the ends of time periods. Two
examples of its use are included, and the usage could be expanded
in some cases into certain configuration options where time
periods are specified.
2009-10-15 15:57 +0000 [r224178] Jeff Peeler <jpeeler@digium.com>
* apps/app_chanspy.c: Readd removed ability to allow listening to
one side of the call in app_chanspy (Option o) (closes issue
#15675) Reported by: john8675309 Patches:
issue15675patchtrunk.txt uploaded by dbrooks (license 790) Tested
by: jgutierrez on users list:
http://lists.digium.com/pipermail/asterisk-users/2009-October/239155.html
2009-10-15 14:37 +0000 [r224144] Doug Bailey <dbailey@digium.com>
* configs/chan_dahdi.conf.sample: chan_dahdi.conf.sample changes
for DTMF CID detect Explains new options for detecting DTMF CID
on fxo lines (issue #9096) Reported by: fleed Patches:
chan_dahid_sample_config.patch uploaded by sum (license 766)
2009-10-15 06:48 +0000 [r224074-224109] Terry Wilson <twilson@digium.com>
* res/res_calendar_caldav.c: Properly handle PUT requests for
CALENDAR_WRITE()
* res/res_calendar.c: Add missing 'getnum' field
2009-10-14 17:48 +0000 [r224035] Jeff Peeler <jpeeler@digium.com>
* configs/sip_notify.conf.sample, channels/chan_sip.c, CHANGES:
Allow for adding message body to the SIP NOTIFY message Ability
has been added to both manager command SIPnotify as well as
console command sip notify. Message body is stored in the
"Content" variable. An example is present in sip_notify.conf.
(closes issue #13926) Reported by: jthurman Patches:
sip-notify-svn189463.diff uploaded by gareth (license 208) Tested
by: gareth
2009-10-13 22:14 +0000 [r223992] Terry Wilson <twilson@digium.com>
* res/res_calendar.c: use Calendar: instead of Calendar/ for
devstate
2009-10-13 17:11 +0000 [r223911-223912] Richard Mudgett <rmudgett@digium.com>
* include/asterisk/pbx.h: Fix some doxygen format problems and trim
trailing whitespace.
* res/res_calendar.c: Fix compiler warning.
2009-10-13 01:58 +0000 [r223874-223875] Terry Wilson <twilson@digium.com>
* apps/app_originate.c: Revert inadvertant code commit to
app_originate
* apps/app_originate.c, include/asterisk/calendar.h,
res/res_calendar.c: Fix handling of notification calls w/ the
dialing api
2009-10-12 23:48 +0000 [r223832] Jeff Peeler <jpeeler@digium.com>
* apps/app_dial.c, /: Merged revisions 223804 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r223804 | jpeeler | 2009-10-12 18:12:50 -0500 (Mon, 12 Oct 2009)
| 8 lines Ensure ringing continues for branched calls after
progress is received While waiting for an answer, don't send
progress for branched calls for which ringing was sent. (closes
issue #15028) Reported by: fnordian ........
2009-10-12 20:58 +0000 [r223756] David Vossel <dvossel@digium.com>
* configs/iax.conf.sample: Clarifies trunkmaxsize, trunkfreq, and
trunkmtu iax2 options SWP-151
2009-10-12 15:32 +0000 [r223652-223693] Kevin P. Fleming <kpfleming@digium.com>
* /: Recorded merge of revisions 223692 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r223692 | kpfleming | 2009-10-12 10:30:40 -0500 (Mon, 12 Oct
2009) | 13 lines Remove automatic switching from T.38 to voice
mode in chan_sip. chan_sip has some code to automatically switch
from T.38 mode to voice mode when a voice frame is written to the
channel while it is in T.38 mode; this was intended to handle the
situation when a FAX transmission has ended and the channel is
not yet hung up, but is causing problems at the beginning of FAX
sessions as well when there are still voice frames 'in flight' at
the time the T.38 negotiation completes. This patch removes the
automatic switchover. (issue #16025) Reported by: jamicque
........
* channels/chan_sip.c, apps/app_fax.c: Remove automatic switching
from T.38 to voice mode in chan_sip. chan_sip has some code to
automatically switch from T.38 mode to voice mode when a voice
frame is written to the channel while it is in T.38 mode; this
was intended to handle the situation when a FAX transmission has
ended and the channel is not yet hung up, but is causing problems
at the beginning of FAX sessions as well when there are still
voice frames 'in flight' at the time the T.38 negotiation
completes. This patch removes the automatic switchover, and
changes app_fax to explicitly switch off T.38 mode when the FAX
transmission process ends. (closes issue #16025) Reported by:
jamicque
2009-10-11 22:19 +0000 [r223617] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Check the proper page for the SENDRPID flag.
If a pending reinvite were sent, we might not properly send
connected party info since we were checking the wrong flag. This
was a rare occurrence, but could still happen nevertheless.
2009-10-11 18:35 +0000 [r223487-223553] Russell Bryant <russell@digium.com>
* /: Merged revisions 223550 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r223550 | russell | 2009-10-11 13:34:37 -0500 (Sun, 11 Oct 2009)
| 2 lines Remove a duplicate ao2_iterator_destroy(). ........
* main/autoservice.c, /: Merged revisions 223485-223486 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r223485 | russell | 2009-10-11 12:22:52 -0500 (Sun, 11 Oct 2009)
| 6 lines Don't use data outside of its scope. The purpose of
this code was to have a hangup frame put on the list of deferred
frames. However, the code that read the hangup frame was outside
of the scope of where the hangup frame was declared. ........
r223486 | russell | 2009-10-11 12:25:06 -0500 (Sun, 11 Oct 2009)
| 2 lines Remove some unnecessary code. ........
2009-10-10 20:02 +0000 [r223449] Terry Wilson <twilson@digium.com>
* res/res_calendar_icalendar.c, res/res_calendar_caldav.c: Fix
handling of floating times and dates
2009-10-10 08:30 +0000 [r223413-223415] Olle Johansson <oej@edvina.net>
* configs/cdr_pgsql.conf.sample: Adding note about TLS usage
* configs/res_ldap.conf.sample: Add an additional note on TLS
support
* configs/res_ldap.conf.sample: Adding some information on TLS
support
2009-10-09 22:04 +0000 [r223370] Terry Wilson <twilson@digium.com>
* res/res_calendar_icalendar.c, res/res_calendar_caldav.c: Properly
return "free" on confirmed events that are free CONFIRMED status
doesn't imply busy or free, that is handled with the TRANSP
field. Luckily, libical already sets the is_busy status on the
span for us.
2009-10-09 20:58 +0000 [r223330] Kevin P. Fleming <kpfleming@digium.com>
* apps/app_fax.c: Initiate T.38 switchover when acting as called
party, regardless of FAX direction. SendFAX() and ReceiveFAX()
can be given options to indicate whether they should act as the
calling or called party; this mode should be used to decide
whether to initiate a switchover to T.38, not the direction that
the FAX transfer will take place. (closes issue #16039) Reported
by: jamicque
2009-10-09 18:34 +0000 [r223273] Matthew Nicholson <mnicholson@digium.com>
* main/channel.c, /: Merged revisions 223225 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r223225 | mnicholson | 2009-10-09 13:20:11 -0500 (Fri, 09 Oct
2009) | 8 lines Signal timeouts by returning AST_CONTROL_RINGING
when originating calls. (closes issue #15104) Reported by:
nblasgen Patches: manager-timeout1.diff uploaded by mnicholson
(license 96) Tested by: nblasgen, mnicholson ........
2009-10-09 18:17 +0000 [r223211-223215] Mark Michelson <mmichelson@digium.com>
* /: Recorded merge of revisions 223213 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r223213 | mmichelson | 2009-10-09 13:17:12 -0500 (Fri, 09 Oct
2009) | 3 lines Fix potential memory leak in app_dial.c ........
* apps/app_dial.c: Fix potential memory leaks. ABE-1998
2009-10-09 17:53 +0000 [r223206] David Vossel <dvossel@digium.com>
* /, channels/chan_sip.c: Merged revisions 223205 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r223205 | dvossel | 2009-10-09 12:52:35 -0500 (Fri, 09 Oct 2009)
| 10 lines fixes sip registration using authuser in user.conf
(closes issue #14954) Reported by: tornblad Tested by:
mmichelson, tornblad, dvossel ........
2009-10-09 17:14 +0000 [r223136] Matthew Nicholson <mnicholson@digium.com>
* cdr/cdr_sqlite3_custom.c: Don't close the sqlite database when
reloading. Only close the database when unloading. (closes issue
#15953) Reported by: frawd Patches: sqlite3_rev220097.diff
uploaded by frawd (license 610) Tested by: frawd
2009-10-09 16:54 +0000 [r223088-223132] David Vossel <dvossel@digium.com>
* channels/chan_sip.c: 'auth=' did not parse md5 secret correctly
(closes issue #15949) Reported by: ebroad Patches:
authparsefix.patch uploaded by ebroad (license 878)
15949_trunk.diff uploaded by dvossel (license 671) Tested by:
ebroad
* channels/chan_sip.c: p->peerauth is always empty in
transmit_register() When using callbackextension or specifing the
peer name in a registration string, the peer's specific auth
settings set by the "auth=" strings within the peer definition
are not used by the registration. Thanks to ebroad for reporting
the issue and providing the patch. (closes issue #15955) Reported
by: ebroad Patches: regauthfix.patch uploaded by ebroad (license
878)
2009-10-09 15:00 +0000 [r223016-223053] Terry Wilson <twilson@digium.com>
* res/res_calendar.c: Don't add Attendees during copy, replace them
* res/res_calendar_exchange.c, res/res_calendar_icalendar.c,
res/res_calendar_caldav.c, include/asterisk/calendar.h,
res/res_calendar.c: Remove global variable that makes dlopen
unhappy This isn't the best way to do this, but it is the
easiest. There are some limitations that are going to need to be
addressed at some point with reloads and when I (or someone else)
work on that, then the API can be updated to handle passing the
private config data that the calendar tech modules need in a
better way as well.
2009-10-08 22:57 +0000 [r222947-223015] David Vossel <dvossel@digium.com>
* channels/chan_sip.c: fixed comment line for do_magic_pickup
* channels/chan_sip.c: Deadlock between ast_cel_report_event and
ast_do_masquerade chan_sip calls pbx_exec on a pvt's owner
channel while only the pvt lock is held. Since pbx_exec calls
ast_cel_report_event which attempts to lock the channel, invalid
locking order occurs. Channels should be locked before pvt's.
(closes issue #15512) Reported by: lmsteffan Patches:
ast_cel_deadlock_15512.diff uploaded by dvossel (license 671)
* channels/chan_sip.c: makes externtcpport and externtlsport static
variables externtcpport and externtlsport need to be declared as
static variables. Thanks to russell for finding and pointing this
out.
2009-10-08 19:52 +0000 [r222880] Russell Bryant <russell@digium.com>
* include/asterisk/file.h, main/frame.c, /, main/file.c,
include/asterisk/frame.h: Merged revisions 222878 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r222878 | russell | 2009-10-08 14:45:47 -0500 (Thu, 08
Oct 2009) | 44 lines Make filestream frame handling safer by
isolating frames before returning them. This patch is related to
a number of issues on the bug tracker that show crashes related
to freeing frames that came from a filestream. A number of fixes
have been made over time while trying to figure out these
problems, but there re still people seeing the crash. (Note that
some of these bug reports include information about other
problems. I am specifically addressing the filestream frame crash
here.) I'm still not clear on what the exact problem is. However,
what is _very_ clear is that we have seen quite a few problems
over time related to unexpected behavior when we try to use
embedded frames as an optimization. In some cases, this
optimization doesn't really provide much due to improvements made
in other areas. In this case, the patch modifies filestream
handling such that the embedded frame will not be returned.
ast_frisolate() is used to ensure that we end up with a
completely mallocd frame. In reality, though, we will not
actually have to malloc every time. For filestreams, the frame
will almost always be allocated and freed in the same thread.
That means that the thread local frame cache will be used. So,
going this route doesn't hurt. With this patch in place, some
people have reported success in not seeing the crash anymore.
(SWP-150) (AST-208) (ABE-1834) (issue #15609) Reported by: aragon
Patches: filestream_frisolate-1.4.diff2.txt uploaded by russell
(license 2) Tested by: aragon, russell (closes issue #15817)
Reported by: zerohalo Tested by: zerohalo (closes issue #15845)
Reported by: marhbere Review:
https://reviewboard.asterisk.org/r/386/ ........
2009-10-08 19:35 +0000 [r222873] David Vossel <dvossel@digium.com>
* include/asterisk/netsock.h, main/netsock.c: fixes an
ast_netsock_list memory leak. ABE-1998 Review:
https://reviewboard.asterisk.org/r/395/
2009-10-08 16:44 +0000 [r222799] Richard Mudgett <rmudgett@digium.com>
* /, channels/misdn_config.c: Merged revisions 222797 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r222797 | rmudgett | 2009-10-08 11:33:06 -0500 (Thu, 08
Oct 2009) | 12 lines Fix memory leak if chan_misdn config
parameter is repeated. Memory leak when the same config option is
set more than once in an misdn.conf section. Why must this be
considered? Templates! Defining a template with default port
options and later adding to or overriding some of them. Patches:
memleak-misdn.patch JIRA ABE-1998 ........
2009-10-07 22:58 +0000 [r222761] David Vossel <dvossel@digium.com>
* main/channel.c, main/pbx.c, channels/chan_misdn.c,
channels/chan_sip.c, main/features.c, include/asterisk/channel.h:
Deadlock in channel masquerade handling Channels are stored in an
ao2_container. When accessing an item within an ao2_container the
proper locking order is to first lock the container, and then the
items within it. In ast_do_masquerade both the clone and original
channel must be locked for the entire duration of the function.
The problem with this is that it attemptes to unlink and link
these channels back into the ao2_container when one of the
channel's name changes. This is invalid locking order as the
process of unlinking and linking will lock the ao2_container
while the channels are locked!!! Now, both the channels in
do_masquerade are unlinked from the ao2_container and then locked
for the entire function. At the end of the function both channels
are unlocked and linked back into the container with their new
names as hash values. This new method of requiring all channels
and tech pvts to be unlocked before ast_do_masquerade() or
ast_change_name() required several changes throughout the code
base. (closes issue #15911) Reported by: russell Patches:
masq_deadlock_trunk.diff uploaded by dvossel (license 671) Tested
by: dvossel, atis (closes issue #15618) Reported by: lmsteffan
Patches: deadlock_local_attended_transfers_trunk.diff uploaded by
dvossel (license 671) Tested by: lmsteffan, dvossel Review:
https://reviewboard.asterisk.org/r/387/
2009-10-07 21:56 +0000 [r222692] Richard Mudgett <rmudgett@digium.com>
* channels/chan_misdn.c, /: Merged revisions 222691 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r222691 | rmudgett | 2009-10-07 16:51:24 -0500 (Wed, 07
Oct 2009) | 14 lines chan_misdn.c:process_ast_dsp() memory leak
misdn.conf: astdtmf must be set to "yes". With "no", buffer loss
does not occur. The translated frame "f2" when passing through
ast_dsp_process() is not freed whenever it is not used further in
process_ast_dsp(). Then in the end it is never ever freed.
Patches: translate.patch JIRA ABE-1993 ........
2009-10-07 20:08 +0000 [r222652] Jeff Peeler <jpeeler@digium.com>
* channels/chan_dahdi.c: Change ringt (ring timeout) styles to be
consistent across chan_dahdi. (closes issue #15684) Reported by:
alecdavis Patches: chan_dahdi.bug15684.diff2.txt uploaded by
alecdavis (license 585) Tested by: alecdavis
2009-10-07 18:57 +0000 [r222614-222615] Olle Johansson <oej@edvina.net>
* res/res_config_ldap.c: Formatting, moving error messages to
ERROR, removing references to unexisting debug output. No
functionality changes.
* cel/cel_pgsql.c, res/res_config_pgsql.c, cdr/cdr_pgsql.c: Use
extref for doxygen references to external libraries (in this case
PostgreSQL)
2009-10-07 18:04 +0000 [r222548] Jason Parker <jparker@digium.com>
* configs/queues.conf.sample: Remove 'keepstats' queue option from
sample config, as it's no longer used.
https://reviewboard.asterisk.org/r/115/ (closes issue #15820)
Reported by: kshumard
2009-10-07 17:44 +0000 [r222543] David Vossel <dvossel@digium.com>
* /, channels/chan_sip.c: Merged revisions 222542 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r222542 | dvossel | 2009-10-07 12:41:21 -0500 (Wed, 07 Oct 2009)
| 8 lines crash on transfer handle_invite_replaces() attempts to
uplock a pvt's owner channel without first verifing that it
exists. (issue #16027) ........
2009-10-06 23:56 +0000 [r222463] Jeff Peeler <jpeeler@digium.com>
* channels/chan_dahdi.c, /: Merged revisions 222462 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r222462 | jpeeler | 2009-10-06 18:51:19 -0500 (Tue, 06
Oct 2009) | 8 lines Add missing unlock(s) in dahdi_read (two
cases in trunk) (closes issue #15683) Reported by: alecdavis
........
2009-10-06 22:49 +0000 [r222398-222399] David Vossel <dvossel@digium.com>
* CHANGES: Updates CHANGES to reflect the new externtcpport and
externtlsport sip options
* channels/chan_sip.c, configs/sip.conf.sample: contact header port
ignored transport when using externip This patch adds support for
TCP/TLS in the Contact header when using NAT, specifically
externip or externhost. The original issue was that Asterisk sent
5060 as the port in the contact header whether TLS was used or
not. Additionally, this patch adds 2 config options to sip.conf,
specifically externtcpport and externtlsport. This allows a user
to specify different external ports for TCP and TLS other than
those used internally, this is especially useful in in a PAT/port
redirection setup. Thanks to ebroad for reporting the issue and
providing the patch! (closes issue #15880) Reported by: ebroad
Patches: portmap.patch uploaded by ebroad (license 878)
externtXXport_v2.patch uploaded by ebroad (license 878) Tested
by: ebroad Review: https://reviewboard.asterisk.org/r/392/
2009-10-06 20:35 +0000 [r222351] Jeff Peeler <jpeeler@digium.com>
* channels/chan_dahdi.c: Fix 222298 (crash during destruction of
second channel when variable set with setvar). I mistakenly
reasoned that setvar would be used on all channels. Since it can
be set per channel, give each dahdi channel a copy of the
variable. (related to #15899)
2009-10-06 19:31 +0000 [r222309] Tilghman Lesher <tlesher@digium.com>
* res/res_config_pgsql.c, cdr/cdr_pgsql.c: Change schema query to
involve the use of an optional schema parameter. This change is
done in such a way as to allow the driver to continue to function
with older databases which don't have these features. (closes
issue #16000) Reported by: jamicque Patches:
20091002__issue16000.diff.txt uploaded by tilghman (license 14)
20091002__issue16000__1.6.1.diff.txt uploaded by tilghman
(license 14) Tested by: jamicque
2009-10-06 19:24 +0000 [r222298] Jeff Peeler <jpeeler@digium.com>
* channels/chan_dahdi.c: Fix crash during destruction of second
channel when variable set with setvar. The setvar line in
chan_dahdi.conf is shared among all the channels, so make sure to
only free the resources only when the last channel is destroyed.
(closes issue #15899) Reported by: tzafrir
2009-10-06 19:17 +0000 [r222273] Tilghman Lesher <tlesher@digium.com>
* res/ael/pval.c: When we call a gosub routine, the variables
should be scoped to avoid contaminating the caller. This affected
the ~~EXTEN~~ hack, where a subroutine might have changed the
value before it was used in the caller. Patch by myself, tested
by ebroad on #asterisk
2009-10-06 16:17 +0000 [r222237] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
* channels/chan_dahdi.c: Make sure digit events are not reported as
"ERROR" dahdievent_to_analogevent used a simple switch statement
to convert DAHDI event numbers to "ANALOG_*" event numbers.
However "digit" events (DAHDI_EVENT_PULSEDIGIT,
DAHDI_EVENT_DTMFDOWN, DAHDI_EVENT_DTMFUP) are accompannied by the
digit in the low word of the event number. This fix makes
dahdievent_to_analogevent() return the event number as-is for
such an event. This is also required to fix #15924 (in addition
to r222108).
2009-10-06 01:24 +0000 [r222110-222176] Kevin P. Fleming <kpfleming@digium.com>
* /, channels/chan_sip.c, funcs/func_dialgroup.c,
include/asterisk/astobj2.h, res/res_phoneprov.c,
channels/chan_console.c, res/res_musiconhold.c, apps/app_queue.c,
channels/chan_iax2.c, main/astobj2.c, res/res_odbc.c,
res/res_calendar.c, res/res_clialiases.c: Recorded merge of
revisions 222152 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r222152 | kpfleming | 2009-10-05 20:16:36 -0500 (Mon, 05 Oct
2009) | 20 lines Fix ao2_iterator API to hold references to
containers being iterated. See Mantis issue for details of what
prompted this change. Additional notes: This patch changes the
ao2_iterator API in two ways: F_AO2I_DONTLOCK has become an enum
instead of a macro, with a name that fits our naming policy;
also, it is now necessary to call ao2_iterator_destroy() on any
iterator that has been created. Currently this only releases the
reference to the container being iterated, but in the future this
could also release other resources used by the iterator, if the
iterator implementation changes to use additional resources.
(closes issue #15987) Reported by: kpfleming Review:
https://reviewboard.asterisk.org/r/383/ ........
* main/udptl.c, channels/chan_sip.c, configs/udptl.conf.sample,
UPGRADE.txt, configs/sip.conf.sample: Allow non-compliant T.38
endpoints to be supportable via configuration option. Many T.38
endpoints incorrectly send the maximum IFP frame size they can
accept as the T38FaxMaxDatagram value in their SDP, when in fact
this value is supposed to be the maximum UDPTL payload size
(datagram size) they can accept. If the value they supply is
small enough (a commonly supplied value is '72'), T.38 UDPTL
transmissions will likely fail completely because the UDPTL
packets will not have enough room for a primary IFP frame and the
redundancy used for error correction. If this occurs, the
Asterisk UDPTL stack will emit log messages warning that data
loss may occur, and that the value may need to be overridden.
This patch extends the 't38pt_udptl' configuration option in
sip.conf to allow the administrator to override the value
supplied by the remote endpoint and supply a value that allows
T.38 FAX transmissions to be successful with that endpoint. In
addition, in any SIP call where the override takes effect, a
debug message will be printed to that effect. This patch also
removes the T38FaxMaxDatagram configuration option from
udptl.conf.sample, since it has not actually had any effect for a
number of releases. In addition, this patch cleans up the T.38
documentation in sip.conf.sample (which incorrectly documented
that T.38 support was passthrough only). (issue #15586) Reported
by: globalnetinc
2009-10-05 19:20 +0000 [r222108] Jeff Peeler <jpeeler@digium.com>
* channels/chan_dahdi.c, channels/sig_analog.c,
channels/sig_analog.h: Add a few missing events to
analog_handle_event. The reported bug was actually only for
pulsedigit, dtmfup, and dtmfdown handling. Also added recognition
for fax events (just some verbose output) and fixed handling for
the ec_disabled_event. In order to make comparing the analog
version of events to the DAHDI events easier, the ordering has
been changed to follow that of the DAHDI events. (closes issue
#15924) Reported by: tzafrir
2009-10-02 17:34 +0000 [r222030] David Vossel <dvossel@digium.com>
* /, channels/chan_iax2.c: Merged revisions 222026 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r222026 | dvossel | 2009-10-02 12:32:13 -0500 (Fri, 02
Oct 2009) | 3 lines Removes unnecessary unlock, clarifies a
memcpy. ........
2009-10-02 16:59 +0000 [r221920-221971] Tilghman Lesher <tlesher@digium.com>
* /, main/astobj2.c: Merged revisions 221970 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r221970 | tilghman | 2009-10-02 11:58:03 -0500 (Fri, 02 Oct 2009)
| 2 lines Ensure the result of the hash function is positive.
Negative array offsets suck. ........
* main/logger.c: Initialize a variable that we check immediately
upon startup. (closes issue #15973) Reported by: atis
2009-10-02 01:49 +0000 [r221844-221881] Richard Mudgett <rmudgett@digium.com>
* channels/misdn/isdn_lib.c: Whitespace change.
* channels/misdn/isdn_lib.c: Whitespace change.
* channels/misdn/isdn_lib_intern.h, /, channels/misdn/isdn_lib.c:
Merged revisions 221769 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r221769 | rmudgett | 2009-10-01 18:18:28 -0500 (Thu, 01 Oct 2009)
| 26 lines Occasionally losing use of B channels in chan_misdn. I
have not been able to reproduce the problem of losing channels.
However, I have seen in the code a reentrancy problem that might
give these symptoms. The reentrancy patch does several things: 1)
Guards B channel and B channel structure allocation. 2) Makes the
B channel structure find routines more precise in locating
records. 3) Never leave a B channel allocated if we received
cause 44. The last item may cause temporary outgoing call
problems, but they should clear when the line becomes idle.
(closes issue #15490) Reported by: slutec18 Patches:
issue15490_channel_alloc_reentrancy.patch uploaded by rmudgett
(license 664) Tested by: rmudgett, slutec18 (closes issue #15458)
Reported by: FabienToune Patches:
issue15458_channel_alloc_reentrancy.patch uploaded by rmudgett
(license 664) Tested by: FabienToune, rmudgett, slutec18 ........
2009-10-02 00:08 +0000 [r221777-221781] Tilghman Lesher <tlesher@digium.com>
* main/say.c: One more off-by-one in trunk
* main/rtp_engine.c, /, main/say.c, main/asterisk.c: Merged
revisions 221776 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r221776 | tilghman | 2009-10-01 18:53:12 -0500 (Thu, 01 Oct 2009)
| 2 lines Fix a bunch of off-by-one errors ........
2009-10-01 20:18 +0000 [r221709] Richard Mudgett <rmudgett@digium.com>
* UPGRADE.txt, CHANGES: Move DAHDI/ISDN channel naming note from
CHANGES to UPGRADE.txt.
2009-10-01 20:09 +0000 [r221705] Tilghman Lesher <tlesher@digium.com>
* channels/chan_sip.c: Revision 220906 (a merge from 1.4) was not
merged correctly, causing a problem with non-dynamic peers.
2009-10-01 19:48 +0000 [r221701] Richard Mudgett <rmudgett@digium.com>
* channels/sig_pri.h, channels/chan_dahdi.c, CHANGES: Prevent
deadlock if chan_dahdi attempts to change PRI channel names. The
PRI channels can no longer change the channel name if a different
B channel is selected during call negotiation. To prevent using
the channel name to infer what B channel a call is using and to
avoid name collisions, the channel name format is changed. The
new channel naming for PRI channels is:
DAHDI/ISDN-<span>-<sequence-number>
2009-10-01 19:33 +0000 [r221697] David Vossel <dvossel@digium.com>
* channels/chan_sip.c: outbound tls connections were not defaulting
to port 5061 (closes issue #15854) Reported by: dvossel Patches:
sip_port_config_trunk.diff uploaded by dvossel (license 671)
Tested by: dvossel Review:
https://reviewboard.asterisk.org/r/357/
2009-10-01 16:27 +0000 [r221592-221627] Kevin P. Fleming <kpfleming@digium.com>
* UPGRADE.txt: Sync up UPGRADE.txt with the 1.6.2 version.
* main/udptl.c, configs/udptl.conf.sample: Remove ability to
control T.38 FAX error correction from udptl.conf. chan_sip has
had the ability to control T.38 FAX error correction mode on a
per-peer (or global) basis for a couple of releases now, which is
where it should have been all along. This patch removes the
ability to configure it in udptl.conf, but issues a warning if
the user tries to do, telling them to look at sip.conf.sample for
how to configure it now. For any SIP peers that are T.38 enabled
in sip.conf, there is already a default for FEC error correction
even if the user does not specify any mode, so this change will
not turn off error correction by default, it will have the same
default value that has been in the udptl.conf sample file.
2009-10-01 15:26 +0000 [r221589] Matthew Nicholson <mnicholson@digium.com>
* /, channels/chan_sip.c: Merged revisions 221588 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r221588 | mnicholson | 2009-10-01 10:24:00 -0500 (Thu, 01 Oct
2009) | 2 lines Use unsigned ints for portinuri flags. ........
2009-10-01 07:00 +0000 [r221554] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: Simplify code for porturi, use TRUE/FALSE
constructs when it's just TRUE or FALSE.
2009-09-30 23:04 +0000 [r221484] Matthew Nicholson <mnicholson@digium.com>
* channels/chan_sip.c: Cleaned up merge from r221432
2009-09-30 21:15 +0000 [r221436] Matthias Nick <mnick@digium.com>
* apps/app_queue.c: Prevents from division by zero
2009-09-30 20:40 +0000 [r221432] Matthew Nicholson <mnicholson@digium.com>
* /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions
221360 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r221360 | mnicholson | 2009-09-30 14:36:06 -0500 (Wed, 30 Sep
2009) | 10 lines Fix SRV lookup and Request-URI generation in
chan_sip. This patch adds a new field "portinuri" to the sip
dialog struct and the sip peer struct. That field is used during
RURI generation to determine if the port should be included in
the RURI. It is also used in some places to determine if an SRV
lookup should occur. (closes issue #14418) Reported by: klaus3000
Tested by: klaus3000, mnicholson Review:
https://reviewboard.asterisk.org/r/369/ ........
2009-09-30 19:42 +0000 [r221368] Matthias Nick <mnick@digium.com>
* configs/cdr_custom.conf.sample, /, funcs/func_strings.c: Merged
revisions 221153,221157,221303 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r221153 | mnick | 2009-09-30 10:37:39 -0500 (Wed, 30 Sep 2009) |
2 lines check bounds - prevents for buffer overflow ........
r221157 | mnick | 2009-09-30 10:41:46 -0500 (Wed, 30 Sep 2009) |
8 lines added a new dialplan function 'CSV_QUOTE' and changed the
cdr_custom.sample.conf (closes issue #15471) Reported by: dkerr
Patches: csv_quote_14.txt uploaded by mnick (license ) Tested by:
mnick ........ r221303 | mnick | 2009-09-30 14:02:00 -0500 (Wed,
30 Sep 2009) | 2 lines changed the prototype definition of
csv_quote ........
2009-09-30 18:47 +0000 [r221266-221300] Terry Wilson <twilson@digium.com>
* res/res_rtp_asterisk.c: Remove spurious debug
* res/res_rtp_asterisk.c, main/rtp_engine.c, channels/chan_sip.c,
include/asterisk/rtp_engine.h: Use rtp properties instead of
adding a callback Thanks, Josh.
* res/res_rtp_asterisk.c, main/rtp_engine.c, /,
channels/chan_sip.c, configs/sip.conf.sample,
include/asterisk/rtp_engine.h: Merged revisions 221086 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r221086 | twilson | 2009-09-30 09:49:11 -0500 (Wed, 30 Sep 2009)
| 25 lines Change the SSRC by default when our media stream
changes Be default, change SSRC when doing an audio stream
changes Asterisk doesn't honor marker bit when reinvited to
already-bridged RTP streams,resulting in far-end stack discarding
packets with "old" timestamps that areactually part of a new
stream. This patch sends AST_CONTROL_SRCUPDATE whenever there is
a reinvite, unless the 'constantssrc' is set to true in sip.conf.
The original issue reported to Digium support detailed the
following situation: ITSP <-> Asterisk 1.4.26.2 <-> SIP-based
Application Server Call comes in fromITSP, Asterisk dials the app
server which sends a re-invite back toAsterisk--not to negotiate
to send media directly to the ITSP, but to indicatethat it's
changing the stream it's sending to Asterisk. The app
servergenerates a new SSRC, sequence numbers, timestamps, and
sets the marker bit on the new stream. Asterisk passes through
the teimstamp of the new stream, butdoes not reset the SSRC,
sequence numbers, or set the marker bit. When the timestamp on
the new stream is older than the timestamp on the originalstream,
the ITSP (which doesn't know there has been any change) discards
the newframes because it thinks they are too old. This patch
addresses this by changing the SSRC on a stream update unless
constantssrc=true is set in sip.conf. Review:
https://reviewboard.asterisk.org/r/374/ ........
2009-09-30 16:56 +0000 [r221201] Tilghman Lesher <tlesher@digium.com>
* main/channel.c, /: Merged revisions 221200 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r221200 | tilghman | 2009-09-30 11:55:21 -0500 (Wed, 30 Sep 2009)
| 7 lines Avoid a potential NULL dereference. (closes issue
#15865) Reported by: kobaz Patches: 20090915__issue15865.diff.txt
uploaded by tilghman (license 14) Tested by: kobaz ........
2009-09-30 15:11 +0000 [r221085-221090] Sean Bright <sean@malleable.com>
* apps/app_voicemail.c: Modify VoiceMailMain()'s a() argument to
allow mailboxes to be specified by name. (closes issue #14740)
Reported by: pj Patches: issue14740_09022009.diff uploaded by
seanbright (license 71) Tested by: seanbright, lmadsen
* apps/app_voicemail.c: Clarify documentation for VoiceMailMain()'s
a() option. We require box numbers, not names as the
documentation implies. (issue #14740) Reported by: pj Patches:
__20090729-app_voicemail-documentation.patch uploaded by lmadsen
(license 10) Tested by: seanbright, lmadsen
2009-09-30 04:32 +0000 [r221044] Tilghman Lesher <tlesher@digium.com>
* funcs/func_lock.c: Allow locks to be inherited through a
masquerade without causing starvation. (closes issue #14859)
Reported by: atis Patches: 20090821__issue14859.diff.txt uploaded
by tilghman (license 14) 20090925__issue14859__1.6.1.diff.txt
uploaded by tilghman (license 14) Tested by: atis, tilghman
2009-09-29 21:28 +0000 [r220920-220995] Mark Michelson <mmichelson@digium.com>
* main/cel.c: Fix channel reference leak. ast_cel_report_event
would geet a reference to the bridged channel. However, certain
return paths, such as if CEL was not enabled, would result in a
reference leak. All return paths now properly unref the channel.
(closes issue #15991) Reported by: mmichelson
* main/rtp_engine.c: Get rid of annoying and cryptic debug
messages.
2009-09-29 19:57 +0000 [r220906] Tilghman Lesher <tlesher@digium.com>
* /, channels/chan_sip.c: Merged revisions 220873 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r220873 | tilghman | 2009-09-29 12:59:26 -0500 (Tue, 29 Sep 2009)
| 9 lines Reduce CPU usage related to building a peer merely for
devicestates. This fixes a 100% CPU problem in the SIP driver,
found by profiling the driver while the problem was occurring.
(closes issue #14309) Reported by: pkempgen Patches:
20090924__issue14309.diff.txt uploaded by tilghman (license 14)
Tested by: pkempgen, vrban ........
2009-09-29 19:49 +0000 [r220904] Matthew Nicholson <mnicholson@digium.com>
* apps/app_confbridge.c: Fix options 'm' and 's'. They were swapped
in the code. Also document the fact that app_confbridge does not
automatically answer the channel. (closes issue #15964) Reported
by: shrift
2009-09-29 16:58 +0000 [r220833] Jeff Peeler <jpeeler@digium.com>
* apps/app_voicemail.c: Make deletion of temporary greetings work
properly with IMAP_STORAGE When imapgreetings was set to yes, the
message was being deleted but wasn't actually being expunged.
When imapgreetings was set to no, the file based message was not
being deleted at all. All good now! (closes issue #14949)
Reported by: noahisaac Patches: vm_tempgreeting_removal.patch
uploaded by noahisaac (license 748), modified by me
2009-09-28 21:02 +0000 [r220792] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c, channels/sig_pri.c: Miscellaneous minor
changes.
2009-09-28 19:11 +0000 [r220721] Sean Bright <sean@malleable.com>
* /, Makefile.rules: Merged revisions 220717 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r220717 | seanbright | 2009-09-28 15:09:25 -0400 (Mon, 28 Sep
2009) | 3 lines When selecting DONT_OPTIMIZE in menuselect,
explicitly pass -O0 to the compiler so we override any default
optimization levels for a particular install. ........
2009-09-28 19:10 +0000 [r220718] Jeff Peeler <jpeeler@digium.com>
* channels/chan_sip.c: Fix building of registration entry in
build_peer when using callbackextension Check for remotesecret
option was unintentionally always true, which therefore caused
the secret option to never be used. Thanks to dvossel for
pointing out the exact fix. (closes issue #15943) Reported by:
tpsast
2009-09-28 15:27 +0000 [r220672] Richard Mudgett <rmudgett@digium.com>
* channels/sig_pri.h, channels/sig_pri.c: Locking issues dealing
with service_lock. * Removed unneeded and uninitialized
service_lock. * Fixed potential locking imbalance in
pri_dchannel():PRI_EVENT_RESTART. * Fixed verbose message typo in
pri_dchannel():PRI_EVENT_RESTART.
2009-09-27 20:40 +0000 [r220629] Michiel van Baak <michiel@vanbaak.info>
* funcs/func_callerid.c: add name argument for the CALLERID
dialplan function to the xml documentation. Pointed out to me on
IRC by snuff-home. Thanks
2009-09-26 15:10 +0000 [r220586] Tilghman Lesher <tlesher@digium.com>
* include/asterisk/aes.h: Allow AES to compile, when OpenSSL is not
present.
2009-09-25 19:56 +0000 [r220543] Richard Mudgett <rmudgett@digium.com>
* channels/sig_pri.c: Reduce indentation in sig_pri_available().
2009-09-25 14:50 +0000 [r220494-220496] Kevin P. Fleming <kpfleming@digium.com>
* main/manager.c: Eliminate unnecessary include of version.h in
manager.c. Including version.h here causes this file to get
recompiled after every commit or update, which is not needed.
* main/channel.c: Correct sense of logic test committed in revision
220494.
* main/channel.c: Don't use hash-based lookups for
ast_channel_get_by_name_prefix(). ast_channel_get_full() tries to
use OBJ_POINTER to optimize name-based channel lookups, but this
will not work properly when the channel's full name was not
supplied; for name-prefix searches, there is no value in doing a
hash-based lookup, and in fact doing so could result in many
channels being skipped.
2009-09-25 10:54 +0000 [r220457] Philippe Sultan <philippe.sultan@gmail.com>
* channels/chan_jingle.c, configs/jabber.conf.sample,
include/asterisk/jabber.h, channels/chan_gtalk.c, CHANGES,
doc/jabber.txt, res/res_jabber.c: Add JABBER_RECEIVE as a
dialplan function, implement SendText in Jingle channels
JABBER_RECEIVE (along with JabberSend) makes Asterisk interact
with users over XMPP to process calls. SendText can be used
instead of JabberSend in the context of XMPP based voice channels
(chan_gtalk and chan_jingle). (closes issue #12569) Reported by:
eech55 Tested by: phsultan, asannucci, lmadsen, jtodd, maxgo
Review: https://reviewboard.asterisk.org/r/88/
2009-09-24 22:53 +0000 [r220417] Tilghman Lesher <tlesher@digium.com>
* UPGRADE.txt, main/asterisk.c: Change the default behavior of Set,
AGI, and pbx_realtime to 1.6 behavior by default (starting in
1.6.3).
2009-09-24 20:37 +0000 [r220365] David Vossel <dvossel@digium.com>
* main/tcptls.c: fixes tcptls_session memory leak caused by ref
count error (closes issue #15939) Reported by: dvossel Review:
https://reviewboard.asterisk.org/r/375/
2009-09-24 20:29 +0000 [r220344] Jeff Peeler <jpeeler@digium.com>
* apps/app_dial.c, main/features.c, include/asterisk/features.h:
Add bridge related dial flags to the bridge app Most of the
functionality here is gained simply by setting the feature flag
on the bridge config. However, the dial limit functionality has
been moved from app_dial to the features code and has been made
public so both app_dial and the bridge app can use it. (closes
issue #13165) Reported by: tim_ringenbach Patches:
app_bridge_options_r138998.diff uploaded by tim ringenbach
(license 540), modified by me
2009-09-24 19:57 +0000 [r220295] Olle Johansson <oej@edvina.net>
* configs/sip.conf.sample: Documentation in the commit messages is
soon forgotten, please add it to the docs in the product.
2009-09-24 19:41 +0000 [r220289] Tilghman Lesher <tlesher@digium.com>
* main/pbx.c, /, apps/app_disa.c, apps/app_playback.c: Merged
revisions 220288 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r220288 | tilghman | 2009-09-24 14:39:41 -0500 (Thu, 24 Sep 2009)
| 6 lines Implicitly sending a progress signal breaks some
applications. Call Progress() in your dialplan if you explicitly
want progress to be sent. (Reverts change 216430, closes issue
#15957) Reported by: Pavel Troller on the Asterisk-Dev mailing
list
http://lists.digium.com/pipermail/asterisk-dev/2009-September/039897.html
........
2009-09-24 18:19 +0000 [r220217] Sean Bright <sean@malleable.com>
* Makefile, /: Merged revisions 220213 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r220213 | seanbright | 2009-09-24 14:18:18 -0400 (Thu, 24 Sep
2009) | 1 line Resolve parallel build warnings. Reported by Klaus
Darilion on the asterisk-dev mailing list. ........
2009-09-24 16:33 +0000 [r220174] Matthew Nicholson <mnicholson@digium.com>
* channels/chan_sip.c: Ensure the numeric portion of the
P-Asserted-Identity header is properly escaped.
2009-09-24 14:44 +0000 [r220100] Sean Bright <sean@malleable.com>
* Makefile, build_tools/mkpkgconfig, /: Merged revisions 220099 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r220099 | seanbright | 2009-09-24 10:41:57 -0400 (Thu, 24 Sep
2009) | 2 lines Remove the remaining bashisms in the
Makefile/mkpkgconfig ........
2009-09-24 08:36 +0000 [r220028] Michiel van Baak <michiel@vanbaak.info>
* build_tools/mkpkgconfig, /: Merged revisions 220027 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r220027 | mvanbaak | 2009-09-24 10:33:50 +0200 (Thu, 24
Sep 2009) | 7 lines mkpkgconfig does not need bash so make it use
/bin/sh This fixes building on all systems that don't have bash
at /bin/bash Reported by _ys on #asterisk-dev Tested by _ys on
#asterisk-dev ........
2009-09-24 07:39 +0000 [r219951-219987] Tilghman Lesher <tlesher@digium.com>
* apps/app_directory.c: Fix two possible crashes, one only in 1.6.1
and one in 1.6.1 forward. (closes issue #15739) Reported by:
DLNoah, jeffg Patches: 20090914__issue15739.diff.txt uploaded by
tilghman (license 14) 20090922__issue15739.diff.txt uploaded by
tilghman (license 14) Tested by: DLNoah, jeffg
* configs/mgcp.conf.sample, CHANGES, channels/chan_mgcp.c: Add
support for 'setvar=' for MGCP device lines, like other channel
drivers provide. (closes issue #14818) Reported by:
alea-soluciones Patches:
chan_mgcp-setvars-svn-trunk-r219899.patch uploaded by alea
(license 514)
* doc/lang/language-criteria.txt: Update fax number to the legal
fax, not the generic fax. (closes issue #15946) Reported by:
jtodd Patches: leif-is-a-wuss.txt uploaded by jtodd (license 870)
Tested by: jparker, tilghman, jtodd, russellb, mmichelson,
seanbright, kpfleming, and the rest of the usual suspects
2009-09-23 17:46 +0000 [r219895] Leif Madsen <lmadsen@digium.com>
* include/asterisk/doxyref.h,
include/asterisk/doxygen/mantisworkflow.h (added): Add Mantis
work flow documention. This commit adds the doxygen changes that
I've made to describe the Mantis work flow documentation for the
open source issue tracker. This should make it easier to
determine the flow of issues through the issue tracker, and what
those statuses mean. (closes issue #15902) Reported by: lmadsen
Patches: mantisworkflow.h uploaded by lmadsen (license 10)
Review: https://reviewboard.asterisk.org/r/367/
2009-09-22 21:43 +0000 [r219818] Tilghman Lesher <tlesher@digium.com>
* /, apps/app_voicemail.c: Merged revisions 219816 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r219816 | tilghman | 2009-09-22 16:37:03 -0500 (Tue, 22
Sep 2009) | 10 lines When IMAP variables were changed during a
reload, Voicemail did not use the new values. This change
introduces a configuration version variable, which ensures that
connections with the old values are not reused but are allowed to
expire normally. (closes issue #15934) Reported by:
viniciusfontes Patches: 20090922__issue15934.diff.txt uploaded by
tilghman (license 14) Tested by: viniciusfontes ........
2009-09-21 16:59 +0000 [r219721] David Vossel <dvossel@digium.com>
* /, channels/chan_iax2.c: Merged revisions 219720 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r219720 | dvossel | 2009-09-21 11:55:53 -0500 (Mon, 21
Sep 2009) | 3 lines Reverting merge 219520. This change was not
necessary. ........
2009-09-20 17:55 +0000 [r219654] Tilghman Lesher <tlesher@digium.com>
* /, main/file.c: Merged revisions 219653 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r219653 | tilghman | 2009-09-20 12:52:05 -0500 (Sun, 20 Sep 2009)
| 8 lines Really stop the stream, when ast_closestream() is
called. (closes issue #15129) Reported by: bmh Patches:
20090918__issue15129.diff.txt uploaded by tilghman (license 14)
Review: https://reviewboard.asterisk.org/r/372/ ........
2009-09-19 02:59 +0000 [r219587] Russell Bryant <russell@digium.com>
* /, channels/chan_iax2.c: Merged revisions 219586 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r219586 | russell | 2009-09-18 21:51:13 -0500 (Fri, 18
Sep 2009) | 6 lines Make sure the iax_pvt exists before
dereferencing it. This fixes the latest crash posted on issue
15609. (issue #15609) ........
2009-09-18 23:20 +0000 [r219451-219520] David Vossel <dvossel@digium.com>
* /, channels/chan_iax2.c: Merged revisions 219519 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r219519 | dvossel | 2009-09-18 18:19:50 -0500 (Fri, 18
Sep 2009) | 9 lines iax2 frame double free The iax frame's
retrans sched id was written over right before iax2_frame_free
was called. In iax2_frame_free that retrans id is used to delete
the sched item. By writing over the retrans field before the
sched item could be deleted, it was possible for a retransmit to
occur on a freed frame. ........
* /, channels/chan_sip.c: Merged revisions 219450 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r219450 | dvossel | 2009-09-18 11:19:15 -0500 (Fri, 18 Sep 2009)
| 14 lines via-header branches not updated correctly on INVITE
INVITE requests must always contain a new unique branch id. When
a new branch id is created for an INVITE, the dialog's
invite_branch variable must be updated so CANCEL requests use the
correct branch id. (closes issue #15262) Reported by: maniax
Patches: asterisk-1.6.1.0-sip-branch.patch uploaded by tweety
(license 608) invite_new_branch_trunk.diff uploaded by dvossel
(license 671) Tested by: maniax, dvossel ........
2009-09-18 13:54 +0000 [r219412] Tilghman Lesher <tlesher@digium.com>
* apps/app_voicemail.c: Missing value setting line for
maxsecs/maxmessage (closes issue #15696) Reported by:
fhackenberger Patches: maxsecs.patch uploaded by fhackenberger
(license 592)
2009-09-17 22:37 +0000 [r219371] David Vossel <dvossel@digium.com>
* channels/chan_sip.c: fixes deadlock when performing directed
pickup w Invite/replaces (closes issue #15340) Reported by:
lmsteffan Patches: deadlock.patch uploaded by lmsteffan (license
779) Tested by: lmsteffan
2009-09-17 22:22 +0000 [r219324] Mark Michelson <mmichelson@digium.com>
* /, channels/chan_sip.c: Merged revisions 219320 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r219320 | mmichelson | 2009-09-17 17:20:50 -0500 (Thu, 17 Sep
2009) | 6 lines Send a 100 Trying response when we detect a
spiral. This was problematic during spiral tests at SIPit...
along with some other things as well. ........
2009-09-17 21:59 +0000 [r219304] David Vossel <dvossel@digium.com>
* /, channels/chan_sip.c: Merged revisions 219303 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r219303 | dvossel | 2009-09-17 16:29:37 -0500 (Thu, 17 Sep 2009)
| 21 lines INVITE w/Replaces deadlock fix This patch cleans up
the locking logic in chan_sip.c's handle_invite_replaces()
function as well as making use of ast_do_masquerade() rather than
forcing the masquerade on an ast_read(). The code had several
redundant unlocks that would result in 'freed more times than
we've locked!' errors. I cleaned these up as well as moving all
the unlock logic to the end of the function. This patch should
also resolve the issue people were having with the replacecall
channel never being unlocked with one legged calls. (closes issue
#15151) Reported by: irroot Patches: invite_w_replaces_1.4.diff
uploaded by dvossel (license 671) Tested by: irroot, dvossel
Review: https://reviewboard.asterisk.org/r/371/ ........
2009-09-17 19:57 +0000 [r219264] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: Ensure no spaces exist before "refresher="
when doing the comparison.
2009-09-17 16:25 +0000 [r219230] Sean Bright <sean@malleable.com>
* apps/app_chanspy.c: Get this compiling under dev-mode.
2009-09-17 15:18 +0000 [r219139] Matthew Nicholson <mnicholson@digium.com>
* main/channel.c, /, include/asterisk/cdr.h: Merged revisions
219136 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r219136 | mnicholson | 2009-09-17 09:58:39 -0500 (Thu, 17 Sep
2009) | 10 lines Prevent a potential race condition and crash
when hanging up a channel by removing the channel from the
channel list before begining channel tear down. This fix may
potentially cause problems with CDR backends that access the
channel a CDR is associated with via the channel list. This fix
makes the channel unavabile at the time when the CDR backend is
invoked. This has been documented in include/asterisk/cdr.h.
(closes issue #15316) Reported by: vmarrone Tested by: mnicholson
Review: https://reviewboard.asterisk.org/r/362/ ........
2009-09-17 00:58 +0000 [r219007-219105] Tilghman Lesher <tlesher@digium.com>
* CHANGES, apps/app_chanspy.c: Add the 'E' option to exit ChanSpy,
once the single channel it spied upon hangs up. In addition,
there's a bit of cleanup to the arguments and documentation, in
which I discovered that the last feature added to this
application duplicated an option (oops!) and changed that option
so that it now works. (closes issue #14909) Reported by: junky
Patches: __20090901-spy_hangup_trunk.diff uploaded by lmadsen
(license 10) Tested by: amilcar, junky, flujan, lmadsen
* /, main/config.c, configs/extensions.conf.sample: Merged
revisions 219023 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r219023 | tilghman | 2009-09-16 18:21:53 -0500 (Wed, 16 Sep 2009)
| 8 lines Properly deal with quotes in the arguments of '#exec'
includes. (closes issue #15583) Reported by: pkempgen Patches:
20090726__issue15583.diff.txt uploaded by tilghman (license 14)
20090726__issue15583-1.4-4.diff.txt uploaded by pkempgen (license
169) Tested by: pkempgen ........
* configure, include/asterisk/autoconfig.h.in, configure.ac: Detect
whether we actually have the long double type, before looking for
those functions. (closes issue #15017) Reported by: tzafrir
Patches: 20090916__issue15017.diff.txt uploaded by tilghman
(license 14) Tested by: tzafrir
2009-09-16 20:32 +0000 [r218973] Sean Bright <sean@malleable.com>
* res/res_jabber.c: Remove some unused defines from res_jabber.
(closes issue #15359) Reported by: snuffy Patches:
bug_res_jabber_unused_defines.diff uploaded by snuffy (license
35)
2009-09-16 19:25 +0000 [r218933] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Reverse order of args to fread. This way, we
don't always write a null byte into byte 1 of the buffer (closes
issue #15905) Reported by: ebroad Patches: freadfix.patch
uploaded by ebroad (license 878) Tested by: ebroad
2009-09-16 18:31 +0000 [r218918] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: On TCP and TLS connections do not attempt to
stop retransmission of the packet internally. This was preventing
responses from being properly processed because the packet was
not being found causing handle_response to return prematurely.
2009-09-16 18:06 +0000 [r218868] David Brooks <dbrooks@digium.com>
* main/pbx.c, /: Merged revisions 218867 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r218867 | dbrooks | 2009-09-16 13:00:45 -0500 (Wed, 16 Sep 2009)
| 13 lines Fixes CID pattern matching behavior to mirror that of
extension pattern matching. Pattern matching for extensions uses
a type of scoring system, giving values for specificity to each
character in the pattern. Unfortunately, this is done character
by character, in order. This does lead to some less specific
patterns being first in line for matching, but it will usually
get the job done. This patch merely brings CID matching to the
same level as extension matching. This patch does not attempt to
tackle the problem shared by extension matching. (closes issue
#14708) Reported by: klaus3000 ........
2009-09-16 13:34 +0000 [r218799] Russell Bryant <russell@digium.com>
* contrib/firmware/iax/iaxy.bin (removed), /, UPGRADE.txt: Merged
revisions 218798 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r218798 | russell | 2009-09-16 08:33:43 -0500 (Wed, 16 Sep 2009)
| 9 lines Remove the IAXy firmware from Asterisk. The firmware
can now be found on downloads.digium.com, where the rest of our
binary downloads live. This was the last part of our Asterisk
tarballs that was considered non-free by Debian. :-) (closes
issue #15838) Reported by: paravoid ........
2009-09-15 22:33 +0000 [r218731] Tilghman Lesher <tlesher@digium.com>
* /, apps/app_voicemail.c: Merged revisions 218730 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r218730 | tilghman | 2009-09-15 17:27:41 -0500 (Tue, 15
Sep 2009) | 6 lines If the user enters the same password as
before, don't signal an error when the change does nothing.
(closes issue #15492) Reported by: cbbs70a Patches:
20090713__issue15492.diff.txt uploaded by tilghman (license 14)
........
2009-09-15 19:22 +0000 [r218687] David Vossel <dvossel@digium.com>
* channels/chan_sip.c: upward bound checking for port string to int
conversion
2009-09-15 16:15 +0000 [r218586] Matthew Nicholson <mnicholson@digium.com>
* /, channels/chan_sip.c: Merged revisions 218578 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r218578 | mnicholson | 2009-09-15 11:03:54 -0500 (Tue, 15 Sep
2009) | 8 lines Send request contact header field with response
to registrer queries instead of the address of record. (closes
issue #14438) Reported by: ravindrad Patches: regquerypatch
uploaded by ravindrad (license 684) Tested by: ravindrad ........
2009-09-15 16:12 +0000 [r218583] Jeff Peeler <jpeeler@digium.com>
* channels/chan_dahdi.c: Add some changes related to 218430. *
Remove thread_spawned in handle_init_event since it was never
used * Always check handle_init_event in case a channel is
destroyed
2009-09-15 16:04 +0000 [r218579] Tilghman Lesher <tlesher@digium.com>
* /, apps/app_followme.c: Merged revisions 218577 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r218577 | tilghman | 2009-09-15 11:01:17 -0500 (Tue, 15 Sep 2009)
| 9 lines Ensure FollowMe sets language in channels it creates.
Also, not in the original bug report, but related fields are
accountcode and musicclass, and the inheritance of datastores.
(closes issue #15372) Reported by: Romik Patches:
20090828__issue15372.diff.txt uploaded by tilghman (license 14)
Tested by: cervajs ........
2009-09-15 15:40 +0000 [r218504-218566] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Use a better method of ensuring
null-termination of the buffer while reading the SDP when using
TCP.
* channels/chan_sip.c: Ensure that SDP read from TCP socket is
null-terminated.
2009-09-15 15:02 +0000 [r218500] Kevin P. Fleming <kpfleming@digium.com>
* /: Merged revisions 218497 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r218497 | kpfleming | 2009-09-15 10:55:58 -0400 (Tue, 15 Sep
2009) | 1 line Use proper hostname for downloading sound files.
........
2009-09-15 14:59 +0000 [r218499] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Fix off-by-one error when reading SDP sent
over TCP.
2009-09-15 10:24 +0000 [r218465] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
* channels/chan_dahdi.c: Fix false error message on
DAHDI_EVENT_REMOVED (RESULT_SUCCESS == 0)
2009-09-14 22:38 +0000 [r218430] Jeff Peeler <jpeeler@digium.com>
* channels/chan_dahdi.c, channels/sig_analog.c, /,
channels/sig_analog.h: Merged revisions 218401 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r218401 | jpeeler | 2009-09-14 16:47:11 -0500 (Mon, 14 Sep 2009)
| 11 lines Fix handling of DAHDI_EVENT_REMOVED event to prevent
crash in do_monitor. After talking to rmudgett about some of his
recent iflist locking changes, it was determined that the only
place that would destroy a channel without being explicitly to do
so was in handle_init_event. The loop to walk the interface list
has been modified to wait to destroy the channel until the
dahdi_pvt of the channel to be destroyed is no longer needed.
(closes issue #15378) Reported by: samy ........
2009-09-14 20:08 +0000 [r218365] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c: Add support for multiple interface lists.
Also unlink the sig_pri_pri.pvts[] pointer in
destroy_dahdi_pvt().
2009-09-14 19:29 +0000 [r218361] Tilghman Lesher <tlesher@digium.com>
* /, configs/voicemail.conf.sample, sounds/Makefile,
apps/app_voicemail.c: Recorded merge of revisions 218331 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r218331 | tilghman | 2009-09-14 14:16:35 -0500 (Mon, 14 Sep 2009)
| 4 lines Don't say "Please try again" if we don't give the user
another chance to try again. (issue #15055, SWP-129) Reported by:
jthurman ........
2009-09-14 18:16 +0000 [r218295] Joshua Colp <jcolp@digium.com>
* main/features.c: Do not attempt to add a parking extension if an
error occurred while reading the configuration.
2009-09-14 14:57 +0000 [r218224] Matthew Nicholson <mnicholson@digium.com>
* /, apps/app_directed_pickup.c: Merged revisions 218223 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r218223 | mnicholson | 2009-09-14 09:53:57 -0500 (Mon, 14 Sep
2009) | 8 lines Ensure we don't pickup ourselves when doing
pickup by exten. (closes issue #15100) Reported by: lmsteffan
Patches: (modified) pickup.patch uploaded by lmsteffan (license
779) ........
2009-09-13 17:34 +0000 [r218184] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
* channels/chan_phone.c: gcc 4.4: Remove a nop memset size 0 that
annoys gcc This memset doesn't write beyond the end of the
buffer. (tmpbuf has size of 4).
2009-09-13 05:51 +0000 [r218150] Moises Silva <moises.silva@gmail.com>
* channels/chan_dahdi.c: get rid of mfcr2 monitor thread condition,
is problematic
2009-09-12 13:08 +0000 [r218107] Michiel van Baak <michiel@vanbaak.info>
* res/res_rtp_asterisk.c: use the actual given ip address for 'rtp
set debug ip <foo>' instead of the word 'ip' (closes issue
#15711) Reported by: davidw Patches: 2009082800-rtpdebug.diff.txt
uploaded by mvanbaak (license 7) Tested by: davidw
2009-09-11 05:58 +0000 [r217990-218050] Tilghman Lesher <tlesher@digium.com>
* main/pbx.c: Check the origination priority for more matches, not
the current priority. Found by Pavel Troller on the -dev list.
* /, apps/app_queue.c: Merged revisions 217989 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r217989 | tilghman | 2009-09-10 18:52:22 -0500 (Thu, 10 Sep 2009)
| 3 lines Don't ring another channel, if there's not enough time
for a queue member to answer. (Fixes AST-228) ........
2009-09-10 23:49 +0000 [r217954-217987] Jeff Peeler <jpeeler@digium.com>
* channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_pri.c:
Cleanup approach in 217804 and don't reach inside the sig_pvt.
* channels/chan_dahdi.c, channels/sig_analog.c,
channels/sig_analog.h: Allow do not disturb to be set on analog
channels via the CLI and AMI.
2009-09-10 23:12 +0000 [r217916] Tilghman Lesher <tlesher@digium.com>
* contrib/scripts/iax-friends.sql, channels/chan_sip.c,
channels/chan_iax2.c: Make calltoken support work with realtime
users and peers. In the course of this, I also found that the
results of ast_gethostbyname were being used incorrectly in both
chan_iax2 and chan_sip, so both have been fixed.
2009-09-10 22:31 +0000 [r217873-217912] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c: Cleaned up chan_dahdi iflist handling and
locking. * Fixed walking the iflist so it is always done with the
iflock locked. * Simplified iflist walking routines. * Created
chan_dahdi iflist insertion and extraction routines. * Fixed
duplicate_pseudo() malloc fail handling. * Fixed infinite loop in
action_dahdishowchannels() when showing a single channel.
* channels/chan_dahdi.c: Miscellaneous minor changes.
2009-09-10 21:07 +0000 [r217807] David Vossel <dvossel@digium.com>
* /, channels/chan_iax2.c: Merged revisions 217806 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r217806 | dvossel | 2009-09-10 16:06:07 -0500 (Thu, 10
Sep 2009) | 22 lines IAX2 encryption regression The IAX2 Call
Token security patch inadvertently broke the use of encryption
due to the reorganization of code in the socket_process()
function. When encryption is used, an incoming full frame must
first be decrypted before the information elements can be parsed.
The security release mistakenly moved IE parsing before
decryption in order to process the new Call Token IE. To resolve
this, decryption of full frames is once again done before looking
into the frame. This involves searching for an existing callno,
checking the pvt to see if encryption is turned on, and
decrypting the packet before the internal fields of the full
frame are accessed. (closes issue #15834) Reported by: karesmakro
Patches: iax2_encryption_fix_1.4.diff uploaded by dvossel
(license 671) Tested by: dvossel, karesmakro Review:
https://reviewboard.asterisk.org/r/355/ ........
2009-09-10 20:52 +0000 [r217744-217804] Jeff Peeler <jpeeler@digium.com>
* channels/chan_dahdi.c: Fix crash during attended transfer over
PRI. The owner pointers in the sig_pri_chan structure were not
getting updated in dahdi_fixup.
* channels/chan_dahdi.c, channels/sig_analog.c,
channels/sig_analog.h: Stop caller id transmission when offhook
event detected. This fixes the problem that would occur if an
analog phone was picked up while the caller id was being sent.
The caller id before sent the whole spill even after pickup and
is now corrected.
2009-09-10 19:39 +0000 [r217730] Matthias Nick <mnick@digium.com>
* res/res_musiconhold.c: Sets the correct musicclass after an
announcement (closes issue #15279) Reported by: mbeckwell
Patches: patch.txt uploaded by mnick (license ) Tested by: mnick
(closes issue #15832) Reported by: mbeckwell Patches: patch.txt
uploaded by mnick (license 874) Tested by: mnick
2009-09-10 18:29 +0000 [r217663] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: Don't assign UINT_MAX to an INT.
2009-09-10 18:17 +0000 [r217638] Tilghman Lesher <tlesher@digium.com>
* res/res_config_odbc.c, configure,
include/asterisk/autoconfig.h.in, configure.ac: Verify support
for wide ODBC character types before using them. (closes issue
#15870) Reported by: nic_bellamy
2009-09-10 12:06 +0000 [r217593] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: Include ActionID in all events that are
responsed to AMI Action SIPShowRegistry (closes issue #15868)
Reported by: nic_bellamy Patches:
manager_SIPshowregistry_actionid.patch uploaded by nic bellamy
(license 299)
2009-09-10 00:35 +0000 [r217560] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c: Fix available() for SS7, MFC/R2, and
pseudo channels.
2009-09-09 21:48 +0000 [r217524] Moises Silva <moises.silva@gmail.com>
* channels/chan_dahdi.c: ast_log replaced for ast_verbose in MFCR2
event notifications
2009-09-09 20:09 +0000 [r217482] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: Don't report transfer success until we
actually know. 1xx messages are not final. Related to #12713
Patch by oej A big thank you to file for finally fixing the
transfer() dialplan application. I've been waiting for years for
this. Great work!
2009-09-09 18:52 +0000 [r217445] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
* res/res_phoneprov.c: gcc 4.4 fix: union instead of cast gcc 4.4
has more strict rules for aliasing. It doesn't like a struct
sockaddr_in pointer pointing to a struct sockaddr. So we make it
a union.
2009-09-09 12:11 +0000 [r217408] Sean Bright <sean@malleable.com>
* main/manager.c: Properly terminate the response to the manager
Ping action. In passing, correct the formatting of the Timestamp
attribute so that there is a space after the colon and before the
value. (closes issue #15861) Reported by: Ivan
2009-09-09 10:39 +0000 [r217367-217368] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: Not having any TLS session to write to is a
serious XMIT_ERROR.
* channels/chan_sip.c: Formatting and doxygen updates
2009-09-08 23:37 +0000 [r217331-217332] Richard Mudgett <rmudgett@digium.com>
* channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_analog.c,
channels/sig_analog.h, channels/sig_pri.c: Fix memory leak of
sig_xxx private structures.
* channels/chan_dahdi.c: Miscellaneous minor code cleanup in
mkintf().
2009-09-08 22:17 +0000 [r217286] Sean Bright <sean@malleable.com>
* apps/app_meetme.c: Fix compilation of app_meetme. Reported by
ebroad in #asterisk-bugs
2009-09-08 21:17 +0000 [r217236] Richard Mudgett <rmudgett@digium.com>
* channels/sig_pri.c: Remove duplicate entry in the sig_pri_pri
private pointer array.
2009-09-08 20:28 +0000 [r217199] Tilghman Lesher <tlesher@digium.com>
* /, apps/app_meetme.c: Merged revisions 217156 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r217156 | tilghman | 2009-09-08 15:01:45 -0500 (Tue, 08 Sep 2009)
| 7 lines When MOH is playing on the channel, announcements sent
through the conference are not heard. (closes issue #14588)
Reported by: voipas Patches: 20090716__issue14588__2.diff.txt
uploaded by tilghman (license 14) Tested by: lmadsen, twisted,
tilghman ........
2009-09-08 20:06 +0000 [r217158] Mark Michelson <mmichelson@digium.com>
* include/asterisk/event.h: Add doxygen to ast_event_subscribe for
the description. Most importantly, note that a NULL description
will cause a crash, as I just experienced that firsthand.
2009-09-08 18:06 +0000 [r217113] Russell Bryant <russell@digium.com>
* addons/format_mp3.c: Fix audio problems with format_mp3. This
problem was introduced when the AST_FRIENDLY_OFFSET patch was
merged. I'm surprised that nobody noticed any trouble when
testing that patch, but this fixes the code that fills in the
buffer to start filling in after the offset portion of the
buffer. (closes issue #15850) Reported by: 99gixxer Patches:
issue15850.diff1.txt uploaded by russell (license 2) Tested by:
99gixxer
2009-09-08 16:37 +0000 [r217074] Kevin P. Fleming <kpfleming@digium.com>
* configure, include/asterisk/autoconfig.h.in, configure.ac: Ensure
that the default autoconf CFLAGS are not used. A recent change to
the configure script that allows the user to specify CFLAGS
and/or LDFLAGS to the script had the unfortunate side effect of
letting autoconf's default CFLAGS (-g -O2) feed in to the rest of
the build system, thereby overriding the DONT_OPTIMIZE setting in
menuselect. That problem is now corrected.
2009-09-08 15:30 +0000 [r217033] Tilghman Lesher <tlesher@digium.com>
* res/res_limit.c: Remove what appears to be an unnecessary define.
(closes issue #15851) Reported by: tzafrir
2009-09-08 15:23 +0000 [r217015] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
* contrib/scripts/live_ast: live_ast: Fix asterisk.conf instead of
regenerating it * Don't write asterisk.conf from scratch. Fix the
existing one. * Pass extra 'make' command-line arguments to
'install' and 'samples'. * Fix some extra typos. closes issue
#15019 .
2009-09-08 14:26 +0000 [r216993] David Vossel <dvossel@digium.com>
* channels/chan_sip.c: caller id number empty parse_uri was not
being given the correct scheme's, as a result, uri parsing did
not parse the username correctly. One of the side effects of this
is an empty caller id. (closes issue #15839) Reported by: ebroad
Patches: blank_cidv2.patch uploaded by ebroad (license 878)
parse_uri_fix.diff uploaded by dvossel (license 671) Tested by:
ebroad, dvossel
2009-09-07 20:23 +0000 [r216883-216956] Olle Johansson <oej@edvina.net>
* doc/manager_1_1.txt: Fixing formatting
* doc/manager_1_1.txt: Add new actions under "new actions" and not
in the top of the document
* channels/chan_sip.c: Moving another function declared in the
middle of forward declarations. Please follow the structure of
the source code, thanks. Chan_sip is messy enough as it is :-)
* channels/chan_sip.c: Move "deprecated_username" to a flag like
the others - unsigned int blah:1
* channels/chan_sip.c: - Doxygen additions - Remove unused string
in sip_registry -- "random" - Someone added a function in the
middle of all forward declarations... Weird. Moved it out of that
section.
* channels/chan_sip.c: Clean up the "offered_media" code - Add
variable for number of known media streams instead of hardcoding
in definition of sip_pvt - Rename "text" to "codecs" - beacuse
it's what it is - Add documentation for future developers so that
we make sure that we define new sdp media types for SRTP-variants
2009-09-07 17:15 +0000 [r216846] Tilghman Lesher <tlesher@digium.com>
* configs/func_odbc.conf.sample, funcs/func_odbc.c, CHANGES: Allow
multiple rows to be fetched within the normal mode of operation.
2009-09-07 16:35 +0000 [r216652-216842] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: Make sure we reset global_exclude_static at
channel reload
* channels/chan_sip.c: Move capability into sip_cfg. While at it,
make sure we reset it at channel reload.
* channels/chan_sip.c: Move global_regcontext into the sip_cfg
structure
* channels/chan_sip.c: Move contact_ha to sip_cfg structure
* channels/chan_sip.c: Doxygen updates
* channels/chan_sip.c: Since it's possible to have more than 999
calls, I'm changing the call counter roof to something higher.
* channels/chan_sip.c: add doxygen and remove duplicate declaration
of variable
* channels/chan_sip.c: After many years, remove VOCAL_DATA_HACK
definition
* channels/chan_sip.c: Remove unneeded header files (tested on
Linux and OS/X)
* channels/chan_sip.c: Don't send MESSAGE with sendtext() if
recepient doesn't allow MESSAGE requests
* channels/chan_sip.c: Add some doxygen
* channels/chan_sip.c: Fix typo
* channels/chan_sip.c: If there is no session timer in the INVITE,
set it to default value (not unset minimum = -1) Patch by oej
closes issue #15621 Reported by: fnordian Tested by: atis
* configs/sip.conf.sample: Update sip.conf.sample documentation,
reorganize a bit
* channels/chan_sip.c: Simplify the code in this function
2009-09-04 19:32 +0000 [r216594] David Vossel <dvossel@digium.com>
* channels/chan_sip.c: sip peer matching by address only with
TCP/TLS This patch removes the contact header matching logic and
adds logic to match all tcp/tls connections by ip only. Thanks to
oej for finding the issue and suggesting solutions. Review:
https://reviewboard.asterisk.org/r/354/
2009-09-04 19:29 +0000 [r216593] Sean Bright <sean@malleable.com>
* apps/app_voicemail.c: Use ast_free() instead of free().
2009-09-04 17:50 +0000 [r216547-216551] Tilghman Lesher <tlesher@digium.com>
* include/asterisk/lock.h: Fix trunk breakage.
* main/pbx.c, UPGRADE-1.6.txt: Enable turning off the application
delimiter warning with the 'dontwarn' option. Suggested on the
-dev list, and implemented in an alternate way by me.
2009-09-04 15:05 +0000 [r216506] Michiel van Baak <michiel@vanbaak.info>
* /, main/utils.c: Merged revisions 216435 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r216435 | mvanbaak | 2009-09-04 15:56:10 +0200 (Fri, 04 Sep 2009)
| 2 lines make asterisk compile under devmode with DEBUG_THREADS
enabled on OpenBSD ........
2009-09-04 14:02 +0000 [r216438] Olle Johansson <oej@edvina.net>
* main/pbx.c, /, channels/chan_sip.c, apps/app_disa.c,
configs/sip.conf.sample, apps/app_playback.c: Merged revisions
216430 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r216430 | oej | 2009-09-04 15:45:48 +0200 (Fre, 04 Sep 2009) | 27
lines Make apps send PROGRESS control frame for early media and
fix too early media issue in SIP The issue at hand is that some
legacy (dying) PBX systems send empty media frames on PRI links
*before* any call progress. The SIP channel receives these frames
and by default signals 183 Session progress and starts sending
media. This will cause phones to play silence and ignore the
later 180 ringing message. A bad user experience. The fix is
twofold: - We discovered that asterisk apps that support early
media ("noanswer") did not send any PROGRESS frame to indicate
early media. Fixed. - We introduce a setting in chan_sip so that
users can disable any relay of media frames before the outbound
channel actually indicates any sort of call progress. In 1.4,
1.6.0 and 1.6.1, this will be disabled for backward
compatibility. In later versions of Asterisk, this will be
enabled. We don't assume that it will change your Asterisk phone
experience - only for the better. We encourage third-party
application developers to make sure that if they have
applications that wants to send early media, add a PROGRESS
control frame transmission to make sure that all channel drivers
actually will start sending early media. This has not been the
default in Asterisk previous to this patch, so if you got
inspiration from our code, you need to update accordingly. Sorry
for the trouble and thanks for your support. This code has been
running for a few months in a large scale installation (over 250
servers with PRI and/or BRI links to old PBX systems). That's no
proof that this is an excellent patch, but, well, it's tested :-)
........
2009-09-04 14:00 +0000 [r216431-216437] Michiel van Baak <michiel@vanbaak.info>
* include/asterisk/lock.h: make sure canlog is set so we can
compile with DEBUG_THREADS enabled on OpenBSD
* /: Recorded merge of revisions 216432 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r216432 | mvanbaak | 2009-09-04 15:53:09 +0200 (Fri, 04 Sep 2009)
| 2 lines make chan_sip compile under devmode again ........
* /: Recorded merge of revisions 216369 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r216369 | mvanbaak | 2009-09-04 15:16:29 +0200 (Fri, 04 Sep 2009)
| 4 lines Make sure 'start' is always initialized. This is the
same as rev 216222 in trunk but 1.4 is affected as well ........
2009-09-04 13:14 +0000 [r216368] Russell Bryant <russell@digium.com>
* channels/chan_sip.c: Do not treat every SIP peer as if they were
configured with insecure=port. There was a problem in the
function responsible for doing peer matching by IP address and
port number such that during the second pass for checking for a
peer configured with insecure=port, it would end up treating
every peer as if it had been configured that way. These changes
fix the logic in the peer IP and port comparison callback to
handle insecure=port checking properly. This problem was
introduced when SIP peers were converted to astobj2. Many thanks
to dvossel for noticing this while working on another peer
matching issue.
2009-09-04 12:05 +0000 [r216335] Olle Johansson <oej@edvina.net>
* doc/janitor-projects.txt: Adding to the janitor list. For new
readers: The janitor list is a list of tasks we need help with in
the Asterisk project. Taking up one of these is often a good way
to get into Asterisk development and getting a lot of developers
in the project to be grateful. It's stuff we could spend time on
when the bug tracker is empty, when our employers hasn't filled
our task lists and our servers is running bugfree and happily
without any issues. If you want to start working on one of these
small projects, feel free to ask for help in the #asterisk-dev
channel on IRC or asterisk-dev mailing list. We'll be more than
happy to help you to start and reach goal. Thank you for your
help. </end of long commit message>
2009-09-04 10:48 +0000 [r216264] Russell Bryant <russell@digium.com>
* /, doc/IAX2-security.txt (added): Merged revisions 216263 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
................ r216263 | russell | 2009-09-04 05:48:00 -0500
(Fri, 04 Sep 2009) | 9 lines Merged revisions 216262 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.2
........ r216262 | russell | 2009-09-04 05:47:37 -0500 (Fri, 04
Sep 2009) | 2 lines Add a plain text version of the IAX2 security
document. ........ ................
2009-09-04 06:08 +0000 [r216222] Michiel van Baak <michiel@vanbaak.info>
* main/astobj2.c: make sure 'start' is always initialized. Makes
asterisk compile with --enable-dev-mode
2009-09-03 21:09 +0000 [r216186] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c, channels/sig_pri.c: Lets try not to use
C++ keywords for variable names.
2009-09-03 19:40 +0000 [r216094] Doug Bailey <dbailey@digium.com>
* include/asterisk/callerid.h, channels/chan_dahdi.c,
channels/sig_analog.c, channels/sig_analog.h: Added detection
DTMF CID without polarity change alert. Added detection of DTMF
tone energy levels on FXO channels in chan_dahdi monitoring loop
so DTMF CID can be detected without the need of a polarity change
precursor. (closes issue #9096) Reported by: fleed Patches:
9096-chan_dahdi-trunk.diff uploaded by dbailey (license 819)
Tested by: cyberplant, sum, maturs
2009-09-03 19:38 +0000 [r216009-216092] Russell Bryant <russell@digium.com>
* /, UPGRADE.txt: Merged revisions 216085 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
................ r216085 | russell | 2009-09-03 14:36:46 -0500
(Thu, 03 Sep 2009) | 9 lines Merged revisions 216080 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.2
........ r216080 | russell | 2009-09-03 14:35:23 -0500 (Thu, 03
Sep 2009) | 2 lines Add a note about IAX2 to UPGRADE.txt.
........ ................
* /, doc/IAX2-security.pdf (added): Merged revisions 216008 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
................ r216008 | russell | 2009-09-03 13:44:58 -0500
(Thu, 03 Sep 2009) | 9 lines Merged revisions 216005 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.2
........ r216005 | russell | 2009-09-03 13:42:24 -0500 (Thu, 03
Sep 2009) | 2 lines Add IAX2 security document related to
AST-2009-006. ........ ................
2009-09-03 18:42 +0000 [r216006] Kevin P. Fleming <kpfleming@digium.com>
* main/file.c, doc/lang/language-criteria.txt (added): Document
language prompt submission process. This patch adds a document
describing the language prompt submission process, licensing
terms and other issues related to that process. In addition, it
modifies the sound file searching process to support language
codes with any number of suffices (not limited to just "xx" or
"xx_YY"), so that prompts can be named with gender,
customer/company, etc. suffices as well. (closes issue #15771)
Reported by: jtodd Patches: language-criteria.txt uploaded by
jtodd
2009-09-03 16:31 +0000 [r215955] David Vossel <dvossel@digium.com>
* configs/iax.conf.sample, include/asterisk/acl.h,
channels/iax2-parser.h, include/asterisk/astobj2.h,
channels/iax2.h, main/acl.c, channels/chan_iax2.c,
channels/iax2-parser.c, main/astobj2.c: Merge code associated
with AST-2009-006 (closes issue #12912) Reported by: rathaus
Tested by: tilghman, russell, dvossel, dbrooks
2009-09-03 13:02 +0000 [r215891] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: Add known internal IP address when
autodomain=yes (closes issue #14573) Reported by: pj Patches:
sip-internip-autodomain1.diff uploaded by mnicholson (license 96)
modified by oej Tested by: pj
2009-09-03 05:57 +0000 [r215838] Michiel van Baak <michiel@vanbaak.info>
* doc/manager_1_1.txt: Document that SIPshowpeer and SKINNYshowline
now include the configured parkinglot in their response. Prodded
by snuff-work on #asterisk-dev IRC channel
2009-09-03 03:43 +0000 [r215800-215801] Tilghman Lesher <tlesher@digium.com>
* channels/chan_sip.c: Default the callback extension to "s". This
is a regression. (closes issue #15764) Reported by: elguero
Change-type: bugfix
* include/asterisk.h: Revert attempt to standardize with
_POSIX_C_SOURCE. This did not function in the way that was
intended, causing more compatibility issues than it solved. It is
best, therefore, that it be simply removed. (Discussed with
kpfleming; agreement to remove was reached.)
2009-09-02 23:31 +0000 [r215758] Terry Wilson <twilson@digium.com>
* /, channels/chan_sip.c: Merged revisions 215682 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r215682 | twilson | 2009-09-02 16:41:22 -0500 (Wed, 02 Sep 2009)
| 18 lines Re-send non-100 provisional responses to prevent
cancellation From section 13.3.1.1 of RFC 3261: If the UAS
desires an extended period of time to answer the INVITE, it will
need to ask for an "extension" in order to prevent proxies from
canceling the transaction. A proxy has the option of canceling a
transaction when there is a gap of 3 minutes between responses in
a transaction. To prevent cancellation, the UAS MUST send a
non-100 provisional response at every minute, to handle the
possibility of lost provisional responses. (closes issue #11157)
Reported by: rjain Tested by: twilson Review:
https://reviewboard.asterisk.org/r/315/ ........
2009-09-02 23:25 +0000 [r215757] Richard Mudgett <rmudgett@digium.com>
* channels/sig_pri.h, channels/chan_dahdi.c,
configs/chan_dahdi.conf.sample, CHANGES, channels/sig_pri.c: Made
chan_dahdi able to ignore incoming calls that are not in a MSN
list for ISDN PTMP CPE spans.
2009-09-02 21:39 +0000 [r215681] David Vossel <dvossel@digium.com>
* channels/chan_sip.c: port string to int conversion using sscanf
There are several instances where a port is parsed from a uri or
some other source and converted to an int value using atoi(), if
for some reason the port string is empty, then a standard port is
used. This logic is used over and over, so I created a function
to handle it in a safer way using sscanf().
2009-09-02 21:23 +0000 [r215622-215665] Michiel van Baak <michiel@vanbaak.info>
* channels/chan_sip.c, channels/chan_skinny.c: add Parkinglot info
to sip show peer <foo> and skinny show line <foo> If we had this
from the start, debugging the 'parking not using configured
parkinglot' bug would have been easier.
* main/features.c: - lock channel before looking for a channel
variable - Init the parkings list member of struct parkinglot.
Thanks Sean for the explanation why this should be here.
2009-09-02 19:49 +0000 [r215608] Doug Bailey <dbailey@digium.com>
* channels/chan_dahdi.c, channels/sig_analog.c: Fix issue where
DTMF CID detect was placing channels into signed linear mode made
analog_set_linear_mode return back the mode that was being
overwritten so it could be restored later.
2009-09-02 18:37 +0000 [r215567] Tilghman Lesher <tlesher@digium.com>
* main/Makefile, main/app.c: Close up to the soft open file limit
(same on Linux, but varies drastically on OS X). Also, a Makefile
fix for Darwin (OS X). (closes issue #14542) Reported by: jtodd
Patches: 20090901__issue14542.diff.txt uploaded by tilghman
(license 14) Tested by: jtodd, tilghman Change-type: bugfix
2009-09-02 17:26 +0000 [r215522] David Vossel <dvossel@digium.com>
* channels/chan_sip.c: SIP uri parsing cleanup Now, the scheme
passed to parse_uri can either be a single scheme, or a list of
schemes ',' delimited. This gets rid of the whole problem of
having to create two buffers and calling parse_uri twice to check
for separate schemes. Review:
https://reviewboard.asterisk.org/r/343/
2009-09-02 16:20 +0000 [r215479] Michiel van Baak <michiel@vanbaak.info>
* channels/chan_skinny.c: like in chan_sip's sip_new skinny should
copy the configured parkinglot from a line to the newly created
channel. This makes callparking honor the configured parkinglot
for skinny lines as well.
2009-09-02 16:08 +0000 [r215466] David Vossel <dvossel@digium.com>
* channels/chan_sip.c: SIP support for keep-alive event keep-alive
events are used by Sipura/Linksys for NAT keepalive. There
currently don't appear to be any problems with NAT, but everytime
a keep-alive event is received, Asterisk responds with a "489 Bad
event". This error may indicate to a user that NAT problems exist
just because this even is not supported. Now, rather than respond
with an error, the packet is consumed and a "200 ok" is sent just
to indicate we received the packet. (issue #15084) Patches:
chan_sip.keepalive.v1.diff uploaded by IgorG (license 20)
2009-09-02 15:56 +0000 [r215419-215462] Michiel van Baak <michiel@vanbaak.info>
* channels/chan_sip.c: Honor configured parkinglot when parking and
retrieving parked calls Thank oej for pointing out the fact that
sip_new did not copy parkinglot from the peer into the newly
created channel. (closes issue #15538) Reported by: gracedman
Patches: 2009090100_sipnewparkinglot-161.diff.txt uploaded by
mvanbaak (license 7) With mod by me to also fix callparking as
well (this uploaded patch only fixed retrieving a parked call)
Tested by: gracedman, mvanbaak
* include/asterisk.h: Let's compile again on OpenBSD
2009-09-02 06:23 +0000 [r215382] Olle Johansson <oej@edvina.net>
* CHANGES, res/res_mutestream.c (added): Adding MUTEAUDIO()
dialplan function and MuteAudio AMI action (pinepeach) Review:
https://reviewboard.asterisk.org/r/345/
2009-09-02 01:16 +0000 [r215338] Dwayne M. Hubbard <dwayne.hubbard@gmail.com>
* /, apps/app_softhangup.c: Merged revisions 215270 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r215270 | dhubbard | 2009-09-01 18:04:52 -0500 (Tue, 01
Sep 2009) | 12 lines Use strrchr() so SoftHangup will correctly
truncate multi-hyphen channel names In general channel names are
in the form Foo/Bar-Z, but the channel name could have multiple
hyphens and look like Foo/B-a-r-Z. Use strrchr to truncate the
channel name at the last hyphen. (closes issue #15810) Reported
by: dhubbard Patches: dw-softhangup-1.4.patch uploaded by
dhubbard (license 733) ........
2009-09-01 23:41 +0000 [r215222-215301] Tilghman Lesher <tlesher@digium.com>
* channels/chan_sip.c, funcs/func_channel.c, CHANGES: Add
MASTER_CHANNEL() dialplan function, as well as a useful usage.
(closes issue #13140) Reported by: cpina Patches:
20090807__issue13140.diff.txt uploaded by tilghman (license 14)
Tested by: lmadsen Change-type: feature
* channels/chan_sip.c: Fix register such that lines with a
transport string, but without an authuser, parse correctly.
(AST-228)
2009-09-01 20:44 +0000 [r215212] Russell Bryant <russell@digium.com>
* addons/format_mp3.c: Fix memory corruption caused by format_mp3.
format_mp3 claimed that it provided AST_FRIENDLY_OFFSET in frames
returned by read(). However, it lied. This means that other parts
of the code that attempted to make use of the offset buffer would
end up corrupting the fields in the ast_filestream structure.
This resulted in quite a few crashes due to unexpected values for
fields in ast_filestream. This patch closes out quite a few bugs.
However, some of these bugs have been open for a while and have
been an area where more than one bug has been discussed. So with
that said, anyone that is following one of the issues closed
here, if you still have a problem, please open a new bug report
for the specific problem you are still having. If you do, please
ensure that the bug report is based on the newest version of
Asterisk, and that this patch is applied if format_mp3 is in use.
Thanks! (closes issue #15109) Reported by: jvandal Tested by:
aragon, russell, zerohalo, marhbere, rgj (closes issue #14958)
Reported by: aragon (closes issue #15123) Reported by:
axisinternet (closes issue #15041) Reported by: maxnuv (closes
issue #15396) Reported by: aragon (closes issue #15195) Reported
by: amorsen Tested by: amorsen (closes issue #15781) Reported by:
jensvb (closes issue #15735) Reported by: thom4fun (closes issue
#15460) Reported by: marhbere
2009-09-01 19:50 +0000 [r215161] Kevin P. Fleming <kpfleming@digium.com>
* main/frame.c: Ensure that frame dumps of
AST_CONTROL_T38_PARAMETERS frames are properly decoded.
2009-09-01 14:40 +0000 [r215110] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: Removing whitespace that causes red dots in
reviewboard
2009-08-31 22:02 +0000 [r215069-215070] Tilghman Lesher <tlesher@digium.com>
* main/http.c: Fix a trunk compilation warning.
* main/manager.c: Properly initialize the session to prevent a
crash. (closes issue #15774) Reported by: lasko Patches:
20090831__issue15774.diff.txt uploaded by tilghman (license 14)
Tested by: lasko
2009-08-31 18:17 +0000 [r215023] Olle Johansson <oej@edvina.net>
* funcs/func_volume.c: By copying this code I got bad comments in
reviewboard... Better fix the original.
2009-08-31 16:18 +0000 [r214819-214945] Tilghman Lesher <tlesher@digium.com>
* channels/chan_local.c, /: Merged revisions 214940 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r214940 | tilghman | 2009-08-31 11:16:52 -0500 (Mon, 31
Aug 2009) | 7 lines Also unlock the "other" channel, when
returning, due to glare. (closes issue #15787) Reported by:
tim_ringenbach Patches: chan_local.diff uploaded by tim
ringenbach (license 540) Tested by: tim_ringenbach ........
* Makefile: Force Darwin on ppc platforms to compile with a target
level that supports aliasing.
* include/asterisk.h, main/poll.c: Various patches, to enable
Asterisk to once again compile on Mac OS X. One note on defining
_POSIX_C_SOURCE: while this feature test macro works to require
certain behaviors on Linux, it works differently on *BSD
platforms to REMOVE certain API calls that are not in the POSIX
specification, such as vasprintf(3). Thus, defining it while
depending upon vasprintf (and other extensions to the POSIX
standard) to be defined is a recipe to ensure that Asterisk is
only buildable on Linux. Hence, this define which was meant to
INCREASE portability, effectively ensures the opposite.
* configure, include/asterisk/autoconfig.h.in, configure.ac,
pbx/pbx_lua.c: If lua is detected with the lua5.1 prefix (or
not), adjust the include path accordingly. Based upon feedback to
a release announcement on the -users list. See
http://lists.digium.com/pipermail/asterisk-users/2009-August/236954.html
2009-08-28 22:44 +0000 [r214777] Russell Bryant <russell@digium.com>
* configure: Update configure script so that CONFIG_CFLAGS and
CONFIG_LDFLAGS doesn't break the build.
2009-08-28 20:14 +0000 [r214702] Tilghman Lesher <tlesher@digium.com>
* main/channel.c, /: Merged revisions 214701 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r214701 | tilghman | 2009-08-28 15:13:32 -0500 (Fri, 28 Aug 2009)
| 8 lines Modify comment to be a bit more accurate. We have kept
this comment around long enough, that it's pretty clear that
we're keeping the code, because changing the code would require a
pretty fundamental architectural shift. We've also taken
criticism in some quarters, because it was believed that it was
referring to the code being nasty. No, the code isn't nasty, just
the operation itself is rather odd. Fixed for eternity (probably
not). ........
2009-08-28 20:01 +0000 [r214696] Kevin P. Fleming <kpfleming@digium.com>
* Makefile, include/asterisk/autoconfig.h.in, configure.ac,
makeopts.in: Ensure that CFLAGS and/or LDFLAGS provided to
configure script are preserved. Cross-compilation environments
want to provide 'defaults' for compiler and linker options, and
frequently do this by specifying CFLAGS and LDFLAGS in the
environment or as command-line arguments to the configure script.
This patch modifies the configure script and Makefile to preserve
these settings and ensure they are used in the build process.
2009-08-28 19:13 +0000 [r214654] Richard Mudgett <rmudgett@digium.com>
* channels/sig_pri.c: Move discardremoteholdretrieval test so it
applies only to the specific notification indicator values.
2009-08-28 18:41 +0000 [r214650] Mark Michelson <mmichelson@digium.com>
* include/asterisk/sched.h: Fix some incorrect documentation of
sched_thread functions.
2009-08-28 16:50 +0000 [r214360-214611] Tilghman Lesher <tlesher@digium.com>
* res/res_musiconhold.c: Remove unnecessary define for Solaris
(closes issue #15358) Reported by: snuffy Patches:
bug_res_moh_remove_unneeded_include.diff uploaded by snuffy
(license 35)
* /, configure, include/asterisk/autoconfig.h.in, configure.ac,
autoconf/libcurl.m4 (added): Merged revisions 214517 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r214517 | tilghman | 2009-08-27 16:45:34 -0500 (Thu, 27
Aug 2009) | 7 lines Use autoconf to detect libcurl, as this
enables cross-compilation checks, something we didn't allow
before. (closes issue #15714) Reported by: pprindeville Patches:
20090813__issue15714.diff.txt uploaded by tilghman (license 14)
Tested by: pprindeville ........
* main/manager.c: Ensure that we check for the special value
CONFIG_STATUS_FILEINVALID. (closes issue #15786) Reported by:
a_villacis Patches:
asterisk-1.6.2.0-beta4-manager-fix-crash-on-include-nonexistent-file.patch
uploaded by a villacis (license 660) (Plus a few of my own, to
catch the remaining places within manager.c where it could have
been a problem)
* /, configure, include/asterisk/autoconfig.h.in, configure.ac,
autoconf/ast_ext_lib.m4: Merged revisions 214436 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r214436 | tilghman | 2009-08-27 11:53:58 -0500 (Thu, 27
Aug 2009) | 2 lines One more build system change, to make the
descriptions look better, if we have better information. ........
* /, configure, include/asterisk/autoconfig.h.in,
autoconf/ast_ext_lib.m4: Merged revisions 214357 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r214357 | tilghman | 2009-08-27 11:03:50 -0500 (Thu, 27
Aug 2009) | 3 lines Make autoheader descriptions render correctly
in our autoconfig.h file. (Figured out while working with issue
#14906) ........
2009-08-27 15:57 +0000 [r214309-214355] Jeff Peeler <jpeeler@digium.com>
* doc/tex/channelvariables.tex: Add forgotten documentation for new
channel variables added in 214309.
* main/features.c, CHANGES: Add two new dialplan variables when
using features Added DYNAMIC_FEATURENAME which holds the last
triggered dynamic feature. Added DYNAMIC_PEERNAME which holds the
unique channel name on the other side and is set when a dynamic
feature is triggered. (closes issue #14663) Reported by: tamiel
Patches: 20090313_features.diff uploaded by tamiel (license 712)
Tested by: tamiel
2009-08-26 21:56 +0000 [r214272] Richard Mudgett <rmudgett@digium.com>
* configs/chan_dahdi.conf.sample: Minor punctuation change.
2009-08-26 16:53 +0000 [r214199] Tilghman Lesher <tlesher@digium.com>
* channels/chan_sip.c: Typo fix ("SIP/2.0 XXX" is 11 chars, not 10)
(closes issue #15362) Reported by: klaus3000 Patches:
chan_sip.c_logmessagefix_patch.txt uploaded by klaus3000 (license
65)
2009-08-26 16:38 +0000 [r214195] David Vossel <dvossel@digium.com>
* main/channel.c, /: Merged revisions 214194 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r214194 | dvossel | 2009-08-26 11:36:42 -0500 (Wed, 26 Aug 2009)
| 19 lines ast_write() ignores ast_audiohook_write() results In
ast_write(), if a channel has a list of audiohooks, those lists
are written to and the resulting frame is what ast_write() should
continue with. The problem was the returned audiohook frame was
not being handled at all, and the original frame passed into it
did not contain the mixed audio, so essentially audio was being
lost. One result of this was chan_spy's whisper mode no longer
worked. To complicate the issue, frames passed into ast_write may
either be a single frame, or a list of frames. So, as the list of
frames is processed in the audiohook_write, the returned frames
had to be added to a new list. (closes issue #15660) Reported by:
corruptor Tested by: dvossel ........
2009-08-25 22:39 +0000 [r213900-214152] Tilghman Lesher <tlesher@digium.com>
* configure, include/asterisk/autoconfig.h.in, configure.ac: Not
all versions of gnu-linux use glibc, which contains iconv. Some
(especially embedded systems) don't have iconv at all. (closes
issue #15169) Reported by: pprindeville
* /, main/say.c: Merged revisions 214068-214069 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r214068 | tilghman | 2009-08-25 14:26:50 -0500 (Tue, 25 Aug 2009)
| 6 lines Fix pronunciation of German dates. (closes issue
#15273) Reported by: Benjamin Kluck Patches: say_c.patch uploaded
by Benjamin Kluck (license 803) ........ r214069 | tilghman |
2009-08-25 14:28:42 -0500 (Tue, 25 Aug 2009) | 2 lines I should
always compile before committing... ........
* pbx/pbx_dundi.c: DUNDILOOKUP function in 1.6 should use comma
delimiters. (closes issue #15322) Reported by: chappell Patches:
dundilookup-0015322.patch uploaded by chappell (license 8)
* main/pbx.c, /: Merged revisions 213970 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r213970 | tilghman | 2009-08-25 01:34:44 -0500 (Tue, 25 Aug 2009)
| 7 lines Improve error message by informing user exactly which
function is missing a parethesis. (closes issue #15242) Reported
by: Nick_Lewis Patches: pbx.c-funcparenthesis.patch2 uploaded by
dbrooks (license 790) pbx.c-funcparenthesis-1.4.diff uploaded by
loloski (license 68) ........
* Makefile: The DTD should be installed in the same path as the
rest of the XML documentation. (closes issue #15344) Reported by:
tzafrir Patches: makefile_appdocs_dtd.diff uploaded by tzafrir
(license 46)
* Makefile, /: Merged revisions 213899 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r213899 | tilghman | 2009-08-24 21:40:22 -0500 (Mon, 24 Aug 2009)
| 4 lines Use the default runlevels for Debian derivatives,
instead of making up our own. (closes issue #14730) Reported by:
pkempgen ........
2009-08-24 16:43 +0000 [r213833] Jeff Peeler <jpeeler@digium.com>
* apps/app_voicemail.c: Fix storage of greetings when using
IMAP_STORAGE The store macro was not getting called preventing
storage of IMAP greetings at all. This has been corrected along
with fixing checking if the imapgreetings option is turned on to
store the greeting in IMAP. Lastly, the attachment filename was
incorrectly using the full path instead of just the basename,
which was causing problems with retrieval of the greeting.
(closes issue #14950) Reported by: noahisaac (closes issue
#15729) Reported by: lmadsen
2009-08-24 04:46 +0000 [r213790] Moises Silva <moises.silva@gmail.com>
* channels/chan_dahdi.c: improve handling of
openr2_chan_disconnect_call API failure, unlikely, but happened
on openr2 library bug
2009-08-21 23:18 +0000 [r213748] Richard Mudgett <rmudgett@digium.com>
* configure, configure.ac, channels/sig_pri.c: Update configure
script for libpri COLP feature dependency requirements.
2009-08-21 22:36 +0000 [r213738] Tilghman Lesher <tlesher@digium.com>
* channels/chan_sip.c: Clarifying comments in sip_register, and
removing a dead section
2009-08-21 22:22 +0000 [r213716] David Vossel <dvossel@digium.com>
* channels/chan_sip.c: Register request line contains wrong address
when user domain and register host differ (closes issue #15539)
Reported by: Nick_Lewis Patches: chan_sip.c-registraraddr.patch
uploaded by Nick (license 657) register_domain_fix_1.6.2 uploaded
by dvossel (license 671) Tested by: Nick_Lewis, dvossel
2009-08-21 21:39 +0000 [r213697] Kevin P. Fleming <kpfleming@digium.com>
* apps/app_voicemail.c: Ensure that realtime mailboxes properly
report status on subscription. This patch modifies
app_voicemail's response to mailbox status subscriptions (via the
internal event system) to ensure that a subscription triggers an
explicit poll of the mailbox, so the subscriber can get an
immediate cached event with that status. Previously, the cache
was only populated with the status of non-realtime mailboxes.
(closes issue #15717) Reported by: natmlt
2009-08-21 21:02 +0000 [r213635] David Vossel <dvossel@digium.com>
* channels/chan_sip.c: fixes sip register parsing when user@domain
is used (issue #15008) (issue #15672)
2009-08-21 16:53 +0000 [r213560] Tilghman Lesher <tlesher@digium.com>
* include/asterisk.h, /: Merged revisions 213559 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r213559 | tilghman | 2009-08-21 11:52:53 -0500 (Fri, 21 Aug 2009)
| 7 lines Permit DEBUG_FD_LEAKS to be used with C++ source files.
(closes issue #15698) Reported by: slavon Patches:
20090817__issue15698.diff.txt uploaded by tilghman (license 14)
Tested by: slavon, tilghman ........
2009-08-21 16:04 +0000 [r213494] Jason Parker <jparker@digium.com>
* /, configs/queues.conf.sample: Merged revisions 213493 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r213493 | qwell | 2009-08-21 11:03:21 -0500 (Fri, 21 Aug 2009) |
5 lines Clarify queues.conf comments to specify that variables
should be set in the dialplan. (closes issue #15755) Reported by:
trendboy ........
2009-08-21 04:09 +0000 [r213454] Moises Silva <moises.silva@gmail.com>
* channels/chan_dahdi.c: increment the mfcr2 monitor count when
clearing the call request
2009-08-21 03:48 +0000 [r213450] Terry Wilson <twilson@digium.com>
* main/loader.c: Make LOAD_ORDER actually work
2009-08-20 22:13 +0000 [r213414] Tilghman Lesher <tlesher@digium.com>
* apps/app_queue.c: Add original position, when logging a caller
entering a queue. (closes issue #15146) Reported by: arabe
Patches: asterisk-trunk.patch uploaded by arabe (license 786)
2009-08-20 21:33 +0000 [r213404] Jeff Peeler <jpeeler@digium.com>
* apps/app_voicemail.c: Fix greeting retrieval from IMAP Properly
check for the current voicemail state and if it doesn't exist,
create it. (closes issue #14597) Reported by: wtca Patches:
14597_v2.patch uploaded by mmichelson (license 60) Tested by:
jpeeler
2009-08-20 20:29 +0000 [r213327] Matthew Nicholson <mnicholson@digium.com>
* main/features.c: Fix a crash by checking the proper pointer for
validity before deferencing it. (closes issue #15751) Reported
by: atis Patches: ast_bridge_call_peer_cdr.patch uploaded by atis
(license 242)
2009-08-20 19:56 +0000 [r213284] Jeff Peeler <jpeeler@digium.com>
* apps/app_voicemail.exports (added), /: Merged revisions 213283
via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r213283 | jpeeler | 2009-08-20 14:53:34 -0500 (Thu, 20 Aug 2009)
| 2 lines Make all the symbols for the C-client callbacks global
........
2009-08-20 15:29 +0000 [r213248] Tilghman Lesher <tlesher@digium.com>
* addons/res_config_mysql.c: Select uncommented lines, not
commented ones. (closes issue #15746) Reported by: makoto
2009-08-20 03:26 +0000 [r213216] Moises Silva <moises.silva@gmail.com>
* channels/chan_dahdi.c: fixed bug caused by calling ast_request
without calling ast_call on an R2 channel, ie, CHANISAVAIL
2009-08-19 22:38 +0000 [r213179] Jason Parker <jparker@digium.com>
* main/ulaw.c, main/alaw.c: Fix compile when certain G711
menuselect options are enabled. (closes issue #15697) Reported
by: slavon
2009-08-19 21:21 +0000 [r213113] David Vossel <dvossel@digium.com>
* /, apps/app_mixmonitor.c: Merged revisions 213103 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r213103 | dvossel | 2009-08-19 16:18:37 -0500 (Wed, 19
Aug 2009) | 8 lines Fixes memory leak caused by incorrectly
freeing mixmonitor (closes issue #15699) Reported by: edantie
Patches: mixmonitor.patch uploaded by edantie (license 862)
........
2009-08-19 21:05 +0000 [r213093-213098] Tilghman Lesher <tlesher@digium.com>
* channels/chan_sip.c, configs/sip.conf.sample: Better parsing for
the "register" line Allows characters that are otherwise used as
delimiters to be used within certain fields (like the secret).
(closes issue #15008, closes issue #15672) Reported by: tilghman
Patches: 20090818__issue15008.diff.txt uploaded by tilghman
(license 14) Tested by: lmadsen, tilghman
* channels/chan_sip.c: If we have realtime caching enabled, 'sip
reload' must purge users/peers, even if the config files haven't
changed. (closes issue #12869) Reported by: bcnit Patches:
20090819__issue12869__2.diff.txt uploaded by tilghman (license
14) Tested by: lasko
2009-08-19 15:32 +0000 [r213046] Russell Bryant <russell@digium.com>
* main/features.c: Don't blow up on a NULL cdr. Reported in
#asterisk-dev.
2009-08-18 23:53 +0000 [r213007] Richard Mudgett <rmudgett@digium.com>
* channels/sig_pri.h, CHANGES, channels/sig_pri.c: Add COLP support
to chan_dahdi/sig_pri. Add Connected Line Presentation (COLP)
support to chan_dahdi/libpri as an addition to issue 8824. This
is the chan_dahdi/sig_pri portion. COLP support is now available
for any switch for which libpri supports COLP (currently ETSI
PTP, ETSI PTMP, and Q.SIG) with this patch. (closes issue #14068)
Tested by: rmudgett Review:
https://reviewboard.asterisk.org/r/340/
2009-08-18 20:33 +0000 [r212922-212939] Kevin P. Fleming <kpfleming@digium.com>
* /: Remove some accidentally-committed properties.
* CREDITS, /, UPGRADE-1.4.txt, sounds/sounds.xml,
build_tools/prep_tarball, sounds/Makefile, doc/tex/asterisk.tex:
Convert this branch to Opsound music-on-hold. For more details:
http://blogs.digium.com/2009/08/18/asterisk-music-on-hold-changes/
2009-08-18 19:49 +0000 [r212857-212883] Tilghman Lesher <tlesher@digium.com>
* addons/res_config_mysql.c: Clarify some of the error messages, to
help upgraders.
* configs/extconfig.conf.sample: Make the default extconfig.conf
match entries with the sample res_mysql.conf. This eliminates a
future source of possible confusion with the configuration of
1.6.1 and higher.
2009-08-18 18:57 +0000 [r212844] Olle Johansson <oej@edvina.net>
* apps/app_meetme.c: Small doxygen changes
2009-08-18 16:38 +0000 [r212764] Sean Bright <sean@malleable.com>
* main/manager.c, /: Merged revisions 212763 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r212763 | seanbright | 2009-08-18 12:36:00 -0400 (Tue, 18 Aug
2009) | 11 lines Delay the creation of temporary files until we
have a valid manager command to handle. Without this patch,
asterisk creates a temporary file before determining if the
specified command is valid. If invalid, we weren't properly
cleaning up the file. (closes issue #15730) Reported by: zmehmood
Patches: M15730.diff uploaded by junky (license 177) Tested by:
zmehmood ........
2009-08-18 16:29 +0000 [r212758] Richard Mudgett <rmudgett@digium.com>
* /, channels/misdn/isdn_lib.c: Merged revisions 212727 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r212727 | rmudgett | 2009-08-18 11:00:56 -0500 (Tue, 18 Aug 2009)
| 1 line Removed some deadwood and added some doxygen comments.
........
2009-08-17 20:40 +0000 [r212672] Kevin P. Fleming <kpfleming@digium.com>
* include/asterisk.h: Relax check for XOPEN_VERSION. It's not clear
that we actually require XOPEN_VERSION to be 600 or greater at
this time, so skip the check for now.
2009-08-17 19:57 +0000 [r212627] Tilghman Lesher <tlesher@digium.com>
* apps/app_voicemail.c: Check the return value of opendir(3), or we
may crash. (closes issue #15720) Reported by: tobias_e
2009-08-17 18:50 +0000 [r212574-212581] Sean Bright <sean@malleable.com>
* channels/chan_agent.c: Correct spelling of AGENTACCEPTDTMF in
chan_agent. (closes issue #15668) Reported by: davidw
* main/logger.c: Correct the return value check for
ast_safe_system. The logic here was reversed as ast_safe_system
returns -1 on error and not on success. Fix suggested by
reporter. (closes issue #15667) Reported by: loic
2009-08-17 16:50 +0000 [r212506] Jeff Peeler <jpeeler@digium.com>
* /, channels/misdn_config.c: Merged revisions 212498 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r212498 | jpeeler | 2009-08-17 11:34:56 -0500 (Mon, 17
Aug 2009) | 12 lines Fix segfault when reloading chan_misdn. If
more ports were specified than configured in misdn.conf a reload
would crash asterisk. The problem was the unconfigured port was
using data from the previously configured port. When the data for
an unconfigured port was freed a crash would result from the
double free. (closes issue #12113) Reported by: agupta Patches:
bug12113.patch uploaded by jpeeler (license 325) ........
2009-08-17 16:25 +0000 [r212463] Kevin P. Fleming <kpfleming@digium.com>
* include/asterisk.h, main/xml.c: Define our desires for POSIX and
X/OPEN API features properly. Based on a post on the gcc-help
mailing list and some subsequent reading, we can increase our
portability to various platforms by directly defining the POSIX
and X/OPEN API feature sets we wish to have available. This patch
does that, and also includes a double-check to ensure that the
system we are compiling on can actually provide the requested
feature sets.
2009-08-17 15:42 +0000 [r212431] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c, channels/sig_analog.c, /: Merged revisions
212430 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 Fix
uninitialized variable causing random MWI indications. (closes
issue #15727) Reported by: doda Patches: dahdi_changes.patch
uploaded by doda (license 853) ........ r212430 | rmudgett |
2009-08-17 10:36:28 -0500 (Mon, 17 Aug 2009) | 1 line Fix
uninitialized variable. ........
2009-08-16 19:27 +0000 [r212390] Joshua Colp <jcolp@digium.com>
* main/rtp_engine.c, include/asterisk/rtp_engine.h: Add two more
API calls for getting the current glue and channel in bridging
code.
2009-08-15 11:36 +0000 [r212339-212343] Michiel van Baak <michiel@vanbaak.info>
* res/res_calendar.c: cast time_t type variables to long where
needed. This makes res_calendar.c compile on OpenBSD and the same
cast is used in a lot of other places where time_t type vars are
used. (closes issue #15656) Reported by: mvanbaak Patches:
2009081100-rescalendarcompilefix.diff.txt uploaded by mvanbaak
(license 7)
* main/xmldoc.c: Add an empty line after each option when printing
the documentation of a function/application. This will make
reading the docs on the CLI way more easy. (closes issue #15694)
Reported by: mvanbaak Patches:
2009081100-extralinesoptionlist.diff.txt uploaded by mvanbaak
(license 7)
2009-08-14 23:07 +0000 [r212287-212291] Jeff Peeler <jpeeler@digium.com>
* channels/sig_analog.c: Add braces where missing and a few
whitespace fixes in sig_analog (closes issue #15678) Reported by:
alecdavis Patches: sig_analog_mainly_braces.diff.txt uploaded by
alecdavis (license 585) Tested by: alecdavis
* channels/chan_dahdi.c, channels/sig_analog.c,
channels/sig_analog.h: More code that somehow got left out of
sig_analog * confirmanswer option now respected * check and set
waiting for dialtone timer * unneeded needcallerid flag removed
from analog_subchannel * ss_astchan does not need to be a void
pointer * swap_channels callback updated to trunk * analog_hangup
now resets channel to default law
2009-08-14 17:36 +0000 [r212249] Tilghman Lesher <tlesher@digium.com>
* funcs/func_curl.c: Add SSL_VERIFYPEER, as requested on the -users
list
2009-08-13 17:33 +0000 [r212199] Richard Mudgett <rmudgett@digium.com>
* channels/chan_misdn.c: Send a generic return result when we
receive a CallDeflection facility message in chan_misdn. ETSI
300-196 implies that a facility return result without arguments
does not have the operation-value. This fact implies for ETSI
that you can only use the invoke-id to match requests with
responses.
2009-08-13 16:44 +0000 [r212161] Joshua Colp <jcolp@digium.com>
* main/rtp_engine.c, include/asterisk/rtp_engine.h: Add an API call
for retrieving the engine in use by an RTP instance.
2009-08-13 15:46 +0000 [r212113] Kevin P. Fleming <kpfleming@digium.com>
* channels/chan_sip.c: Ensure that T38FaxVersion is put into
outgoing SDP in the proper case.
2009-08-13 13:51 +0000 [r212067] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: Check an actual populated variable when
seeing if we need to do video or not.
2009-08-13 11:37 +0000 [r212027] Gavin Henry <ghenry@suretecsystems.com>
* contrib/scripts/asterisk.ldap-schema,
contrib/scripts/asterisk.ldif: Fixed typo (closes issue #15710)
Reported by: suretec
2009-08-12 23:14 +0000 [r211947-211957] Matthew Nicholson <mnicholson@digium.com>
* /, apps/app_queue.c: Merged revisions 211953 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r211953 | mnicholson | 2009-08-12 18:04:02 -0500 (Wed, 12 Aug
2009) | 10 lines This patch adds additional checking when
generating queue log TRANSFER events. The additional checks
prevent generation of false TRANSFER events in certain
situations. (closes issue #14536) Reported by: aragon Patches:
queue-log-xfer-fix1.diff uploaded by mnicholson (license 96)
Tested by: aragon, mnicholson ........
* channels/chan_sip.c, configs/sip.conf.sample: This patch adds
support for choosing a realm based on the domain in the From or
To header in the incoming request. Eligible domains are taken
from the domains list in the config file. This functionality is
enabled when domainsasrealm is enabled in the config file.
(closes issue #11361) Reported by: arkadia Patches:
sip_realm_mnich_to_added_2.patch uploaded by arkadia (license
233) Tested by: arkadia
2009-08-12 20:47 +0000 [r211908] Jeff Peeler <jpeeler@digium.com>
* channels/chan_dahdi.c, channels/sig_analog.c,
channels/sig_analog.h: Fix chan_dahdi option ringtimeout
dahdi_read relies on the dahdi_pvt copy of ringt which was not
getting set in sig_analog. This patch adds a callback to do so.
(closes issue #15288) Reported by: alecdavis Patches:
chan_dahdi.ringtimeout.diff.txt uploaded by alecdavis (license
585) Tested by: alecdavis
2009-08-12 19:53 +0000 [r211876] Matthew Nicholson <mnicholson@digium.com>
* channels/chan_sip.c: Make asterisk handle 423 Interval Too Short
messages better. This change uses separate values for the
acceptable minimum expiry provided by the 423 error and the
expiry value stored in the configuration file. Previously, the
value pulled from the configuration file would be overwritten.
(closes issue #14366) Reported by: Nick_Lewis Patches:
sip-expiry-fix1.diff uploaded by mnicholson (license 96)
chan_sip.c-reqexpiry.patch uploaded by Nick (license 657) Tested
by: mnicholson
2009-08-12 16:00 +0000 [r211767] Gavin Henry <ghenry@suretecsystems.com>
* res/res_config_ldap.c, contrib/scripts/asterisk.ldap-schema,
contrib/scripts/asterisk.ldif: Added three new attributes and
applied a patch to res_config_ldap.c attributetype (
AstAccountSubscribeContext NAME 'AstAccountSubscribeContext' DESC
'Asterisk subscribe context' EQUALITY caseIgnoreMatch SUBSTR
caseIgnoreSubstringsMatch SYNTAX 1.3.6.1.4.1.1466.115.121.1.15)
attributetype ( AstAccountIpAddr NAME 'AstAccountIpAddr' DESC
'Asterisk aaccount IP address' EQUALITY caseIgnoreMatch SUBSTR
caseIgnoreSubstringsMatch SYNTAX 1.3.6.1.4.1.1466.115.121.1.15)
attributetype ( AstAccountUserAgent NAME 'AstAccountUserAgent'
DESC 'Asterisk account user context' EQUALITY caseIgnoreMatch
SUBSTR caseIgnoreSubstringsMatch SYNTAX
1.3.6.1.4.1.1466.115.121.1.15) and patch
fix_empty_attributes_1.6.1.4_v2.patch (closes issue #13725)
Reported by: macogeek Patches:
fix_empty_attributes_1.6.1.4_v2.patch uploaded by xvisor (license
863) Tested by: suretec
2009-08-12 10:11 +0000 [r211732] Russell Bryant <russell@digium.com>
* channels/chan_jingle.c, channels/chan_unistim.c,
channels/chan_skinny.c, channels/chan_h323.c,
channels/chan_gtalk.c, channels/chan_mgcp.c: Always specify which
RTP engine is desired for a new RTP instance. This fixes a crash
reported in #asterisk-dev where chan_mgcp unexpectedly allocated
an RTP instance from res_rtp_multicast, since by not specifying
an engine, you get the first one in the list of engines.
2009-08-10 23:21 +0000 [r211675] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c: Encapsulate testing for which signaling
styles are used by sig_pri. Created the
dahdi_sig_pri_lib_handles() function and SIG_PRI_LIB_HANDLE_CASES
macro to simplify testing for which signaling styles are handled
by sig_pri.
2009-08-10 19:49 +0000 [r211539-211584] Tilghman Lesher <tlesher@digium.com>
* doc/CODING-GUIDELINES, /: Merged revisions 211583 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r211583 | tilghman | 2009-08-10 14:48:48 -0500 (Mon, 10
Aug 2009) | 1 line Conversion specifiers, not format specifiers
........
* cel/cel_pgsql.c, funcs/func_speex.c, funcs/func_rand.c,
apps/app_dahdibarge.c, main/frame.c, addons/chan_ooh323.c,
apps/app_readfile.c, /, apps/app_record.c,
apps/app_alarmreceiver.c, cdr/cdr_adaptive_odbc.c,
res/res_http_post.c, channels/chan_iax2.c, main/indications.c,
main/config.c, main/cli.c, pbx/pbx_loopback.c,
channels/chan_dahdi.c, pbx/pbx_spool.c, res/res_smdi.c,
channels/chan_skinny.c, main/features.c, main/http.c, main/pbx.c,
funcs/func_sprintf.c, funcs/func_timeout.c, apps/app_privacy.c,
codecs/codec_speex.c, channels/chan_agent.c, funcs/func_math.c,
apps/app_disa.c, apps/app_morsecode.c, channels/iax2-provision.c,
funcs/func_cut.c, apps/app_talkdetect.c, main/netsock.c,
res/res_config_curl.c, channels/chan_misdn.c,
apps/app_waitforring.c, funcs/func_channel.c, apps/app_macro.c,
addons/cdr_mysql.c, pbx/pbx_config.c, apps/app_mixmonitor.c,
apps/app_chanspy.c, main/asterisk.c, res/res_odbc.c,
cel/cel_adaptive_odbc.c, main/timing.c, apps/app_voicemail.c,
doc/CODING-GUIDELINES, addons/app_mysql.c, utils/muted.c,
apps/app_meetme.c, main/utils.c, res/res_musiconhold.c,
cdr/cdr_pgsql.c, apps/app_followme.c, res/res_config_sqlite.c,
main/enum.c, utils/frame.c, channels/misdn_config.c,
main/channel.c, res/ael/pval.c, main/cdr.c, funcs/func_enum.c,
channels/chan_phone.c, main/manager.c, apps/app_setcallerid.c,
apps/app_osplookup.c, funcs/func_odbc.c, res/res_agi.c,
apps/app_minivm.c, channels/xpmr/xpmr.c, res/res_config_ldap.c,
apps/app_rpt.c, channels/chan_mgcp.c, apps/app_adsiprog.c,
res/res_config_pgsql.c, funcs/func_dialplan.c, main/dnsmgr.c,
channels/chan_sip.c, res/res_limit.c, apps/app_waitforsilence.c,
agi/eagi-test.c, main/acl.c, apps/app_waituntil.c,
apps/app_originate.c, channels/sig_pri.c, apps/app_queue.c,
channels/chan_oss.c, agi/eagi-sphinx-test.c,
channels/chan_usbradio.c, res/snmp/agent.c, pbx/pbx_dundi.c,
apps/app_sms.c, utils/extconf.c, apps/app_stack.c,
apps/app_verbose.c, addons/app_saycountpl.c, main/dsp.c,
addons/res_config_mysql.c: AST-2009-005
2009-08-10 18:01 +0000 [r211475] Michiel van Baak <michiel@vanbaak.info>
* channels/chan_skinny.c: add manager events when a skinny device
registers/unregisters like we have in chan_sip (closes issue
#15499) Reported by: arifzaman Patches:
2009072600-skinnymanagerevents.diff.txt uploaded by mvanbaak
(license 7)
2009-08-10 17:17 +0000 [r211435] Jeff Peeler <jpeeler@digium.com>
* channels/chan_dahdi.c, channels/sig_pri.c: Fix PRI/BRI channels
when in alarm condition to only be marked for hangup if T309 is
not enabled.
2009-08-10 15:53 +0000 [r211392] Richard Mudgett <rmudgett@digium.com>
* channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_pri.c:
Restoring some code to sig_pri. Not sure if it is really needed.
Putting some DSP code back into sig_pri that was removed by the
chan_dahdi/sig_pri reorganization.
2009-08-10 15:46 +0000 [r211390] Russell Bryant <russell@digium.com>
* main/channel.c: Fix up some issues with getting a channel by
"name". Even though the get_channel_by_name() API advertised that
you could search by name or uniqueid (just as the old API did),
searching by uniqueid was not actually implemented. This patch
fixes that problem. The ast_channel_get_full() function now makes
a second search attempt by uniqueid if the parameter was a name.
The channel comparison function also now knows how to compare by
unqieueid. Finally, a bug was fixed in passing where OBJ_POINTER
was being passed in some scenarios where it should not have been.
2009-08-10 14:07 +0000 [r211347] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: Fix retrieval of the port used for the video
stream when adding SDP to a SIP message. (closes issue #15121)
Reported by: jsmith
2009-08-09 15:42 +0000 [r211232-211275] Tilghman Lesher <tlesher@digium.com>
* /, main/astfd.c: Merged revisions 211274 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r211274 | tilghman | 2009-08-09 10:41:01 -0500 (Sun, 09 Aug 2009)
| 2 lines Small oops. Clear the flags which have been checked.
........
* apps/app_stack.c: Check for NULL frame, before dereferencing
pointer. (closes issue #15617) Reported by: rain
2009-08-07 23:30 +0000 [r211191-211197] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c: Fixed some unsafe down cast pointer
operations for sig_pri. You cannot cast the struct
dahdi_pvt.sig_pvt pointer to a specific signaling private pointer
without first checking that it is in fact pointing to the correct
signaling private structure.
* channels/sig_pri.c: Fix static on line when PRI does overlap
dialing. The wrong encoding law was used because = was used when
it should have been ==.
2009-08-07 20:12 +0000 [r211113] Russell Bryant <russell@digium.com>
* /: Recorded merge of revisions 211112 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r211112 | russell | 2009-08-07 15:11:31 -0500 (Fri, 07 Aug 2009)
| 4 lines Resolve a deadlock involving app_chanspy and
masquerades. (ABE-1936) ........
2009-08-07 18:17 +0000 [r211040] Tilghman Lesher <tlesher@digium.com>
* /, apps/app_queue.c: Merged revisions 211038 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r211038 | tilghman | 2009-08-07 13:16:28 -0500 (Fri, 07 Aug 2009)
| 14 lines QUEUE_MEMBER_LIST _really_ wants the interface name,
not the membername. This is a partial revert of revision 82590,
which was an attempted cleanup, but in reality, it broke
QUEUE_MEMBER_LIST, which has always been intended as a method by
which component interfaces could be queried from the queue.
Membername isn't useful here, because that field cannot be used
to obtain further information about the member. See the
documentation on QUEUE_MEMBER_LIST, RemoveQueueMember,
QUEUE_MEMBER_PENALTY, and the various AMI commands which take a
member argument for further justification. (closes issue #15664)
Reported by: rain Patches: app_queue-queue_member_list.diff
uploaded by rain (license 327) ........
2009-08-07 13:08 +0000 [r210992] Kevin P. Fleming <kpfleming@digium.com>
* main/udptl.c: Workaround broken T.38 endpoints that offer tiny
MaxDatagram sizes. Some T.38 endpoints treat T38FaxMaxDatagram as
the maximum IFP size that should be sent to them, rather than the
maximum packet payload size. If such an endpoint also requests
UDPRedundancy as the error correction mode, we'll end up
calculating a tiny maximum IFP size, so small as to be unusable.
This patch sets a lower bound on what we'll consider the remote's
maximum IFP size to be, assuming that endpoints that do this
really can accept larger packets than they've offered to accept.
(closes issue #15649) Reported by: dazza76
2009-08-06 21:46 +0000 [r210908-210914] Tilghman Lesher <tlesher@digium.com>
* main/channel.c, /: Merged revisions 210913 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r210913 | tilghman | 2009-08-06 16:45:01 -0500 (Thu, 06 Aug 2009)
| 7 lines Because channel information can be accessed outside of
the channel thread, we must lock the channel prior to modifying
it. (closes issue #15397) Reported by: caspy Patches:
20090714__issue15397.diff.txt uploaded by tilghman (license 14)
Tested by: caspy ........
* include/asterisk/app.h, main/app.c, apps/app_stack.c: Allow Gosub
to recognize quote delimiters without consuming them. (closes
issue #15557) Reported by: rain Patches:
20090723__issue15557.diff.txt uploaded by tilghman (license 14)
Tested by: rain Review: https://reviewboard.asterisk.org/r/316/
2009-08-06 20:15 +0000 [r210866-210869] Richard Mudgett <rmudgett@digium.com>
* channels/sig_analog.c: Miscellaneous minor fixes to sig_analog. *
Sanity adjustments to __analog_ss_thread for sig_analog
environment. * Deleted some duplicated code. * Fixed
analog_ss_thread_start passing the wrong pointer.
* channels/sig_pri.c: Sanity adjustments to pri_ss_thread for
sig_pri environment.
2009-08-06 17:47 +0000 [r210817] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: Accept additional T.38 reinvites after an
initial one has been handled. Discussion of this subject has
yielded that it is not actually acceptable to change T.38
parameters after the initial reinvite but declining is harsh and
can cause the fax to fail when it may be possible to allow it to
continue. This patch changes things so that additional T.38
reinvites are accepted but parameter changes ignored. This gives
the fax a fighting chance. (closes issue #15610) Reported by:
huangtx2009
2009-08-06 16:07 +0000 [r210777] Kevin P. Fleming <kpfleming@digium.com>
* configure, include/asterisk/autoconfig.h.in, apps/app_fax.c,
configure.ac: Minor improvements to app_fax. This patch makes
some small changes to handle watchdog timeouts in a better way,
and also uses a 'cleaner' method of including the spandsp header
files. (closes issue #14769) Reported by: andrew Patches:
app_fax-20090406.diff uploaded by andrew (license 240)
v1-14769.patch uploaded by dimas (license 88) Tested by: freh,
deti, caspy, dimas, sgimeno, Dovid
2009-08-05 23:44 +0000 [r210640-210732] Richard Mudgett <rmudgett@digium.com>
* channels/sig_pri.c: Fix potential deadlock issue with
USERUSERINFO channel variable.
* channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_pri.c:
More changes from chan_dahdi that did not make it into sig_pri. *
Q.SIG channel mapping option. * discardremoteholdretrieval
option. * libPRI debug defines. * pri_set_overlapdial() now set
correctly. * pthread creation of pri_ss_thread now matches.
* /, channels/sig_pri.c: Merged revisions 210575 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r210575 | rmudgett | 2009-08-05 14:18:56 -0500 (Wed, 05 Aug 2009)
| 14 lines Dialplan starts execution before the channel setup is
complete. * Issue 15655: For the case where dialing is complete
for an incoming call, dahdi_new() was asked to start the PBX and
then the code set more channel variables. If the dialplan hungup
before these channel variables got set, asterisk would likely
crash. * Fixed potential for overlap incoming call to erroneously
set channel variables as global dialplan variables if the
ast_channel structure failed to get allocated. * Added missing
set of CALLINGSUBADDR in the dialing is complete case. (closes
issue #15655) Reported by: alecdavis ........
2009-08-05 18:49 +0000 [r210564] Leif Madsen <lmadsen@digium.com>
* doc/tex/imapstorage.tex, /: Merged revisions 210563 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r210563 | lmadsen | 2009-08-05 13:46:21 -0500 (Wed, 05
Aug 2009) | 11 lines Update imapstorage.txt documentation.
Updated the imapstorage.txt documentation to reflect that issues
with c-client versions older than 2007 seem to cause crashing
issues that are not seen with more recent versions. Documentation
has been updated to reflect this. (closes issue #14496) Reported
by: vbcrlfuser Patches: __20090727-imap-documentation-patch.txt
uploaded by lmadsen (license 10) Tested by: lmadsen, mmichelson,
dbrooks ........
2009-08-05 14:09 +0000 [r210522] Russell Bryant <russell@digium.com>
* main/file.c: Revert some silly code that snuck into trunk from my
working copy. Sorry!
2009-08-05 08:03 +0000 [r210478] Michiel van Baak <michiel@vanbaak.info>
* addons/mp3: ignore the .i files when compiling in 'DONT_OPTIMIZE'
in the addons/mp3 directory
2009-08-04 17:46 +0000 [r210353-210387] Richard Mudgett <rmudgett@digium.com>
* channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_pri.c:
Fix CALLERID() values for sig_pri on incoming calls.
* main/channel.c, include/asterisk/channel.h: Initial minimum
ast_party_caller support.
* channels/chan_dahdi.c: Removed some dead code.
2009-08-04 15:35 +0000 [r210302] Jeff Peeler <jpeeler@digium.com>
* main/features.c: Fix broken call pickup The find_channel_by_group
callback was only looking at the channel that was attempting to
make the pickup instead of the other channels in the container.
2009-08-04 14:53 +0000 [r210190-210238] Kevin P. Fleming <kpfleming@digium.com>
* Makefile, /: Merged revisions 210237 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r210237 | kpfleming | 2009-08-04 09:51:39 -0500 (Tue, 04 Aug
2009) | 10 lines Eliminate spurious compiler warnings from system
headers on *BSD platforms. Ensure that system headers located in
/usr/local/include are actually treated as system headers by the
compiler, and not as local headers which are subject to warnings
from the -Wundef compiler option and others. (closes issue
#15606) Reported by: mvanbaak ........
* contrib/scripts/realtime_pgsql.sql, channels/chan_sip.c,
channels/chan_skinny.c, configs/mgcp.conf.sample,
doc/res_config_sqlite.txt, doc/tex/phoneprov.tex, UPGRADE.txt,
configs/res_ldap.conf.sample, configs/sip.conf.sample,
configs/skinny.conf.sample, channels/chan_mgcp.c,
doc/chan_sip-perf-testing.txt: Rename 'canreinvite' option to
'directmedia', with backwards compatibility. It is clear from
multiple mailing list, forum, wiki and other sorts of posts that
users don't really understand the effects that the 'canreinvite'
config option actually has, and that in some cases they think
that setting it to 'no' will actually cause various other
features (T.38, MOH, etc.) to not work properly, when in fact
this is not the case. This patch changes the proper name of the
option to what it should have been from the beginning
('directmedia'), but preserves backwards compatibility for
existing configurations.
2009-08-03 18:05 +0000 [r210094-210154] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c, channels/sig_pri.c: Changes from
chan_dahdi that did not make it into sig_pri. * Moved
SUPPORT_USERUSER to sig_pri.c * Fix PRI_DEADLOCK_AVOIDANCE
parameter. * Whitespace changes. * Added missing unlock in
pri_dchannel():PRI_EVENT_RING case. * Balanced curly braces. *
ast_debug/ast_log changes from chan_dahdi. * sig_pri_indicate()
should default to return -1 if the indication is not handled.
* channels/sig_pri.h, channels/sig_analog.c, channels/sig_pri.c:
Trim trailing whitespace.
2009-08-03 14:29 +0000 [r210027] Mark Michelson <mmichelson@digium.com>
* main/channel.c: Fix order and redundancy of channel rename
manager events in ast_do_masquerade. Patch contributed by Mark
Spencer.
2009-08-03 14:01 +0000 [r209993] Matthew Nicholson <mnicholson@digium.com>
* addons/chan_mobile.c, configs/chan_mobile.conf.sample: Add an
'sms' option to mobile.conf to manually enable or disable SMS
support. (closes issue #15071) Reported by: ughnz Patches:
optional-sms1.diff uploaded by mnicholson (license 96) Tested by:
ughnz, mnicholson
2009-08-01 23:33 +0000 [r209958-209959] Bradley Latus <brad.latus@gmail.com>
* doc/tex/realtime.tex: Update documentation in relation to
UnixODBC (closes issue #15516) Reported by: snuffy Patches:
bug_odbc_tex_update_v2.diff uploaded by snuffy (license 35)
* doc/CODING-GUIDELINES: (closes issue #15515)
2009-08-01 11:29 +0000 [r209835-209887] Russell Bryant <russell@digium.com>
* /, main/db1-ast/mpool/mpool.c: Merged revisions 209879 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r209879 | russell | 2009-08-01 06:27:25 -0500 (Sat, 01 Aug 2009)
| 5 lines Resolve a valgrind warning about a read from
uninitialized memory. (issue #15396) Reported by: aragon ........
* /, apps/app_milliwatt.c: Merged revisions 209838 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r209838 | russell | 2009-08-01 05:59:05 -0500 (Sat, 01
Aug 2009) | 13 lines Modify how Playtones() is used in
Milliwatt() to resolve gain issue. When Milliwatt() was changed
internally to use Playtones() so that the proper tone was used,
it introduced a drop in gain in the output signal. So, use the
playtones API directly and specify a volume argument such that
the output matches the gain of the original Milliwatt() code.
(closes issue #15386) Reported by: rue_mohr Patches:
issue_15386.rev2.diff uploaded by russell (license 2) Tested by:
rue_mohr ........
* main/event.c: Fix ast_event_queue_and_cache() to actually do the
cache() part. (closes issue #15624) Reported by: ffossard Tested
by: russell
2009-08-01 01:04 +0000 [r209760-209761] Kevin P. Fleming <kpfleming@digium.com>
* Makefile: Revert accidental Makefile change.
* Makefile, channels/chan_dahdi.c, channels/chan_misdn.c, /,
main/Makefile, channels/misdn/ie.c, pbx/pbx_config.c,
utils/frame.c: Merged revisions 209759 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r209759 | kpfleming | 2009-07-31 19:52:00 -0500 (Fri, 31 Jul
2009) | 7 lines Minor changes inspired by testing with latest
GCC. The latest GCC (what will become 4.5.x) has a few new
warnings, that in these cases found some either downright buggy
code, or at least seriously poorly designed code that could be
improved. ........
2009-07-31 21:53 +0000 [r209711] Russell Bryant <russell@digium.com>
* main/event.c: Fix some places where ast_event_type was used
instead of ast_event_ie_type.
2009-07-31 17:57 +0000 [r209673-209674] Mark Michelson <mmichelson@digium.com>
* configs/sip.conf.sample: Add configuration sample code for
previous commit.
* channels/chan_sip.c: Improve chan_sip's ability to determine what
methods should and should not be used in a dialog. The previous
effort here was to store what a peer is capable of receiving by
parsing REGISTER requests from the peer and keeping that
information for as long as the registration was active. The
problem with this is that there are a great number of SIP devices
which give no indication of the methods allowed in their REGISTER
requests, and it is unreasonable to try to guess what the device
may or may not support. In addition, some SIP devices have been
found to claim support for a specific method, but their handling
the method is less than ideal, or they are actually lying. With
this patch, we now determine what methods a device supports by
parsing the Allow header we receive from them, and we do this
with each new dialog. In addition, a configuration option has
been added so that an administrator can essentially blacklist
certain methods from being used with certain peers if the admin
knows that support for a specific method is dodgy or nonexistent.
ABE-1822
2009-07-30 23:37 +0000 [r209623] Sean Bright <sean@malleable.com>
* configure, configure.ac, makeopts.in: Allow passing 'noisy' to
configure's --enable-dev-mode argument to turn on verbose builds.
(closes issue #15607) Reported by: mvanbaak Patches:
20090730_issue15607.patch uploaded by seanbright (license 71)
Tested by: seanbright
2009-07-30 23:31 +0000 [r209619] Jeff Peeler <jpeeler@digium.com>
* channels/sig_pri.h, channels/sig_pri.c: Add missing ifdef-s for
service maintenance message functionality (closes issue #15614)
Reported by: fabled
2009-07-30 16:07 +0000 [r209554] David Brooks <dbrooks@digium.com>
* channels/sig_pri.h, apps/app_forkcdr.c, channels/chan_dahdi.c,
contrib/init.d/rc.debian.asterisk, addons/chan_ooh323.c,
addons/ooh323c/src/ooGkClient.h, funcs/func_math.c,
apps/app_sms.c, codecs/lpc10/pitsyn.c, channels/chan_console.c,
include/asterisk/abstract_jb.h: Fixes numerous spelling errors.
Patch submitted by alecdavis. (closes issue #15595) Reported by:
alecdavis
2009-07-30 14:38 +0000 [r209516] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Fix a crash that can result if text codecs
are allowed but textsupport is disabled. (closes issue #15596)
Reported by: fabled Patches: sip-red.patch uploaded by fabled
(license 448)
2009-07-29 21:46 +0000 [r209453-209484] Matthew Nicholson <mnicholson@digium.com>
* addons/chan_mobile.c: This patch adds the ability to send a CUSD
command to a bluetooth device. (closes issue #15278) Reported by:
Artem Patches: cusd5.patch uploaded by Artem (license 800) Tested
by: mnicholson, Artem Review:
https://reviewboard.asterisk.org/r/274/
* addons/chan_mobile.c: Fixed a comment for hfp_parse_clip
2009-07-28 13:49 +0000 [r209400] Kevin P. Fleming <kpfleming@digium.com>
* channels/chan_usbradio.c, include/asterisk/utils.h,
channels/chan_sip.c, channels/chan_alsa.c,
channels/chan_console.c, channels/chan_oss.c, main/poll.c: Define
side-effect-safe MIN and MAX macros and remove duplicate
definitions from various files.
2009-07-28 00:20 +0000 [r209317-209331] Tilghman Lesher <tlesher@digium.com>
* sounds/sounds.xml: Regex FTL
* /, sounds/sounds.xml: Merged revisions 209315 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r209315 | tilghman | 2009-07-27 19:12:03 -0500 (Mon, 27 Jul 2009)
| 2 lines Publish French extra sounds ........
2009-07-27 21:43 +0000 [r209256-209279] Kevin P. Fleming <kpfleming@digium.com>
* apps/app_fax.c: Cleanup T.38 negotiation changes. Convert
LOG_NOTICE messages about T.38 negotiation in debug level 1
messages, clean up some looping logic, and correct an improper
use of ast_free() for freeing an ast_frame.
* apps/app_fax.c: Make T.38 switchover in ReceiveFAX synchronous.
In receive mode, if the channel that ReceiveFAX is running on
supports T.38, we should *always* attempt to switch T.38, rather
than listening for an incoming CNG tone and only triggering on
that. The channel may be using a low-bitrate codec that distorts
the CNG tone, the sending FAX endpoint may not send CNG at all,
or there could be a variety of other reasons that we don't detect
it, but in all those cases if T.38 is available we certainly want
to use it.
2009-07-27 20:54 +0000 [r209132-209235] Mark Michelson <mmichelson@digium.com>
* res/res_rtp_asterisk.c: Gracefully handle malformed RTP text
packets. AST-2009-004
* res/res_musiconhold.c: Honor channel's music class when using
realtime music on hold. (closes issue #15051) Reported by: alexh
Patches: 15051.patch uploaded by mmichelson (license 60) Tested
by: alexh
* main/udptl.c, /, configs/udptl.conf.sample: Merged revisions
209131 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r209131 | mmichelson | 2009-07-27 12:44:06 -0500 (Mon, 27 Jul
2009) | 18 lines Allow for UDPTL to use only even-numbered ports
if desired. There are some VoIP providers out there that will not
accept SDP offers with odd numbered UDPTL ports. While it is my
personal opinion that these VoIP providers are misinterpreting
RFC 2327, it really is not a big deal to play along with their
silly little games. Of course, since restricting UDPTL ports to
only even numbers reduces the range of available ports by half,
so the option to use only even port numbers is off by default. A
user can enable the behavior by setting use_even_ports=yes in
udptl.conf. (closes issue #15182) Reported by: CGMChris Patches:
15182.patch uploaded by mmichelson (license 60) Tested by:
CGMChris ........
2009-07-27 16:33 +0000 [r209098] David Brooks <dbrooks@digium.com>
* channels/chan_dahdi.c, channels/chan_vpb.cc, res/res_smdi.c,
include/asterisk/module.h, main/features.c, pbx/pbx_dundi.c,
res/res_jabber.c, addons/chan_mobile.c, apps/app_rpt.c,
main/loader.c: Fixing typos. Replaces "recieved" with "received"
and "initilize" with "initialize" (closes issue #15571) Reported
by: alecdavis
2009-07-27 15:38 +0000 [r209056] Kevin P. Fleming <kpfleming@digium.com>
* Makefile: Restore explicit export of ASTCFLAGS/ASTLDFLAGS and
underscore-variants to sub-makes. During the recent Makefile
improvements I made, it seemed the 'make' was automatically
carrying down the ASTCFLAGS/ASTLDFLAGS settings to sub-makes, so
I removed the explict export of them. However, there are some
circumstances where make does this, and some where it does not,
so I've brought them back to ensure they are always exported. I
also removed an extraneous double setting of _ASTLDFLAGS on *BSD
platforms.
2009-07-27 01:20 +0000 [r208924] Jeff Peeler <jpeeler@digium.com>
* /, main/translate.c, channels/chan_iax2.c: Merged revisions
208923 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r208923 | jpeeler | 2009-07-26 20:18:31 -0500 (Sun, 26 Jul 2009)
| 2 lines Fix logic errors from 208746 ........
2009-07-26 14:00 +0000 [r208886] Michiel van Baak <michiel@vanbaak.info>
* contrib/scripts/install_prereq: add OpenBSD to the install_prereq
script
2009-07-25 12:28 +0000 [r208813-208848] Michiel van Baak <michiel@vanbaak.info>
* contrib/scripts/install_prereq: libxml2-dev is needed as well by
default.
* configs/cli_aliases.conf.sample, main/cli.c: add default alias
reload to run module reload. Requiring 'module reload' to reload
everything, including core etc makes russell very unhappy. The
default configuration already loads the 'friendly' aliases
template. Added 'reload=module reload' to that template. Also
removed the comment in main/cli.c that reload should come back.
2009-07-25 06:23 +0000 [r208749] Jeff Peeler <jpeeler@digium.com>
* /, channels/chan_skinny.c, main/translate.c,
channels/chan_iax2.c: Merged revisions 208746 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r208746 | jpeeler | 2009-07-25 01:19:50 -0500 (Sat, 25 Jul 2009)
| 7 lines Fix compiling under dev-mode with gcc 4.4.0. Mostly
trivial changes, but I did not know of any other way to fix the
"dereferencing type-punned pointer will break strict-aliasing
rules" error without creating a tmp variable in chan_skinny.
........
2009-07-24 21:12 +0000 [r208593-208709] Russell Bryant <russell@digium.com>
* pbx/pbx_dundi.c: Remove trailing whitespace.
* main/cli.c: Note that "reload" needs to be added back. I keep
getting annoyed at having to type "module reload" to reload
everything, so I'm adding a note that we need to add "reload"
back. "module reload" doesn't really make sense as the command to
reload everything, including the core.
* main/cli.c: Don't log a warning for something that does not
affect operation.
* apps/app_dial.c, /: Merged revisions 208592 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r208592 | russell | 2009-07-24 13:38:24 -0500 (Fri, 24 Jul 2009)
| 7 lines Do not log an ERROR if autoservice_stop() returns -1.
This does not indicate an error. A return of -1 just means that
the channel has been hung up. (reported in #asterisk-dev)
........
2009-07-24 18:31 +0000 [r208588] Mark Michelson <mmichelson@digium.com>
* /, channels/chan_sip.c: Merged revisions 208587 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r208587 | mmichelson | 2009-07-24 13:26:50 -0500 (Fri, 24 Jul
2009) | 10 lines Only send a BYE when hanging up a channel that
is up. For cases where Asterisk sends an INVITE and receives a
non 2XX final response, Asterisk would follow the INVITE
transaction by immediately sending a BYE, which was unnecessary.
(closes issue #14575) Reported by: chris-mac ........
2009-07-24 15:02 +0000 [r208548] Kevin P. Fleming <kpfleming@digium.com>
* main/udptl.c, channels/chan_sip.c, include/asterisk/udptl.h:
Resolve a T.38 negotiation issue left over from the udptl-updates
merge. The udptl-updates branch that was merged yesterday failed
to properly send back T.38 SDP responses with the correct error
correction mode, if the incoming SDP from the other end caused us
to change error correction modes. This patch corrects that
situation.
2009-07-24 14:35 +0000 [r208542] Michiel van Baak <michiel@vanbaak.info>
* contrib/scripts/install_prereq: use aptitude for debian based
systems The function to check wether we need to install packages
was using dpkg-query which was gives wrong output on Debian 5
Also, the apt-get has been replaced with aptitude because
aptitude is now the preferred way to handle packages on Debian
(closes issue #15570) Reported by: mvanbaak Patches:
2009072400_installprereq-aptitude.diff uploaded by mvanbaak
(license 7)
2009-07-23 22:32 +0000 [r208464-208504] Kevin P. Fleming <kpfleming@digium.com>
* UPGRADE.txt: T.38 change note is not necessary in this branch
* main/channel.c, main/udptl.c, main/frame.c, main/rtp_engine.c,
channels/chan_sip.c, apps/app_fax.c, UPGRADE.txt,
include/asterisk/udptl.h, include/asterisk/frame.h: Rework of
T.38 negotiation and UDPTL API to address interoperability
problems Over the past couple of months, a number of issues with
Asterisk negotiating (and successfully completing) T.38 sessions
with various endpoints have been found. This patch attempts to
address many of them, primarily focused around ensuring that the
endpoints' MaxDatagram size is honored, and in addition by
ensuring that T.38 session parameter negotiation is performed
correctly according to the ITU T.38 Recommendation. The major
changes here are: 1) T.38 applications in Asterisk (app_fax) only
generate/receive IFP packets, they do not ever work with UDPTL
packets. As a result of this, they cannot be allowed to generate
packets that would overflow the other endpoints' MaxDatagram size
after the UDPTL stack adds any error correction information. With
this patch, the application is told the maximum *IFP* size it can
generate, based on a calculation using the far end MaxDatagram
size and the active error correction mode on the T.38 session.
The same is true for sending *our* MaxDatagram size to the remote
endpoint; it is computed from the value that the application says
it can accept (for a single IFP packet) combined with the active
error correction mode. 2) All treatment of T.38 session
parameters as 'capabilities' in chan_sip has been removed; these
parameters are not at all like audio/video stream capabilities.
There are strict rules to follow for computing an answer to a
T.38 offer, and chan_sip now follows those rules, using the
desired parameters from the application (or channel) that wants
to accept the T.38 negotiation. 3) chan_sip now stores and
forwards ast_control_t38_parameters structures for tracking 'our'
and 'their' T.38 session parameters; this greatly simplifies
negotiation, especially for pass-through calls. 4) Since T.38
negotiation without specifying parameters or receiving the final
negotiated parameters is not very worthwhile, the AST_CONTROL_T38
control frame has been removed. A note has been added to
UPGRADE.txt about this removal, since any out-of-tree
applications that use it will no longer function properly until
they are upgraded to use AST_CONTROL_T38_PARAMETERS. Review:
https://reviewboard.asterisk.org/r/310/
2009-07-23 19:34 +0000 [r208388] Mark Michelson <mmichelson@digium.com>
* /, channels/chan_sip.c: Merged revisions 208386 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r208386 | mmichelson | 2009-07-23 14:24:21 -0500 (Thu, 23 Jul
2009) | 17 lines Fix a problem where a 491 response could be sent
out of dialog. This generalizes the fix for issue 13849. The
initial fix corrected the problem that Asterisk would reply with
a 491 if a reinvite were received from an endpoint and we had not
yet received an ACK from that endpoint for the initial INVITE it
had sent us. This expansion also allows Asterisk to appropriately
handle an INVITE with authorization credentials if Asterisk had
not received an ACK from the previous transaction in which
Asterisk had responded to an unauthorized INVITE with a 407.
(closes issue #14239) Reported by: klaus3000 Patches: 14239.patch
uploaded by mmichelson (license 60) Tested by: klaus3000 ........
2009-07-23 19:21 +0000 [r208383] Jeff Peeler <jpeeler@digium.com>
* channels/chan_dahdi.c, /: Merged revisions 208380 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r208380 | jpeeler | 2009-07-23 14:19:53 -0500 (Thu, 23
Jul 2009) | 6 lines Only set the priindication setting when not
performing a reload (closes issue #14696) Reported by: fdecher
........
2009-07-23 16:29 +0000 [r208314] Mark Michelson <mmichelson@digium.com>
* /, channels/chan_sip.c: Merged revisions 208312 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r208312 | mmichelson | 2009-07-23 11:29:18 -0500 (Thu, 23 Jul
2009) | 3 lines Remove inaccurate XXX comment. ........
2009-07-23 15:59 +0000 [r208267] Jeff Peeler <jpeeler@digium.com>
* channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_pri.c:
Fix sending of interface identifier unconditionally in sig_pri
The wrong logic was being used in chan_dahdi to convert a
sig_pri_chan to the proper libpri channel number. The most
significant bit must only be set only when trunk groups are being
used. (closes issue #15452) Reported by: alecdavis Patches:
bug15452.patch uploaded by jpeeler (license 325) Tested by:
alecdavis
2009-07-23 15:46 +0000 [r208229-208263] Mark Michelson <mmichelson@digium.com>
* /, channels/chan_sip.c: Merged revisions 208262 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r208262 | mmichelson | 2009-07-23 10:43:07 -0500 (Thu, 23 Jul
2009) | 8 lines Properly handle 183 responses which do not
contain an SDP. (closes issue #15442) Reported by: ffloimair
Patches: 15442.patch uploaded by mmichelson (license 60) Tested
by: tkarl, ffloimair ........
* channels/chan_sip.c: Fix potential crash if p->owner is NULL.
Problem was observed when a call-forwarding loop was accidentally
configured. ABE-1906
2009-07-23 01:31 +0000 [r208193] Russell Bryant <russell@digium.com>
* main/cel.c: Resolve compiler warning on mac.
2009-07-22 22:42 +0000 [r208155] Jeff Peeler <jpeeler@digium.com>
* channels/chan_dahdi.c: Reset the fax buffers back to default
settings regardless of signaling in use - Pointed out by Matt F.
Also in the case of not using a signaling module, set the law
back to the default as well.
2009-07-22 22:35 +0000 [r208151] Tilghman Lesher <tlesher@digium.com>
* /, include/asterisk/compat.h, main/strcompat.c,
main/asterisk.exports: Merged revisions 208083 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r208083 | tilghman | 2009-07-22 15:23:53 -0500 (Wed, 22 Jul 2009)
| 4 lines Export symbols for functions included in our
compatibility headers. (closes issue #15556) Reported by: smw1218
........
2009-07-22 21:43 +0000 [r208113] Jason Parker <jparker@digium.com>
* apps/app_festival.c: Restore an int declaration on PPC platforms.
This x is one crafty little bugger... It was used for 2 different
things (one of which was only done on PPC) in 1.4. One of the
uses were removed in trunk, and with it went the declaration.
(closes issue #14038) Reported by: ffloimair
2009-07-22 16:49 +0000 [r208052] Tilghman Lesher <tlesher@digium.com>
* res/res_realtime.c: Clarify documentation on 'realtime update2'
to show more than one condition. (closes issue #15357) Reported
by: snuffy Patches: bug_fix_doc_update2.diff uploaded by snuffy
(license 35) (slightly modified by me)
2009-07-22 14:35 +0000 [r208018] Russell Bryant <russell@digium.com>
* include/asterisk/channel.h: Remove trailing whitespace.
2009-07-22 14:35 +0000 [r208017] Mark Michelson <mmichelson@digium.com>
* apps/app_directed_pickup.c: Fix the crash in directed pickups.
For real this time. A shallow pointer copy was causing an
ast_party_connected_line structure to be freed multiple times,
thus causing a crash. (closes issue #15441) Reported by:
lmsteffan Patches: 15441.patch uploaded by mmichelson (license
60) Tested by: lmsteffan
2009-07-21 22:51 +0000 [r207950] Jeff Peeler <jpeeler@digium.com>
* channels/sig_pri.c: Do not dial digits when none were specified
for sig_pri based calls (closes issue #15524) Reported by:
elguero Patches: pri-sig-no-dest-set.patch uploaded by elguero
(license 37)
2009-07-21 22:45 +0000 [r207946] Tilghman Lesher <tlesher@digium.com>
* /, funcs/func_strings.c: Merged revisions 207945 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r207945 | tilghman | 2009-07-21 17:38:54 -0500 (Tue, 21
Jul 2009) | 8 lines Force an error if a blank is passed to QUOTE
(because the documentation states the argument is not optional).
This change makes URIENCODE and QUOTE behave similarly, since the
documentation states that the argument is not optional, for both.
(closes issue #15439) Reported by: pkempgen Patches:
20090706__issue15439.diff.txt uploaded by tilghman (license 14)
........
2009-07-21 22:24 +0000 [r207934] Jeff Peeler <jpeeler@digium.com>
* channels/chan_dahdi.c: whitespace fix only
2009-07-21 22:22 +0000 [r207925] Russell Bryant <russell@digium.com>
* doc/CODING-GUIDELINES: Note that we use tabs instead of spaces
for indentation. I'm surprised this was never actually in here...
2009-07-21 22:02 +0000 [r207854-207902] Jeff Peeler <jpeeler@digium.com>
* channels/chan_dahdi.c: Fix my_is_off_hook to check rxbits only
for FXS signaling
* channels/chan_dahdi.c, channels/sig_analog.c, /: Merged revisions
207827 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r207827 | jpeeler | 2009-07-21 15:16:55 -0500 (Tue, 21 Jul 2009)
| 9 lines Wait for wink before dialing when using E&M wink
signaling There was already code for other signaling types in
dahdi_handle_event to handle dialing if a dial operation dial
string was present. Simply add SIG_EMWINK to the list. (closes
issue #14434) Reported by: araasch ........
2009-07-21 14:29 +0000 [r207723] Mark Michelson <mmichelson@digium.com>
* main/manager.c, /: Merged revisions 207714 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r207714 | mmichelson | 2009-07-21 09:26:00 -0500 (Tue, 21 Jul
2009) | 5 lines Document default timeout for AMI originations.
AST-224 ........
2009-07-21 13:28 +0000 [r207680] Kevin P. Fleming <kpfleming@digium.com>
* /, main/Makefile, codecs/gsm/Makefile, Makefile.moddir_rules,
res/Makefile, pbx/Makefile, Makefile.rules, channels/Makefile,
doc/video_console.txt, Makefile, utils/Makefile, codecs/Makefile,
agi/Makefile, addons/Makefile, funcs/Makefile,
codecs/lpc10/Makefile, main/db1-ast/Makefile: Merged revisions
207647 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r207647 | kpfleming | 2009-07-21 08:04:44 -0500 (Tue, 21 Jul
2009) | 12 lines Ensure that user-provided CFLAGS and LDFLAGS are
honored. This commit changes the build system so that
user-provided flags (in ASTCFLAGS and ASTLDFLAGS) are supplied to
the compiler/linker *after* all flags provided by the build
system itself, so that the user can effectively override the
build system's flags if desired. In addition, ASTCFLAGS and
ASTLDFLAGS can now be provided *either* in the environment before
running 'make', or as variable assignments on the 'make' command
line. As a result, the use of COPTS and LDOPTS is no longer
necessary, so they are no longer documented, but are still
supported so as not to break existing build systems that supply
them when building Asterisk. ........
2009-07-20 23:08 +0000 [r207522-207551] Mark Michelson <mmichelson@digium.com>
* apps/app_directed_pickup.c: Okay, that didn't fix the crash. It
didn't really do anything useful.
* apps/app_directed_pickup.c: Initialize connected line instance
when doing a directed pickup. This helps to prevent a crash which
may occur due to our freeing garbage due to a struct being
uninitialized.
2009-07-20 20:45 +0000 [r207484] David Vossel <dvossel@digium.com>
* channels/chan_sip.c: reg->username is parsed only once on sip
reload The registration string can contain an expanded user
portion of the form user@domain. This expanded user portion was
stored in reg->username and parsed each time there is a
registration refresh. Now, the domain portion of the user is
parsed and stored separately in the regdomain field. (closes
issue #14331) Reported by: Nick_Lewis Patches:
chan_sip.c.domainparse3.patch uploaded by Nick (license 657)
Tested by: Nick_Lewis, dvossel
2009-07-20 19:48 +0000 [r207424] Mark Michelson <mmichelson@digium.com>
* /, channels/chan_sip.c: Merged revisions 207423 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r207423 | mmichelson | 2009-07-20 14:39:59 -0500 (Mon, 20 Jul
2009) | 33 lines Answer video SDP offers properly when
videosupport is not enabled. Copied from Review board: In issue
12434, the reporter describes a situation in which audio and
video is offered on the call, but because videosupport is
disabled in sip.conf, Asterisk gives no response at all to the
video offer. According to RFC 3264, all media offers should have
a corresponding answer. For offers we do not intend to actually
reply to with meaningful values, we should still reply with the
port for the media stream set to 0. In this patch, we take note
of what types of media have been offered and save the information
on the sip_pvt. The SDP in the response will take into account
whether media was offered. If we are not otherwise going to
answer a media offer, we will insert an appropriate m= line with
the port set to 0. It is important to note that this patch is
pretty much a bandage being applied to a broken bone. The patch
*only* helps for situations where video is offered but
videosupport is disabled and when udptl_pt is disabled but T.38
is offered. Asterisk is not guaranteed to respond to every media
offer. Notable cases are when multiple streams of the same type
are offered. The 2 media stream limit is still present with this
patch, too. In trunk and the 1.6.X branches, things will be a bit
different since Asterisk also supports text in SDPs as well.
(closes issue #12434) Reported by: mnnojd Review:
https://reviewboard.asterisk.org/r/311 Review:
https://reviewboard.asterisk.org/r/313 ........
2009-07-20 16:36 +0000 [r207361] Russell Bryant <russell@digium.com>
* main/channel.c, /: Merged revisions 207360 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r207360 | russell | 2009-07-20 11:26:24 -0500 (Mon, 20 Jul 2009)
| 9 lines Only do the chan->fdno check in ast_read() in a
developer build. I changed this check to only happen in a
dev-mode build. I also added a comment explaining what is going
on. I also made it so that detection of this situation does not
affect ast_read() operation. (closes issue #14723) Reported by:
seadweller ........
2009-07-18 04:17 +0000 [r207318] Richard Mudgett <rmudgett@digium.com>
* channels/chan_misdn.c, CHANGES: Merged 207316 from
https://origsvn.digium.com/svn/asterisk/be/branches/C.2-...
.......... r207316 | rmudgett | 2009-07-17 23:05:05 -0500 (Fri,
17 Jul 2009) | 20 lines Fixed incoming calls being matched to
MSNs without type-of-number prefix added. For an incoming ISDN
call the dialed.number is incorrectly matched against the
configured MSNs in misdn.conf. The numbers passed to the dialplan
include the configured prefix for the dialed.number_type, whereas
the check against the configured MSNs (to decide if the call is
accepted at all), is executed without the configured prefix.
e.g., dialed.number = 241168020, TON = national, configured
national prefix is "0". (This is the TON which is used by ISDN
providers in the Netherlands.) In chan_misdn.c:cb_events() in
case EVENT_SETUP the call to misdn_cfg_is_msn_valid() uses the
unnormalized number 241168020, but 57 lines later the call to
read_config() adds the prefix, and the dialed.number is now
0241168020, which is then used in the dialplan.
misdn_cfg_is_msn_valid() must use the normalized number, too.
JIRA ABE-1912
2009-07-18 04:16 +0000 [r207317] Tilghman Lesher <tlesher@digium.com>
* apps/app_voicemail.c: Flag field in wrong position. Reported by
"Hoggins!" on asterisk-dev list.
2009-07-18 01:31 +0000 [r207285] Richard Mudgett <rmudgett@digium.com>
* /: Recorded merge of revisions 145293,158010 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
................ r145293 | rmudgett | 2008-09-30 18:55:24 -0500
(Tue, 30 Sep 2008) | 54 lines channels/chan_misdn.c
channels/misdn/isdn_lib.c * Miscellaneous other fixes from trunk
to make merging easier later. ........ r145200 | rmudgett |
2008-09-30 16:00:54 -0500 (Tue, 30 Sep 2008) | 7 lines *
Miscellaneous formatting changes to make v1.4 and trunk more
merge compatible in the mISDN area. channels/chan_misdn.c *
Eliminated redundant code in cb_events() EVENT_SETUP ........
r144257 | crichter | 2008-09-24 03:42:55 -0500 (Wed, 24 Sep 2008)
| 9 lines improved helptext of misdn_set_opt. ........ r142181 |
rmudgett | 2008-09-09 12:30:52 -0500 (Tue, 09 Sep 2008) | 1 line
Cleaned up comment ........ r138738 | rmudgett | 2008-08-18
16:07:28 -0500 (Mon, 18 Aug 2008) | 30 lines
channels/chan_misdn.c * Made bearer2str() use
allowed_bearers_array[] * Made use the causes.h defines instead
of hardcoded numbers. * Made use Asterisk presentation indicator
values if either of the mISDN presentation or screen options are
negative. * Updated the misdn_set_opt application option
descriptions. * Renamed the awkward Caller ID presentation
misdn_set_opt application option value not_screened to
restricted. Deprecated the not_screened option value.
channels/misdn/isdn_lib.c * Made use the causes.h defines instead
of hardcoded numbers. * Fixed some spelling errors and typos. *
Added all defined facility code strings to fac2str().
channels/misdn/isdn_lib.h * Added doxygen comments to struct
misdn_bchannel. channels/misdn/isdn_lib_intern.h * Added doxygen
comments to struct misdn_stack. channels/misdn_config.c
configs/misdn.conf.sample * Updated the mISDN presentation and
screen parameter descriptions. doc/misdn.txt (doc/tex/misdn.tex)
* Updated the misdn_set_opt application option descriptions. *
Fixed some spelling errors and typos. ................ r158010 |
rmudgett | 2008-11-19 19:46:09 -0600 (Wed, 19 Nov 2008) | 9 lines
Merged revision 157977 from
https://origsvn.digium.com/svn/asterisk/team/group/issue8824
........ Fixes JIRA ABE-1726 The dial extension could be empty if
you are using MISDN_KEYPAD to control ISDN provider features.
................
2009-07-17 22:29 +0000 [r207255] Tilghman Lesher <tlesher@digium.com>
* doc/voicemail_odbc_postgresql.txt: Add flag here, too (as
requested by jsmith)
2009-07-17 22:07 +0000 [r207225] David Vossel <dvossel@digium.com>
* channels/chan_iax2.c: fixes an error in r203638 CEL commit
(closes issue #15525) Reported by: elguero Patches:
iax2-double-unlock.patch uploaded by elguero (license 37)
15525.diff uploaded by dvossel (license 671) Tested by: dvossel
2009-07-17 22:04 +0000 [r207224] Tilghman Lesher <tlesher@digium.com>
* doc/tex/odbcstorage.tex, UPGRADE.txt: Document the "flag" field
in the voicemessages table.
2009-07-17 19:37 +0000 [r207095-207156] Jeff Peeler <jpeeler@digium.com>
* channels/chan_dahdi.c, /: Merged revisions 207155 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r207155 | jpeeler | 2009-07-17 14:36:19 -0500 (Fri, 17
Jul 2009) | 7 lines Fix format specifier to print out an unsigned
long long. Yep, it's even ifdefed out code. But it made it to the
RR list... (closes issue #14726) Reported by: lmadsen ........
* configs/chan_dahdi.conf.sample: Update some missing allowed
options for overlapdial
2009-07-17 17:51 +0000 [r207029] David Vossel <dvossel@digium.com>
* channels/chan_sip.c: sip option flags handled incorrectly (closes
issue #15376) Reported by: Takehiko Ooshima Tested by: dvossel,
Takehiko_Ooshima
2009-07-17 17:02 +0000 [r206998] Jeff Peeler <jpeeler@digium.com>
* channels/chan_dahdi.c, channels/sig_analog.c: Fix segfault in
sig_analog when using callwaiting, respect callwaiting options
Sig_analog handles allocating the sub channel for callwaiting, so
no longer try to do it in chan_dahdi. Modified analog_alloc_sub
to only mark the sub as allocated upon success of the alloc_sub
callback, which was responsible for the segfault. Also, the
callwaiting and callwaitingcallerid options were being
unconditionally set to true. Now, the options are properly set
from chan_dahdi.conf. (closes issue #15508) Reported by: elguero
Tested by: elguero
2009-07-17 16:13 +0000 [r206868-206939] David Vossel <dvossel@digium.com>
* /, channels/chan_sip.c: Merged revisions 206938 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r206938 | dvossel | 2009-07-17 11:05:06 -0500 (Fri, 17 Jul 2009)
| 14 lines SIP incorrect From: header information when callpres
is prohib Some ITSP make use of the "Anonymous" display name to
detect a requirement to withhold caller id across the PSTN. This
does not work if the display name is "Unknown". (closes issue
#14465) Reported by: Nick_Lewis Patches:
chan_sip.c-callerpres.patch uploaded by Nick (license 657)
chan_sip.c-callerpres_trunk.patch uploaded by dvossel (license
671) Tested by: Nick_Lewis, dvossel ........
* funcs/func_timeout.c: TIMEOUT(absolute) returned negative value.
(closes issue #15513) Reported by: ys
* configs/iax.conf.sample, /: Merged revisions 206872 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r206872 | dvossel | 2009-07-16 16:33:19 -0500 (Thu, 16
Jul 2009) | 6 lines error in iax.conf related IP-based access
control (closes issue #15518) Reported by: pkempgen ........
* /, main/callerid.c: Merged revisions 206867 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r206867 | dvossel | 2009-07-16 16:24:16 -0500 (Thu, 16 Jul 2009)
| 8 lines avoid segfault caused by user error If the CALLERPRES()
dialplan function is set to nothing, a segfault occurs. This is
user error to begin with, but I'd rather see a cli warning
message than have Asterisk crash on me. ........
2009-07-16 16:51 +0000 [r206808] Tilghman Lesher <tlesher@digium.com>
* /, funcs/func_realtime.c: Merged revisions 206807 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r206807 | tilghman | 2009-07-16 11:27:35 -0500 (Thu, 16
Jul 2009) | 6 lines Fix a memory leak. (closes issue #15517)
Reported by: adomjan Patches:
func_realtime.c-ast_variable_destroy.diff uploaded by adomjan
(license 487) ........
2009-07-15 22:04 +0000 [r206768] David Vossel <dvossel@digium.com>
* channels/chan_sip.c: Session timer were not activated if
Supported header field in INVITE had both "timer" and other
options. (closes issue #15403) Reported by: makoto Patches:
sip-session-timer.patch uploaded by makoto (license 38)
2009-07-15 22:02 +0000 [r206767] Jeff Peeler <jpeeler@digium.com>
* channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_analog.c,
channels/sig_analog.h, channels/sig_pri.c: The dialing flag was
mistakingly removed from sig_pri. This readds the proper setting
of the flag and is really a continuation of r205731. The flag was
being set properly in sig_analog, but use of the newly added
set_dialing callback allowed for some simplification in
chan_dahdi. (closes issue #15486) Reported by: rmudgett
2009-07-15 21:14 +0000 [r206707] Richard Mudgett <rmudgett@digium.com>
* channels/misdn/isdn_lib_intern.h, /, channels/misdn/isdn_lib.c:
Merged revisions 206706 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
................ r206706 | rmudgett | 2009-07-15 15:44:55 -0500
(Wed, 15 Jul 2009) | 26 lines Merged revision 206700 from
https://origsvn.digium.com/svn/asterisk/be/branches/C.2-...
.......... Fixed chan_misdn crash because mISDNuser library is
not thread safe. With Asterisk the mISDNuser library is driven by
two threads concurrently: 1.
channels/misdn/isdn_lib.c::manager_event_handler() 2.
channels/misdn/isdn_lib.c::misdn_lib_isdn_event_catcher() Calls
into the library are done concurrently and recursively from
isdn_lib.c. Both threads can fiddle with the master/child
layer3_proc_t lists. One thread may traverse the list when the
other interrupts it and then removes the list element which the
first thread was currently handling. This is exactly what caused
the crash. About 60 calls were needed to a Gigaset CX475 before
it occurred once. This patch adds locking when calling into the
mISDNuser library. This also fixes some cb_log calls with wrong
port parameter. JIRA ABE-1913 Patches: misdn-locking.patch
(Modified with mostly cosmetic changes) ..........
................
2009-07-15 20:20 +0000 [r206702] David Vossel <dvossel@digium.com>
* channels/chan_sip.c: callerid(num) is wrong when username is
missing A domain only sip uri <sip:123.123.123.123> would return
123.123.123.123 as callid num. Now, if the username is missing
from a uri, the callerid num field is left empty. (closes issue
#15476) Reported by: viraptor
2009-07-15 16:00 +0000 [r206636] Sean Bright <sean@malleable.com>
* /, codecs/codec_dahdi.c: Merged revisions 206635 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r206635 | seanbright | 2009-07-15 11:57:51 -0400 (Wed,
15 Jul 2009) | 1 line Only print debug info in codec_dahdi if we
are asking for it. ........
2009-07-14 20:38 +0000 [r206603] Jeff Peeler <jpeeler@digium.com>
* configs/chan_dahdi.conf.sample: fix a typo in sample config file
for option change
2009-07-14 20:14 +0000 [r206567] Tilghman Lesher <tlesher@digium.com>
* apps/app_meetme.c, contrib/scripts/meetme.sql: Document all
meetme realtime fields, and in the process, make some field
lengths more consistent. (closes issue #15493) Reported by: lasko
Patches: meetme.diff uploaded by lasko (license 833)
2009-07-14 20:01 +0000 [r206566] Jeff Peeler <jpeeler@digium.com>
* channels/chan_dahdi.c, channels/sig_analog.c,
channels/sig_analog.h: Restore some missing functionality to
sig_analog. The main purpose of this commit is to restore missing
functionality present in the ss_thread before all the sig related
work was done. Two of the biggest missing things were distinctive
ring detection and cid handling for V23. fxsoffhookstate and
associated mwi variables have been moved inside sig_analog as
they were not being set properly as well.
2009-07-14 17:03 +0000 [r206490] Mark Michelson <mmichelson@digium.com>
* apps/app_dial.c: I AM A TERRIBLE PERSON
2009-07-14 17:01 +0000 [r206489] Richard Mudgett <rmudgett@digium.com>
* channels/misdn/isdn_lib.h, channels/chan_misdn.c, /,
channels/misdn/isdn_lib.c: Merged revisions 206487 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r206487 | rmudgett | 2009-07-14 11:44:47 -0500 (Tue, 14
Jul 2009) | 28 lines Fixes several call transfer issues with
chan_misdn. * issue #14355 - Crash if attempt to transfer a call
to an application. Masquerade the other pair of the four asterisk
channels involved in the two calls. The held call already must be
a bridged call (not an applicaton) or it would have been
rejected. * issue #14692 - Held calls are not automatically
cleared after transfer. Allow the core to initate disconnect of
held calls to the ISDN port. This also fixes a similar case where
the party on hold hangs up before being transferred or taken off
hold. * JIRA ABE-1903 - Orphaned held calls left in
music-on-hold. Do not simply block passing the hangup event on
held calls to asterisk core. * Fixed to allow held calls to be
transferred to ringing calls. Previously, held calls could only
be transferred to connected calls. * Eliminated unused call
states to simplify hangup code. * Eliminated most uses of
"holded" because it is not a word. (closes issue #14355) (closes
issue #14692) Reported by: sodom Patches:
misdn_xfer_v14_r205839.patch uploaded by rmudgett (license 664)
Tested by: rmudgett ........
2009-07-14 16:09 +0000 [r206455] Mark Michelson <mmichelson@digium.com>
* apps/app_dial.c: Reset the sentringing indication when redirects
occur. If a redirecting control frame is processed or a call
forward occurs, we need to reset the sentringing flag so that we
can send another ringing indication to the phone that may contain
a connected line update. AST-164
2009-07-14 14:51 +0000 [r206386] Russell Bryant <russell@digium.com>
* /, channels/chan_iax2.c: Merged revisions 206385 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
................ r206385 | russell | 2009-07-14 09:48:00 -0500
(Tue, 14 Jul 2009) | 13 lines Merged revisions 206384 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r206384 | russell | 2009-07-14 09:45:47 -0500 (Tue, 14 Jul 2009)
| 6 lines Ensure apathetic replies are sent out on the proper
socket. chan_iax2 supports multiple address bindings. The
send_apathetic_reply() function did not attempt to send its
response on the same socket that the incoming message came in on.
........ ................
2009-07-14 00:48 +0000 [r206341] Richard Mudgett <rmudgett@digium.com>
* channels/chan_misdn.c, /, channels/misdn/isdn_lib.c: Merged
revisions 206284 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r206284 | rmudgett | 2009-07-13 19:17:28 -0500 (Mon, 13 Jul 2009)
| 4 lines Fix some memory leaks in chan_misdn. JIRA ABE-1911
........
2009-07-13 23:26 +0000 [r206280] David Vossel <dvossel@digium.com>
* channels/chan_sip.c: dns lookup of peername rather than peer's
host in transmit_register() (closes issue #15052) Reported by:
fsantulli Patches: chan_sip_bug_15052_[20090626204511].patch
uploaded by fsantulli (license 818) Tested by: fsantulli
2009-07-13 18:46 +0000 [r206225] Sean Bright <sean@malleable.com>
* contrib/upstart/asterisk.upstart-0.3.9: Make sure that since we
are passing -c to asterisk that we have a console. Without this
line, Asterisk will busy-loop trying to read and write to
/dev/null (woops... my bad).
2009-07-13 16:23 +0000 [r206185] Tilghman Lesher <tlesher@digium.com>
* apps/app_voicemail.c: Remove reference to non-existent help file
(closes issue #15427) Reported by: brushtyler Patches:
app_voicemail.c.diff uploaded by brushtyler (license 821)
2009-07-13 14:06 +0000 [r206092-206094] Kevin P. Fleming <kpfleming@digium.com>
* .cleancount: Bump up cleancount so that existing checkouts will
update themselves properly for the 'Addons' -> 'ADDONS' change.
* addons/Makefile: Make the menuselect category for Add-Ons
consistent with the other directories (uppercase).
2009-07-11 19:30 +0000 [r206021-206049] Russell Bryant <russell@digium.com>
* CHANGES: note the security events API in CHANGES
* doc/tex/security-events.tex (added), tests/test_security_events.c
(added), main/manager.c, main/security_events.c (added),
include/asterisk/event_defs.h, main/event.c,
include/asterisk/security_events.h (added), doc/tex/asterisk.tex,
include/asterisk/security_events_defs.h (added),
res/res_security_log.c (added), tests/test_ami_security_events.sh
(added): Add an API for reporting security events, and a security
event logging module. This commit introduces the security events
API. This API is to be used by Asterisk components to report
events that have security implications. A simple example is when
a connection is made but fails authentication. These events can
be used by external tools manipulate firewall rules or something
similar after detecting unusual activity based on security
events. Inside of Asterisk, the events go through the ast_event
API. This means that they have a binary encoding, and it is easy
to write code to subscribe to these events and do something with
them. One module is provided that is a subscriber to these events
- res_security_log. This module turns security events into a
parseable text format and sends them to the "security" logger
level. Using logger.conf, these log entries may be sent to a
file, or to syslog. One service, AMI, has been fully updated for
reporting security events. AMI was chosen as it was a fairly
straight forward service to convert. The next target will be
chan_sip. That will be more complicated and will be done as its
own project as the next phase of security events work. For more
information on the security events framework, see the
documentation generated from doc/tex/. "make asterisk.pdf"
Review: https://reviewboard.asterisk.org/r/273/
2009-07-10 21:42 +0000 [r205985] David Vossel <dvossel@digium.com>
* channels/chan_sip.c: SIP register not using peer's outbound proxy
If callbackextension is defined for a peer it successfully causes
a registration to occur, but the registration ignores the
outboundproxy settings for the peer. This patch allows the peer
to be passed to obproxy_get() in transmit_register(). (closes
issue #14344) Reported by: Nick_Lewis Patches:
callbackextension_peer_trunk.diff uploaded by dvossel (license
671) Tested by: dvossel Review:
https://reviewboard.asterisk.org/r/294/
2009-07-10 18:44 +0000 [r205939] Kevin P. Fleming <kpfleming@digium.com>
* main/udptl.c: Update comments about the level of T.38 support in
Asterisk.
2009-07-10 17:39 +0000 [r205878] Mark Michelson <mmichelson@digium.com>
* /, channels/chan_sip.c: Merged revisions 205877 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
................ r205877 | mmichelson | 2009-07-10 12:39:13 -0500
(Fri, 10 Jul 2009) | 23 lines Merged revisions 205776 via
svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
................ r205776 | mmichelson | 2009-07-10 10:56:45 -0500
(Fri, 10 Jul 2009) | 16 lines Merged revisions 205775 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r205775 | mmichelson | 2009-07-10 10:51:36 -0500 (Fri, 10 Jul
2009) | 10 lines Ensure that outbound NOTIFY requests are
properly routed through stateful proxies. With this change, we
make note of Record-Route headers present in any SUBSCRIBE
request that we receive so that our outbound NOTIFY requests will
have the proper Route headers in them. (closes issue #14725)
Reported by: ibc ........ ................ ................
2009-07-10 16:42 +0000 [r205840] David Vossel <dvossel@digium.com>
* /, channels/chan_sip.c: Merged revisions 205804 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r205804 | dvossel | 2009-07-10 11:23:59 -0500 (Fri, 10 Jul 2009)
| 31 lines SIP registration auth loop caused by stale nonce If an
endpoint sends two registration requests in a very short period
of time with the same nonce, both receive 401 responses from
Asterisk, each with a different nonce (the second 401 containing
the current nonce and the first one being stale). If the endpoint
responds to the first 401, it does not match the current nonce so
Asterisk sends a third 401 with a newly generated nonce (which
updates the current nonce)... Now if the endpoint responds to the
second 401, it does not match the current nonce either and
Asterisk sends a fourth 401 with a newly generated nonce... This
loop goes on and on. There appears to be a simple fix for this.
If the nonce from the request does not match our nonce, but is a
good response to a previous nonce, instead of sending a 401 with
a newly generated nonce, use the current one instead. This breaks
the loop as the nonce is not updated until a response is
received. Additional logic has been added to make sure no nonce
can be responded to twice though. (closes issue #15102) Reported
by: Jamuel Patches: patch-bug_0015102 uploaded by Jamuel (license
809) nonce_sip.diff uploaded by dvossel (license 671) Tested by:
Jamuel Review: https://reviewboard.asterisk.org/r/289/ ........
2009-07-10 16:00 +0000 [r205780] Kevin P. Fleming <kpfleming@digium.com>
* apps/app_fax.c: Eliminate extraneous LOG_DEBUG messages generated
by app_fax. The transmit_audio() and transmit_t38() functions in
app_fax have processing loops that are supposed to wait for
frames to arrive on the channel and then handle them, but they
also have short timeouts so that the loops can have watchdog
timers and do other required processing. This commit changes the
loops to not actually call ast_read() and attempt to process the
returned frame unless a frame actually arrived, eliminating
hundreds of LOG_DEBUG messages and slightly improving
performance.
2009-07-10 15:56 +0000 [r205776] Mark Michelson <mmichelson@digium.com>
* /, channels/chan_sip.c: Merged revisions 205775 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r205775 | mmichelson | 2009-07-10 10:51:36 -0500 (Fri, 10 Jul
2009) | 10 lines Ensure that outbound NOTIFY requests are
properly routed through stateful proxies. With this change, we
make note of Record-Route headers present in any SUBSCRIBE
request that we receive so that our outbound NOTIFY requests will
have the proper Route headers in them. (closes issue #14725)
Reported by: ibc ........
2009-07-10 15:28 +0000 [r205770] Kevin P. Fleming <kpfleming@digium.com>
* apps/app_fax.c: Fix some remaining T.38 negotiation problems in
app_fax. Revision 205696 did not quite fix all the issues with
the T.38 negotiation changes and app_fax; this patch corrects
them, along with a couple of other minor issues. (closes issue
#15480) Reported by: dimas Patches: test2-15480.patch uploaded by
dimas (license 88)
2009-07-09 21:32 +0000 [r205700] Matthew Nicholson <mnicholson@digium.com>
* addons/chan_mobile.c: Fix mbl_fixup() in chan_mobile to update
newchan->tech_pvt instead of oldchan. (closes issue #15299)
Reported by: nikkk
2009-07-09 21:20 +0000 [r205696] Kevin P. Fleming <kpfleming@digium.com>
* channels/chan_sip.c, apps/app_fax.c, include/asterisk/frame.h:
Repair ability of SendFAX/ReceiveFAX to respond to T.38
switchover. Recent changes in T.38 negotiation in Asterisk caused
these applications to not respond when the other endpoint
initiated a switchover to T.38; this resulted in the T.38
switchover failing, and the FAX attempt to be made using an audio
connection, instead of T.38 (which would usually cause the FAX to
fail completely). This patch corrects this problem, and the
applications will now correctly respond to the T.38 switchover
request. In addition, the response will include the appopriate
T.38 session parameters based on what the other end offered and
what our end is capable of. (closes issue #14849) Reported by:
afosorio
2009-07-09 20:04 +0000 [r205666] Matthew Nicholson <mnicholson@digium.com>
* funcs/func_odbc.c: Convert func_odbc to use
ast_dummy_alloc_channel() Review:
https://reviewboard.asterisk.org/r/290/
2009-07-09 16:19 +0000 [r205600] David Vossel <dvossel@digium.com>
* /, include/asterisk/time.h: Merged revisions 205599 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r205599 | dvossel | 2009-07-09 11:18:09 -0500 (Thu, 09
Jul 2009) | 2 lines Changing ast_samp2tv to not use floating
point. ........
2009-07-09 14:10 +0000 [r205532-205562] Michiel van Baak <michiel@vanbaak.info>
* main/cel.c: make this compile again under devmode
* main/ssl.c: pthread_self returns a pthread_t which is not an
unsigned int on all pthread implementations. Casting it to an
unsigned int fixes compiler warnings. Tested on OpenBSD and Linux
both 32 and 64 bit
2009-07-08 23:19 +0000 [r205479] David Vossel <dvossel@digium.com>
* res/res_rtp_asterisk.c, /, channels/chan_iax2.c,
include/asterisk/frame.h: Merged revisions 205471 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r205471 | dvossel | 2009-07-08 18:15:54 -0500 (Wed, 08
Jul 2009) | 10 lines Fixes 8khz assumptions Many calculations
assume 8khz is the codec rate. This is not always the case. This
patch only addresses chan_iax.c and res_rtp_asterisk.c, but I am
sure there are other areas that make this assumption as well.
Review: https://reviewboard.asterisk.org/r/306/ ........
2009-07-08 23:07 +0000 [r205469] Matthew Nicholson <mnicholson@digium.com>
* main/pbx.c: Fix a CEL related regression with hints updating by
subscribing to AST_DEVICE_STATE instead of
AST_DEVICE_STATE_CHANGED. (closes issue #15440) Reported by:
lmsteffan
2009-07-08 22:15 +0000 [r205410-205412] David Vossel <dvossel@digium.com>
* include/asterisk/devicestate.h, main/pbx.c, /,
main/devicestate.c, include/asterisk/pbx.h: Merged revisions
205409 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r205409 | dvossel | 2009-07-08 16:35:12 -0500 (Wed, 08 Jul 2009)
| 6 lines moving ast_devstate_to_extenstate to pbx.c from
devicestate.c ast_devstate_to_extenstate belongs in pbx.c. This
change fixes a compile time error with chan_vpb as well. ........
* main/devicestate.c: missing comma in devstatestring array
2009-07-08 19:26 +0000 [r205350] Mark Michelson <mmichelson@digium.com>
* /, apps/app_queue.c: Merged revisions 205349 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r205349 | mmichelson | 2009-07-08 14:26:13 -0500 (Wed, 08 Jul
2009) | 14 lines Prevent phantom calls to queue members. If a
caller were to hang up while a periodic announcement or position
were being said, the return value for those functions would
incorrectly indicate that the caller was still in the queue. With
these changes, the problem does not occur. (closes issue #14631)
Reported by: latinsud Patches: queue_announce_ghost_call2.diff
uploaded by latinsud (license 745) (with small modification from
me) ........
2009-07-08 18:19 +0000 [r205291] Jason Parker <jparker@digium.com>
* config.sub, /, config.guess: Merged revisions 205288 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r205288 | qwell | 2009-07-08 13:19:03 -0500 (Wed, 08 Jul
2009) | 1 line Update config.guess and config.sub from the
savannah.gnu.org git repo. ........
2009-07-08 17:26 +0000 [r205254] David Brooks <dbrooks@digium.com>
* main/features.c: Fixes Park() argument handling Park() was not
respecting the arguments passed to it. Any
extension/context/priority given to it was being ignored. This
patch remedies this. (closes issue #15380) Reported by: DLNoah
2009-07-08 16:59 +0000 [r205221] Tilghman Lesher <tlesher@digium.com>
* main/say.c: Oops, fixing build
2009-07-08 16:54 +0000 [r205216] David Vossel <dvossel@digium.com>
* /, include/asterisk/time.h: Merged revisions 205215 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r205215 | dvossel | 2009-07-08 11:53:40 -0500 (Wed, 08
Jul 2009) | 10 lines ast_samp2tv needs floating point for 16khz
audio In ast_samp2tv(), (1000000 / _rate) = 62.5 when _rate is
16000. The .5 is currently stripped off because we don't
calculate using floating points. This causes madness with 16khz
audio. (issue ABE-1899) Review:
https://reviewboard.asterisk.org/r/305/ ........
2009-07-08 16:43 +0000 [r205214] Sean Bright <sean@malleable.com>
* utils/muted.c, configure, include/asterisk/autoconfig.h.in,
configure.ac, main/dns.c: Fix a few compilation problems found
when building Asterisk against uClibc.
2009-07-08 16:27 +0000 [r205196] Tilghman Lesher <tlesher@digium.com>
* /, main/say.c: Merged revisions 205188 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r205188 | tilghman | 2009-07-08 11:26:15 -0500 (Wed, 08 Jul 2009)
| 2 lines Add redirection warnings for the invalid language codes
previously removed. ........
2009-07-08 15:56 +0000 [r205120-205151] Russell Bryant <russell@digium.com>
* main/ssl.c: Use tabs instead of spaces for indentation.
* res/res_crypto.c, main/ssl.c (added),
include/asterisk/_private.h, res/res_jabber.c, main/asterisk.c:
Move OpenSSL initialization to a single place, make library usage
thread-safe. While doing some reading about OpenSSL, I noticed a
couple of things that needed to be improved with our usage of
OpenSSL. 1) We had initialization of the library done in multiple
modules. This has now been moved to a core function that gets
executed during Asterisk startup. We already link OpenSSL into
the core for TCP/TLS functionality, so this was the most logical
place to do it. 2) OpenSSL is not thread-safe by default.
However, making it thread safe is very easy. We just have to
provide a couple of callbacks. One callback returns a thread ID.
The other handles locking. For more information, start with the
"Is OpenSSL thread-safe?" question on the FAQ page of
openssl.org.
2009-07-08 14:45 +0000 [r205118] Luigi Rizzo <rizzo@icir.org>
* bootstrap.sh: FreeBSD now has autoconf 2.62 in the ports, 2.61
has disappeared.
2009-07-07 21:10 +0000 [r205086] Tilghman Lesher <tlesher@digium.com>
* channels/chan_sip.c: Permit setting custom headers from the peer
definition. (closes issue #14059) Reported by: fnordian
2009-07-07 18:24 +0000 [r205014-205047] Matthew Nicholson <mnicholson@digium.com>
* channels/sig_analog.c: Fix a deadlock in sig_analog
* channels/sig_analog.c: Add CEL transfer events to analog
(chan_dahdi) transfers.
2009-07-06 21:37 +0000 [r204986] Tilghman Lesher <tlesher@digium.com>
* addons/res_config_mysql.c: Merged revisions 981 via svnmerge from
https://origsvn.digium.com/svn/asterisk-addons/branches/1.4
........ r981 | tilghman | 2009-07-06 16:30:13 -0500 (Mon, 06 Jul
2009) | 7 lines Don't reset reconnect time, unless a reconnect
really occurred. (closes issue #15375) Reported by: kowalma
Patches: 20090628__issue15375.diff.txt uploaded by tilghman
(license 14) Tested by: kowalma, jacco ........
2009-07-06 13:38 +0000 [r204948] Kevin P. Fleming <kpfleming@digium.com>
* main/channel.c: Improve handling of AST_CONTROL_T38 and
AST_CONTROL_T38_PARAMETERS for non-T.38-capable channels. This
change allows applications that request T.38 negotiation on a
channel that does not support it to get the proper indication
that it is not supported, rather than thinking that negotiation
was started when it was not.
2009-07-03 15:44 +0000 [r204893-204919] Sean Bright <sean@malleable.com>
* channels/sig_pri.h, channels/chan_dahdi.c, configure,
include/asterisk/autoconfig.h.in, configure.ac,
channels/sig_pri.c: Add a configure check for Reverse Charging
Indication support in LibPRI. Also go back and wrap all of the
places that use the specific reverse charge APIs with
preprocessor conditionals.
* include/asterisk/rtp_engine.h: Wrap rtp_engine.h header comments
to 80 characters.
2009-07-02 22:01 +0000 [r204835] Richard Mudgett <rmudgett@digium.com>
* channels/chan_misdn.c, /: Merged revisions 204834 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r204834 | rmudgett | 2009-07-02 16:59:43 -0500 (Thu, 02
Jul 2009) | 10 lines Removed confusing warning message "Got Busy
in Connected State" If an incoming mISDN call is answered with
the Answer application and a subsequent Dial gets a busy endpoint
then it is valid for that already connected channel to get the
busy indication. Asterisk will play the busy tones until the
dialplan plays something else or hangs up the call. (closes issue
#11974) Reported by: fvdb ........
2009-07-02 20:37 +0000 [r204807] Matthew Nicholson <mnicholson@digium.com>
* main/channel.c, main/features.c: Moved trigger for BRIDGE_END CEL
event so that it is more accurate.
2009-07-02 17:46 +0000 [r204749] Sean Bright <sean@malleable.com>
* channels/sig_pri.h, channels/chan_dahdi.c,
configs/chan_dahdi.conf.sample, funcs/func_channel.c, CHANGES,
channels/sig_pri.c: Support setting and receiving Reverse
Charging Indication over ISDN PRI. This is a continuation of
revision 885 to LibPRI (Capture and expose the Reverse Charging
Indication IE on ISDN PRI) which added the ability to get/set
Reverse Charging Indication in LibPRI. This patch adds the
ability to specify RCI on the outbound leg of a PRI call from
within Asterisk, by prefixing the dialed number with a capital
'C' like: ...,Dial(DAHDI/g1/C4445556666) And to read it off an
inbound channel: exten => s,1,Set(RCI=${CHANNEL(reversecharge)})
Thanks again to rmudgett for the thorough review. (closes issue
#13760) Reported by: mrgabu Review:
https://reviewboard.asterisk.org/r/303/
2009-07-02 16:03 +0000 [r204710] David Vossel <dvossel@digium.com>
* include/asterisk/devicestate.h, main/pbx.c, /,
main/devicestate.c: Merged revisions 204681 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r204681 | dvossel | 2009-07-02 10:05:57 -0500 (Thu, 02 Jul 2009)
| 14 lines Improved mapping of extension states from combined
device states. This fixes a few issues with incorrect extension
states and adds a cli command, core show device2extenstate, to
display all possible state mappings. (closes issue #15413)
Reported by: legart Patches: exten_helper.diff uploaded by
dvossel (license 671) Tested by: dvossel, legart, amilcar Review:
https://reviewboard.asterisk.org/r/301/ ........
2009-07-01 19:47 +0000 [r204654] Ryan Brindley <rbrindley@digium.com>
* configs/http.conf.sample: - cfgbasic.html has been replaced by
index.html in the GUI for some time now
2009-07-01 16:06 +0000 [r204622] Sean Bright <sean@malleable.com>
* apps/app_voicemail.c: A bunch of CODING_GUIDELINES related fixes.
Not even close to done.
2009-06-30 20:41 +0000 [r204563] Tilghman Lesher <tlesher@digium.com>
* /, main/say.c, UPGRADE.txt: Merged revisions 204556 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r204556 | tilghman | 2009-06-30 15:23:51 -0500 (Tue, 30
Jun 2009) | 6 lines More incorrect language codes, plus ensuring
that regionalizations use the specified language, and not English
for grammar. (closes issue #15022) Reported by: greenfieldtech
Patches: 20090519__issue15022.diff.txt uploaded by tilghman
(license 14) ........
2009-06-30 20:39 +0000 [r204561] Sean Bright <sean@malleable.com>
* apps/app_voicemail.c: Remove an unnecessary #ifdef
2009-06-30 19:59 +0000 [r204530-204532] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Move the masquerade in
local_attended_transfer to a point where we hold the channel
lock. Masquerading without the channel's lock held is a
*horrible* idea.
* channels/chan_sip.c: Remove some bogus deadlock avoidance code
from local_attended_transfer. First of all, the code was
unnecessary. The goal was to lock a channel which was already
locked. Second, the assumption of the deadlock avoidance loop was
that the sip_pvt was already locked and we were trying to get the
channel lock. The problem is that the sip_pvt was unlocked a few
lines above. Basically, I'm removing 5 lines of no-op.
2009-06-30 18:48 +0000 [r204475] Jason Parker <jparker@digium.com>
* /, main/say.c: Merged revisions 204474 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r204474 | qwell | 2009-06-30 13:47:06 -0500 (Tue, 30 Jun 2009) |
1 line Fix ast_say_counted_noun to correctly handle Polish. Fix a
comment typo in passing. ........
2009-06-30 18:36 +0000 [r204470] Tilghman Lesher <tlesher@digium.com>
* /, main/say.c, UPGRADE.txt, apps/app_voicemail.c: Recorded merge
of revisions 204469 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r204469 | tilghman | 2009-06-30 13:23:35 -0500 (Tue, 30 Jun 2009)
| 11 lines "tw" is the language specification for Twi (from
Ghana) not Taiwanese. (closes issue #15346) Reported by: volivier
Patches: 20090617__issue15346__1.4.diff.txt uploaded by tilghman
(license 14) 20090617__issue15346__trunk.diff.txt uploaded by
tilghman (license 14) 20090617__issue15346__1.6.0.diff.txt
uploaded by tilghman (license 14)
20090617__issue15346__1.6.1.diff.txt uploaded by tilghman
(license 14) 20090617__issue15346__1.6.2.diff.txt uploaded by
tilghman (license 14) Tested by: volivier ........
2009-06-30 17:22 +0000 [r204417-204440] Russell Bryant <russell@digium.com>
* configs/res_config_sqlite.conf (removed),
configs/res_config_sqlite.conf.sample (added): Rename
res_config_sqlite.conf to res_config_sqlite.conf.sample (missing
.sample).
* addons/chan_ooh323.c, configs/chan_ooh323.conf.sample (added),
configs/ooh323.conf.sample (removed): Rename ooh323.conf to
chan_ooh323.conf, make module support both names
* configs/mobile.conf.sample (removed), addons/chan_mobile.c,
configs/chan_mobile.conf.sample (added): Rename mobile.conf to
chan_mobile.conf, make module support old name, too
* configs/res_config_mysql.conf.sample (added),
configs/res_mysql.conf.sample (removed),
addons/res_config_mysql.c: Rename res_mysql.conf to
res_config_mysql.conf, make module support both
* Makefile: Make addons build last - this is for Qwell.
* addons/app_mysql.c, configs/app_mysql.conf.sample (added),
configs/mysql.conf.sample (removed): Rename mysql.conf to
app_mysql.conf, make module support both names
* addons/Makefile, addons/cdr_mysql.c (added),
addons/cdr_addon_mysql.c (removed): Rename cdr_addon_mysql to
cdr_mysql
* addons/app_mysql.c (added), addons/app_addon_sql_mysql.c
(removed), addons/Makefile: Rename app_addon_sql_mysql to
app_mysql
2009-06-30 17:04 +0000 [r204415] Kevin P. Fleming <kpfleming@digium.com>
* build_tools/embed_modules.xml, Makefile.moddir_rules,
addons/Makefile: Add-ons related build system improvements.
Ensure that add-on modules can be embedded, fix up
Makefile.moddir_rules to allow module directory Makefiles to more
easily specify the modules to be built, and explicitly list the
addons modules in its Makefile, since the module names don't
follow any pattern.
2009-06-30 16:40 +0000 [r204413] Russell Bryant <russell@digium.com>
* autoconf/ast_ext_tool_check.m4, addons/ooh323c/src/oochannels.h,
addons/ooh323c/src/printHandler.h, addons/chan_ooh323.c,
addons/ooh323c/src/ooq931.h, include/asterisk/autoconfig.h.in,
addons/ooh323c/src/ootrace.h, addons/chan_ooh323.h,
addons/ooh323c/src/ooasn1.h, configs/res_mysql.conf.sample
(added), addons/ooh323c/src/ooStackCmds.c,
addons/ooh323c/src/errmgmt.c, addons/ooh323c/src/ooStackCmds.h,
addons/ooh323c/src/eventHandler.c,
addons/ooh323c/src/h323/H235-SECURITY-MESSAGES.h,
addons/mp3/huffman.h, configure,
addons/ooh323c/src/eventHandler.h, addons/ooh323cDriver.c,
include/asterisk/mod_format.h, addons/mp3/interface.c,
doc/tex/asterisk.tex, addons/ooh323cDriver.h,
addons/cdr_addon_mysql.c, addons/ooh323c/src/encode.c,
addons/mp3/MPGLIB_README,
addons/ooh323c/src/h323/H235-SECURITY-MESSAGESEnc.c,
configure.ac, doc/tex/chan_mobile.tex (added),
addons/ooh323c/src/ooports.c, addons/mp3/mpg123.h,
addons/mp3/mpglib.h, addons (added),
addons/ooh323c/src/h323/MULTIMEDIA-SYSTEM-CONTROL.c,
addons/ooh323c/src/ooports.h, addons/ooh323c/src/memheap.c,
Makefile, addons/ooh323c/src/h323/MULTIMEDIA-SYSTEM-CONTROL.h,
addons/ooh323c/src/ooh245.c, addons/mp3/common.c,
addons/ooh323c/src/memheap.h, addons/ooh323c/src/perutil.c,
addons/mp3/decode_i386.c, addons/ooh323c/src/ooh245.h,
addons/mp3/dct64_i386.c, addons/ooh323c/src/ooSocket.c,
addons/ooh323c/src/h323/H235-SECURITY-MESSAGESDec.c,
addons/mp3/layer3.c, addons/ooh323c/src/ooper.h,
addons/ooh323c/src/ooCmdChannel.c, addons/ooh323c/src/ooSocket.h,
addons/ooh323c/src/ooCommon.h, addons/ooh323c/src/ooCmdChannel.h,
addons/ooh323c/COPYING, addons/format_mp3.c,
addons/ooh323c/src/Makefile.in, configs/mobile.conf.sample
(added), addons/ooh323c/src/ootypes.h, addons/mp3,
addons/ooh323c/src/ooLogChan.c, addons/ooh323c/src/ooTimer.c,
addons/ooh323c/src/ooLogChan.h, addons/ooh323c/src/dlist.c,
addons/ooh323c/src/ooCapability.c, addons/ooh323c/src/oohdr.h,
README-addons.txt (added), addons/app_addon_sql_mysql.c,
addons/ooh323c/src/ooTimer.h, addons/ooh323c/src/ooCapability.h,
addons/ooh323c/src/dlist.h, addons/mp3/Makefile, addons/Makefile,
addons/ooh323c/README, addons/ooh323c, doc/tex/cdrdriver.tex,
addons/ooh323c/src/h323/H323-MESSAGESEnc.c, addons/chan_mobile.c,
configs/cdr_mysql.conf.sample (added),
addons/ooh323c/src/ooDateTime.c, addons/ooh323c/src/rtctype.c,
addons/ooh323c/src/ooCalls.c, addons/ooh323c/src/ooGkClient.c,
addons/ooh323c/src/h323, addons/ooh323c/src/ooUtils.c,
addons/ooh323c/src/ooDateTime.h,
addons/ooh323c/src/h323/MULTIMEDIA-SYSTEM-CONTROLEnc.c,
addons/ooh323c/src/rtctype.h, addons/ooh323c/src/ooCalls.h,
configs/mysql.conf.sample (added), addons/ooh323c/src/ooh323ep.c,
addons/ooh323c/src/ooGkClient.h,
addons/ooh323c/src/h323/H323-MESSAGES.c,
addons/ooh323c/src/ooUtils.h, addons/mp3/README, UPGRADE.txt,
addons/mp3/MPGLIB_TODO, addons/ooh323c/src/ooh323ep.h,
addons/ooh323c/src/h323/H323-MESSAGES.h,
addons/mp3/decode_ntom.c, configs/ooh323.conf.sample (added),
addons/ooh323c/src/ooh323.c,
addons/ooh323c/src/h323/H323-MESSAGESDec.c, addons/ooh323c/src,
build_tools/menuselect-deps.in, addons/mp3/tabinit.c,
addons/ooh323c/src/ooh323.h, doc/tex/Makefile,
addons/ooh323c/src/decode.c, addons/ooh323c/src/context.c,
main/file.c,
addons/ooh323c/src/h323/MULTIMEDIA-SYSTEM-CONTROLDec.c,
makeopts.in, addons/ooh323c/src/oochannels.c,
addons/app_saycountpl.c, addons/ooh323c/src/printHandler.c,
addons/ooh323c/src/ooq931.c, addons/ooh323c/src/ootrace.c,
addons/res_config_mysql.c: Move Asterisk-addons modules into the
main Asterisk source tree. Someone asked yesterday, "is there a
good reason why we can't just put these modules in Asterisk?".
After a brief discussion, as long as the modules are clearly set
aside in their own directory and not enabled by default, it is
perfectly fine. For more information about why a module goes in
addons, see README-addons.txt. chan_ooh323 does not currently
compile as it is behind some trunk API updates. However, it will
not build by default, so it should be okay for now.
2009-06-29 23:50 +0000 [r204355] Sean Bright <sean@malleable.com>
* apps/app_meetme.c: A few const changes in app_meetme.c that I
noticed while browsing the source.
2009-06-29 22:50 +0000 [r204247-204301] Mark Michelson <mmichelson@digium.com>
* /, channels/chan_sip.c: Merged revisions 204300 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r204300 | mmichelson | 2009-06-29 17:45:34 -0500 (Mon, 29 Jun
2009) | 9 lines Add error message so that it is clear why a SIP
peer was not processed when a DNS lookup fails on a host or
outboundproxy. (closes issue #13432) Reported by: p_lindheimer
Patches: outboundproxy.patch uploaded by p (license 558) ........
* /, channels/chan_sip.c: Merged revisions 204243,204246 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r204243 | mmichelson | 2009-06-29 16:23:43 -0500 (Mon, 29 Jun
2009) | 22 lines Fix a problem where chan_sip would ignore "old"
but valid responses. chan_sip has had a problem for quite a long
time that would manifest when Asterisk would send multiple SIP
responses on the same dialog before receiving a response. The
problem occurred because chan_sip only kept track of the highest
outgoing sequence number used on the dialog. If Asterisk sent two
requests out, and a response arrived for the first request sent,
then Asterisk would ignore the response. The result was that
Asterisk would continue retransmitting the requests and ignoring
the responses until the maximum number of retransmissions had
been reached. The fix here is to rearrange the code a bit so that
instead of simply comparing the sequence number of the response
to our latest outgoing sequence number, we walk our list of
outstanding packets and determine if there is a match. If there
is, we continue. If not, then we ignore the response. In doing
this, I found a few completely useless variables that I have now
removed. (closes issue #11231) Reported by: flefoll Review:
https://reviewboard.asterisk.org/r/298 ........ r204246 |
mmichelson | 2009-06-29 16:37:05 -0500 (Mon, 29 Jun 2009) | 3
lines Fix build oops. ........
2009-06-29 20:29 +0000 [r204119-204217] Sean Bright <sean@malleable.com>
* configs/cel_adaptive_odbc.conf.sample: Reorganize this adaptive
CEL config a bit.
* apps/app_rpt.c: Get app_rpt compiling again. I doubt seriously
that it actually works. Also, the code in this module is
horrendous and we should remove it from the tree. I'm not sure
who is supposed to be maintaning this thing, but they clearly are
not. I don't see the sense of leaving it in the main tree. If it
lives *anywhere* it should be in addons.
* configs/cel_sqlite3_custom.conf.sample, configs/cel.conf.sample,
configs/cel_adaptive_odbc.conf.sample,
configs/cel_pgsql.conf.sample, configs/cel_custom.conf.sample:
Add common headers to CEL related configs.
2009-06-29 17:56 +0000 [r204069-204118] Tilghman Lesher <tlesher@digium.com>
* main/channel.c, include/asterisk/channel.h: Allow trunk to once
again compile under MALLOC_DEBUG
* configs/cel_adaptive_odbc.conf.sample: Remove invalid entries in
the config. This might seem like a legitimate comment that merely
needed semicolon prefixes, but in reality, the adaptive layer is
designed to allow arbitrary CDR variables, without needing the
use of a userfield to store multiple items. It's therefore not
only invalid syntax but also goes against the intent of the
adaptive method.
2009-06-27 20:26 +0000 [r203985] Sean Bright <sean@malleable.com>
* CHANGES: Another CHANGES spelling fix.
2009-06-27 10:04 +0000 [r203960-203962] Russell Bryant <russell@digium.com>
* main/app.c: Only update total silence counter after a counter
reset. (closes issue #2264) Reported by: pfn Patches:
silent-vm-1.6.2-fix2.txt uploaded by pfn (license 810) Tested by:
pfn
* UPGRADE.txt, CHANGES: Minor tweaks and spelling fixes for CHANGES
and UPGRADE.txt.
2009-06-27 01:07 +0000 [r203909] Richard Mudgett <rmudgett@digium.com>
* /, channels/sig_pri.c: Merged revisions 203908 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r203908 | rmudgett | 2009-06-26 19:55:12 -0500 (Fri, 26 Jun 2009)
| 16 lines The ISDN CPE side should not exclusively pick B
channels normally. Before this patch, Asterisk unconditionally
picked B channels exclusively on the CPE side and normally
allowed alternative B channels on the network side. Now Asterisk
does the opposite. Reasons for the CPE side to normally not pick
B channels exclusively: * For CPE point-to-multipoint mode (i.e.
phone side), the CPE side does not have enough information to
exclusively pick B channels. (There may be other devices on the
line.) * Q.931 gives preference to the network side picking B
channels. * Some telcos require the CPE side to not pick B
channels exclusively. (closes issue #14383) Reported by:
mbrancaleoni ........
2009-06-26 22:11 +0000 [r203853] Jeff Peeler <jpeeler@digium.com>
* channels/chan_dahdi.c, /: Merged revisions 203848 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r203848 | jpeeler | 2009-06-26 17:09:19 -0500 (Fri, 26
Jun 2009) | 5 lines Make sure to recreate the dahdi pseudo
channel after dahdi restart (closes issue #14477) Reported by:
timking ........
2009-06-26 22:08 +0000 [r203846] Sean Bright <sean@malleable.com>
* cdr/cdr_syslog.c (added), build_tools/menuselect-deps.in,
configure, configure.ac, configs/cdr_syslog.conf.sample (added),
CHANGES: Add a new module, cdr_syslog, which allows writing CDRs
to syslog. The original patch for this was written by Brett
Bryant, and I split it out into it's own module. (closes issue
#12876) Reported by: bbryant Patches:
06162008_cdr_custom_syslog.diff uploaded by bbryant (license 36)
05212009_cdr_syslog.patch uploaded by seanbright (license 71)
Tested by: seanbright Review:
https://reviewboard.asterisk.org/r/297/
2009-06-26 21:48 +0000 [r203802-203842] Russell Bryant <russell@digium.com>
* CHANGES, apps/app_chanspy.c: Add 's' option to ChanSpy, which
makes the app exit when no channels are left to spy on. (closes
issue #14594) Reported by: JimDickenson Patches: chanspy.diff
uploaded by JimDickenson (license 710)
* /, main/file.c: Merged revisions 203785 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r203785 | russell | 2009-06-26 16:16:39 -0500 (Fri, 26 Jun 2009)
| 15 lines Don't fast forward past the end of a message. This is
nice change for users of the voicemail application. If someone
gets a little carried away with fast forwarding through a
message, they can easily get to the end and accidentally exit the
voicemail application by hitting the fast forward key during the
following prompt. This adds some safety by not allowing a fast
forward past the end of a message. (closes issue #14554) Reported
by: lacoursj Patches: 21761.patch uploaded by lacoursj (license
707) Tested by: lacoursj ........
2009-06-26 20:52 +0000 [r203783] Mark Michelson <mmichelson@digium.com>
* doc/manager_1_1.txt, main/manager.c: Add timestamp to response to
"Ping" manager action. (closes issue #14596) Reported by:
JimDickenson Patches: pong2.diff uploaded by JimDickenson
(license 710)
2009-06-26 20:45 +0000 [r203779] Russell Bryant <russell@digium.com>
* channels/chan_sip.c: Ensure the TCP read buffer is fully
initialized before handling each packet. (closes issue #14452)
Reported by: umberto71
2009-06-26 20:19 +0000 [r203735] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c, configs/sip.conf.sample, CHANGES: Fix the
'nat' option to actually do RFC3581 as expected and extend the
configurable values for finer control. (closes issue #8855)
Reported by: mikma Tested by: klaus3000, file
2009-06-26 20:13 +0000 [r203721] David Brooks <dbrooks@digium.com>
* apps/app_voicemail.c: Fixing voicemail's error in checking max
silence vs min message length Max silence was represented in
milliseconds, yet vmminsecs (minmessage) was represented as
seconds. Also, the inequality was reversed. The warning, if
triggered, was "Max silence should be less than minmessage or you
may get empty messages", which should have been logged if max
silence was greater than minmessage, but the check was for less
than. Also, conforming if statement to coding guidelines. closes
issue #15331) Reported by: markd Review:
https://reviewboard.asterisk.org/r/293/
2009-06-26 19:47 +0000 [r203710] David Vossel <dvossel@digium.com>
* channels/chan_iax2.c: moving debug message from level 0 to 1.
(closes issue #15404) Reported by: leobrown Patches:
iax_codec_debug.patch uploaded by leobrown (license 541)
2009-06-26 19:31 +0000 [r203702] Russell Bryant <russell@digium.com>
* include/asterisk/devicestate.h, main/pbx.c, main/devicestate.c:
Make invalid hints report Unavailable instead of Idle. (closes
issue #14413) Reported by: pj
2009-06-26 19:27 +0000 [r203699] Joshua Colp <jcolp@digium.com>
* main/channel.c, main/frame.c, main/rtp_engine.c,
channels/chan_sip.c, apps/app_fax.c, configs/sip.conf.sample,
include/asterisk/frame.h: Improve T.38 negotiation by exchanging
session parameters between application and channel.
2009-06-26 19:03 +0000 [r203672] Jeff Peeler <jpeeler@digium.com>
* channels/sig_analog.c: Check if polarityonanswerdelay has elapsed
before setting a channel as answered after a polarity reversal.
Previously on a polarity switch event chan_dahdi would set the
channel immediately as answered. This would cause problems if a
polarity reversal occurred when the line was picked up as the
dial would not have yet occurred. Now if the polarity reversal
occurs before delay has elapsed after coming off hook or an
answer, it is ignored. Also, some refactoring was done in
_handle_event. (closes issue #13917) Reported by: alecdavis
Patches: chan_dahdi.bug13917.feb09.diff2.txt uploaded by
alecdavis (license 585) Tested by: alecdavis
2009-06-26 15:42 +0000 [r203638-203640] Russell Bryant <russell@digium.com>
* include/asterisk/doxyref.h, include/asterisk/channel.h: Note a
new API call, and one that changed in doxygen.
* cel/cel_pgsql.c, configs/cel_sqlite3_custom.conf.sample (added),
cdr/cdr_sqlite3_custom.c, configs/cel.conf.sample (added),
channels/chan_local.c, include/asterisk/cel.h (added),
main/devicestate.c, apps/app_chanisavail.c, channels/chan_iax2.c,
doc/tex/cel-doc.tex (added), main/loader.c, main/cli.c,
channels/chan_dahdi.c, channels/sig_analog.c,
channels/chan_skinny.c, include/asterisk/event_defs.h,
main/features.c, res/ais/evt.c, channels/sig_analog.h,
channels/chan_alsa.c, doc/tex/asterisk.tex, cdr/cdr_manager.c,
apps/app_dial.c, main/pbx.c, include/asterisk/utils.h,
channels/chan_bridge.c, cel/cel_tds.c, channels/chan_agent.c,
configs/cel_adaptive_odbc.conf.sample (added),
include/asterisk/cdr.h, include/asterisk/channel.h, CHANGES,
main/cel.c (added), Makefile, channels/chan_misdn.c,
funcs/func_channel.c, funcs/func_cdr.c, doc/tex/celdriver.tex
(added), main/asterisk.c, cel/cel_adaptive_odbc.c,
apps/app_voicemail.c, res/res_calendar.c,
channels/chan_unistim.c, tests/test_substitution.c,
cel/cel_radius.c, channels/chan_multicast_rtp.c,
channels/chan_vpb.cc, apps/app_meetme.c, channels/chan_gtalk.c,
apps/app_followme.c, configs/cel_tds.conf.sample (added),
main/channel.c, main/cdr.c, channels/chan_phone.c, main/dial.c,
main/manager.c, include/asterisk/event.h,
bridges/bridge_builtin_features.c, funcs/func_odbc.c,
cel/cel_custom.c, cel/cel_manager.c, cdr/cdr_sqlite.c,
res/res_agi.c, apps/app_minivm.c, main/logger.c,
apps/app_confbridge.c, configs/cel_custom.conf.sample (added),
channels/chan_mgcp.c, apps/app_parkandannounce.c,
cdr/cdr_custom.c, channels/chan_sip.c, cel (added),
configs/cel_pgsql.conf.sample (added), channels/chan_console.c,
include/asterisk/_private.h, channels/sig_pri.c,
apps/app_queue.c, channels/chan_oss.c, channels/sig_pri.h,
channels/chan_usbradio.c, channels/chan_jingle.c, cel/Makefile,
apps/app_celgenuserevent.c (added), apps/app_directed_pickup.c,
channels/chan_h323.c, cel/cel_sqlite3_custom.c, main/event.c,
channels/chan_nbs.c: Merge the new Channel Event Logging (CEL)
subsystem. CEL is the new system for logging channel events. This
was inspired after facing many problems trying to represent what
is possible to happen to a call in Asterisk using CDR records.
For more information on CEL, see the built in HTML or PDF
documentation generated from the files in doc/tex/. Many thanks
to Steve Murphy (murf) and Brian Degenhardt (bmd) for their hard
work developing this code. Also, thanks to Matt Nicholson
(mnicholson) and Sean Bright (seanbright) for their assistance in
the final push to get this code ready for Asterisk trunk. Review:
https://reviewboard.asterisk.org/r/239/
2009-06-26 13:00 +0000 [r203569-203605] Sean Bright <sean@malleable.com>
* include/asterisk/syslog.h, main/syslog.c: Add functions to map
syslog facilities and priorities constants to strings. Also
change the default casing of the string contants to lowercase.
This really just saves us from have to lowercase them later when
displaying them.
* configure, include/asterisk/autoconfig.h.in, configure.ac,
main/syslog.c: Add checks in configure for non-POSIX syslog
facilities.
2009-06-26 00:23 +0000 [r203525-203534] Russell Bryant <russell@digium.com>
* main/syslog.c: One more formatting nit ... use spaces for inline
indentation.
* main/syslog.c: Convert spaces to tabs for indentation.
2009-06-25 23:54 +0000 [r203508] Sean Bright <sean@malleable.com>
* include/asterisk/syslog.h (added), main/logger.c, main/syslog.c
(added): Move syslog utility functions into a separate file so
they can be re-used. This has the pleasant side effect of
cleaning up the header inclusion process in logger.c.
2009-06-25 22:48 +0000 [r203479] Jeff Peeler <jpeeler@digium.com>
* channels/chan_dahdi.c: make sure chan_dahdi compiles with only
libss7 and not libpri installed
2009-06-25 21:45 +0000 [r203444] David Vossel <dvossel@digium.com>
* main/ast_expr2.fl, main/ast_expr2.c: fixes a few redundant
conditions (issue #15269)
2009-06-25 21:34 +0000 [r203443] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c: Picking nits
2009-06-25 21:22 +0000 [r203402] Jeff Peeler <jpeeler@digium.com>
* channels/chan_dahdi.c, configs/chan_dahdi.conf.sample: Remove
some unnecessary code and update sample config file with respect
to GR-303.
2009-06-25 21:15 +0000 [r203381] Terry Wilson <twilson@digium.com>
* /, main/cli.c: Merged revisions 203380 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r203380 | twilson | 2009-06-25 16:13:10 -0500 (Thu, 25 Jun 2009)
| 4 lines I didn't see that Mark already fixed the underlying
issue! Yay for removing useless code. ........
2009-06-25 21:04 +0000 [r203376] Russell Bryant <russell@digium.com>
* /, main/features.c: Merged revisions 203375 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r203375 | russell | 2009-06-25 16:02:18 -0500 (Thu, 25 Jun 2009)
| 9 lines Fix a case where CDR answer time could be before the
start time involving parking. (closes issue #13794) Reported by:
davidw Patches: 13794.patch uploaded by murf (license 17)
13794.patch.160 uploaded by murf (license 17) Tested by: murf,
dbrooks ........
2009-06-25 20:25 +0000 [r203338] Terry Wilson <twilson@digium.com>
* /, main/cli.c: Merged revisions 203311 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r203311 | twilson | 2009-06-25 15:09:15 -0500 (Thu, 25 Jun 2009)
| 2 lines Don't try to free NULL ........
2009-06-25 19:54 +0000 [r203304] Jeff Peeler <jpeeler@digium.com>
* channels/sig_pri.h (added), channels/chan_dahdi.c,
channels/sig_analog.c, channels/sig_analog.h, channels/sig_pri.c
(added), channels/Makefile: New signaling module to handle
PRI/BRI operations in chan_dahdi This merge splits the PRI/BRI
signaling logic out of chan_dahdi.c into sig_pri.c. Functionality
in theory should not change (mostly). A few trivial changes were
made in sig_analog with verbose messages and commenting.
2009-06-25 19:22 +0000 [r203258] Jason Parker <jparker@digium.com>
* channels/chan_dahdi.c: Unmute when we get a dtmfup (we muted on
dtmfdown) event. This would occasionally cause one-way audio when
using hardware DTMF detection. (closes issue #14761) Reported by:
tzafrir Patches: v1-14761.patch uploaded by dimas (license 88)
Tested by: tzafrir, dimas
2009-06-25 18:25 +0000 [r203227] Joshua Colp <jcolp@digium.com>
* res/res_rtp_multicast.c (added), channels/chan_multicast_rtp.c
(added), CHANGES: Add support for multicast RTP paging. (closes
issue #11797) Reported by: macbrody Review:
https://reviewboard.asterisk.org/r/270/
2009-06-25 17:01 +0000 [r203188] Sean Bright <sean@malleable.com>
* main/logger.c: Pass a logmsg to ast_log_vsyslog instead of
separate arguments.
2009-06-25 16:18 +0000 [r203126] Doug Bailey <dbailey@digium.com>
* channels/chan_dahdi.c: Insure ring cadence is set for fxs ports
Moved SETCADENCE ioctl call to before call into new analog signal
module to insure that it gets set. (closes issue #15381) Reported
by: alecdavis Patches: fix15381.diff uploaded by dbailey (license
819) Tested by: dbailey
2009-06-25 16:04 +0000 [r203116] Russell Bryant <russell@digium.com>
* /, channels/chan_sip.c: Merged revisions 203115 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r203115 | russell | 2009-06-25 11:02:16 -0500 (Thu, 25 Jun 2009)
| 11 lines Resolve a crash related to a T.38 reinvite race
condition. This change resolves a crash observed locally during
some T.38 testing. A call was set up using a call file, and when
the T.38 reinvite came in, the channel state was still
AST_STATE_DOWN. The reason is explained by a comment in the code
that previously lived in the handling of AST_STATE_RINGING. This
change modifies the logic to handle the same race condition for
any channel state that is not UP. (closes ABE-1895) ........
2009-06-24 21:08 +0000 [r203037] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c, /: Merged revisions 203036 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r203036 | rmudgett | 2009-06-24 16:01:43 -0500 (Wed, 24
Jun 2009) | 8 lines Improved chan_dahdi.conf pritimer error
checking. Valid format is: pritimer=timer_name,timer_value *
Fixed segfault if the ',' is missing. * Completely check the
range returned by pri_timer2idx() to prevent possible access
outside array bounds. ........
2009-06-24 18:29 +0000 [r202967] Mark Michelson <mmichelson@digium.com>
* /, channels/chan_sip.c: Merged revisions 202966 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r202966 | mmichelson | 2009-06-24 13:28:47 -0500 (Wed, 24 Jun
2009) | 3 lines Use the handy UNLINK macro instead of hand-coding
the same thing in-line. ........
2009-06-24 18:08 +0000 [r202925] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: Ensure the default settings are applied for
T.38 when we set it up for a peer.
2009-06-24 13:53 +0000 [r202840-202889] Sean Bright <sean@malleable.com>
* doc/tex: Ignore some files generated when asterisk.pdf is
created.
* configs/cdr_tds.conf.sample, cdr/cdr_tds.c: Update sample cdr_tds
configuration to try and eliminate some confusion. Also change
the preferred configuration option from 'hostname' (which was
misleading because it didn't actually treat the value as a
hostname) to 'connection' and added some verbage explaining that
the user would need to refer to their freetds.conf file for those
settings. 'hostname' was kept as a backwards compatible
configuration parameter.
* doc/tex/billing.tex, doc/tex/cdrdriver.tex: Change some section
names in the CDR tex documentation.
* doc/tex/cdrdriver.tex: Remove some trailing whitespace before
making content changes.
2009-06-23 22:47 +0000 [r202804] Russell Bryant <russell@digium.com>
* doc/tex/cdrdriver.tex: Clean up section hierarchy for the CDR
chapter.
2009-06-23 22:08 +0000 [r202761] Matthew Fredrickson <creslin@digium.com>
* channels/chan_dahdi.c: I could have sworn I committed this patch
ages ago, but... bug fix with setting NAI properly on linksets in
certain situations.
2009-06-23 21:38 +0000 [r202755] Richard Mudgett <rmudgett@digium.com>
* channels/chan_misdn.c: Make outgoing_colp=2 misdn.conf port
parameter not send redirecting or transfer messages. If the
outgoing_colp parameter is set to not send COLP information, then
it does not make sense to send redirecting or transfer messages
announcing new COLP information that is blocked. The service
provider may supply the listed number for that line when it
passes the messages to the next hop. Why tell the switch that
these events happened when the information is otherwise
suppressed? Also blocked the number of previous redirects that
may have occurred to calls going out the port when outgoing_colp
is 2. Follow on to JIRA ABE-1853.
2009-06-23 21:25 +0000 [r202753] Ryan Brindley <rbrindley@digium.com>
* main/config.c: If we delete the info, lets also delete the lines
(closes issue #14509) Reported by: timeshell Patches:
20090504__bug14509.diff.txt uploaded by tilghman (license 14)
Tested by: awk, timeshell
2009-06-23 16:31 +0000 [r202672] David Vossel <dvossel@digium.com>
* /, channels/chan_sip.c: Merged revisions 202671 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r202671 | dvossel | 2009-06-23 11:28:46 -0500 (Tue, 23 Jun 2009)
| 12 lines MWI NOTIFY contains a wrong URI if Asterisk listens to
non-standard port and transport (closes issue #14659) Reported
by: klaus3000 Patches: patch_chan_sip_fixMWIuri_1.4.txt uploaded
by klaus3000 (license 65) mwi_port-transport_trunk.diff uploaded
by dvossel (license 671) Tested by: dvossel, klaus3000 Review:
https://reviewboard.asterisk.org/r/288/ ........
2009-06-23 14:54 +0000 [r202497-202570] Russell Bryant <russell@digium.com>
* main/app.c, CHANGES: Ignore voicemail messages that are just
silence. (closes issue #2264) Reported by: pfn Patches:
silent-vm-1.6.2.txt uploaded by pfn (license 810)
* main/channel.c, /: Merged revisions 202496 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r202496 | russell | 2009-06-22 15:08:53 -0500 (Mon, 22 Jun 2009)
| 4 lines Report CallerID change during a masquerade. Reported
by: markster ........
2009-06-22 16:09 +0000 [r202417] Sean Bright <sean@malleable.com>
* cdr/cdr_sqlite3_custom.c: Fix lock usage in cdr_sqlite3_custom to
avoid potential crashes during reload. Pointed out by Russell
while working on the CEL branch.
2009-06-22 16:05 +0000 [r202415] Russell Bryant <russell@digium.com>
* /, channels/chan_sip.c: Merged revisions 202414 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r202414 | russell | 2009-06-22 11:00:00 -0500 (Mon, 22 Jun 2009)
| 2 lines Make Polycom subscription type override check more
explicit. ........
2009-06-22 15:33 +0000 [r202410] David Vossel <dvossel@digium.com>
* include/asterisk/module.h, main/loader.c: attempting to load
running modules Modules placed in the priority heap for loading
were not properly removed from the linked list. This resulted in
some modules attempting to load twice.
2009-06-22 14:58 +0000 [r202337-202343] Mark Michelson <mmichelson@digium.com>
* /, channels/chan_sip.c: Merged revisions 202341-202342 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r202341 | mmichelson | 2009-06-22 09:42:55 -0500 (Mon, 22 Jun
2009) | 26 lines Fix a situation in which Asterisk would not stop
retransmitting 487s. If a CANCEL were received by Asterisk, we
would send a 487 in response to the original INVITE and a 200 OK
for the CANCEL. If there were a network hiccup which caused the
200 OK and the 487 to be lost, then the UA communicating with
Asterisk may try to retransmit its CANCEL. Asterisk's response to
this used to be to try sending another 487 to the canceled INVITE
and another 200 OK to the CANCEL. The problem here is that the
originally-sent 487 was sent "reliably" meaning that it will be
retransmitted until it is received properly. So when we receive
the second CANCEL it is likely that the first batch of 487s we
sent is still going strong and reaches the UA. The result was
that the second set of 487s would be retransmitted constantly
until the maximum number of retries had been reached. The fix for
this is that if we receive a second CANCEL for an INVITE, then we
cancel the retransmission of the first set of 487s and start a
second set. This causes the dialog to be terminated reasonably.
(closes issue #14584) Reported by: klaus3000 Patches:
14584_v2.patch uploaded by mmichelson (license 60) Tested by:
klaus3000 ........ r202342 | mmichelson | 2009-06-22 09:44:58
-0500 (Mon, 22 Jun 2009) | 3 lines Remove an extra debug line
left from previous commit. ........
* /, channels/chan_sip.c: Merged revisions 202336 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r202336 | mmichelson | 2009-06-22 09:34:05 -0500 (Mon, 22 Jun
2009) | 25 lines Fix a possible infinite loop in SDP parsing
during glare situation. There was a while loop in
get_ip_and_port_from_sdp which was controlled by a call to
get_sdp_iterate. The loop would exit either if what we were
searching for was found or if the return was NULL. The problem is
that get_sdp_iterate never returns NULL. This means that if what
we were searching for was not present, the loop would run
infinitely. This modification of the loop fixes the problem.
(closes issue #15213) Reported by: schmidts (closes issue #15349)
Reported by: samy (closes issue #14464) Reported by: pj (closes
issue #15345) Reported by: aragon Patches: sip_inf_loop.patch
uploaded by mmichelson (license 60) Tested by: aragon ........
2009-06-21 16:36 +0000 [r202223-202301] Russell Bryant <russell@digium.com>
* cdr/cdr_sqlite3_custom.c: Note a bug in cdr_sqlite3_custom so I
don't forget about it.
* cdr/cdr_manager.c: Fix possibility of crashiness during reload in
custom fields handling.
* cdr/cdr_manager.c: Standardize return values of load_config() so
reload() doesn't report an error on success.
* cdr/cdr_manager.c: Leave a note about some unsafe code in
cdr_manager
2009-06-20 19:09 +0000 [r202183] Sean Bright <sean@malleable.com>
* apps/app_fax.c: Fix version detection for API changes in spandsp.
(closes issue #15355) Reported by: deuffy
2009-06-20 14:09 +0000 [r202109] Russell Bryant <russell@digium.com>
* main/cdr.c, cdr/cdr_adaptive_odbc.c, cdr/cdr_pgsql.c: Remove
unnecessary usleep() from a couple of module unload callbacks. In
passing, also tweak cdr_unregister() to hold the list lock a bit
less time.
2009-06-19 21:25 +0000 [r202039] Matthew Nicholson <mnicholson@digium.com>
* channels/chan_sip.c: Use sched_yield() instead of usleep(1)
2009-06-19 20:24 +0000 [r201994] David Vossel <dvossel@digium.com>
* /, channels/chan_iax2.c: Merged revisions 201993 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r201993 | dvossel | 2009-06-19 15:22:02 -0500 (Fri, 19
Jun 2009) | 8 lines timestamp was being converted to host order
as a short rather than a long (closes issue #15361) Reported by:
ffloimair Patches: ts_issue.diff uploaded by dvossel (license
671) ........
2009-06-19 17:40 +0000 [r201944] Terry Wilson <twilson@digium.com>
* CHANGES: Add note about the addition of calendar support
2009-06-19 15:47 +0000 [r201904] Tilghman Lesher <tlesher@digium.com>
* res/res_config_odbc.c: Fix 2 typos and add support for wide
character types. Reported by Benny Amorsen via the asterisk-users
mailing list.
http://lists.digium.com/pipermail/asterisk-users/2009-June/233622.html
2009-06-19 15:41 +0000 [r201902] Joshua Colp <jcolp@digium.com>
* main/rtp_engine.c, channels/chan_sip.c,
include/asterisk/rtp_engine.h: Add support for allowing an RTP
engine to decide on whether it is possible for specific formats
to be transcoded for an RTP instance.
2009-06-19 00:43 +0000 [r201745-201829] Tilghman Lesher <tlesher@digium.com>
* /, main/features.c: Merged revisions 201828 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r201828 | tilghman | 2009-06-18 19:40:41 -0500 (Thu, 18 Jun 2009)
| 6 lines If the "h" extension fails, give it another chance in
main/pbx.c. If the "h" extension fails, give it another chance in
main/pbx.c, when it returns from the bridge code. Fixes an issue
where the "h" extension may occasionally not fire, when a Dial is
executed from a Macro. Debugged in #asterisk with user tompaw.
........
* apps/Makefile: One of the changes in 1.6.1 was to allow
app_directory to use functionality within app_voicemail for
directory functions. It is therefore no longer necessary for
app_directory to be linked against the ODBC libraries (and it
never was necessary for app_directory to be linked against IMAP,
though it was).
* funcs/func_cut.c: Clarify CUT code, and in the process, fix a bug
in trunk only (closes issue #15320) Reported by: chappell
Patches: cut_fix.patch uploaded by chappell (license 8)
cut_clarify.patch uploaded by chappell (license 8)
2009-06-18 17:41 +0000 [r201717] Matthew Nicholson <mnicholson@digium.com>
* channels/chan_sip.c: Added deadlock protection to
try_suggested_sip_codec in chan_sip.c. Review:
https://reviewboard.asterisk.org/r/285/
2009-06-18 16:37 +0000 [r201678] David Vossel <dvossel@digium.com>
* codecs/gsm/src/gsm_destroy.c, channels/h323/ast_h323.cxx,
main/ast_expr2f.c, res/ael/ael_lex.c, utils/ael_main.c,
utils/extconf.c, channels/xpmr/xpmr.c, pbx/pbx_config.c,
res/res_config_ldap.c, apps/app_rpt.c, channels/misdn/isdn_lib.c,
main/asterisk.c, utils/conf2ael.c, main/ast_expr2.c,
utils/stereorize.c: fixes some memory leaks and redundant
conditions (closes issue #15269) Reported by: contactmayankjain
Patches: patch.txt uploaded by contactmayankjain (license 740)
memory_leak_stuff.trunk.diff uploaded by dvossel (license 671)
Tested by: contactmayankjain, dvossel
2009-06-18 15:27 +0000 [r201610] Russell Bryant <russell@digium.com>
* /, res/res_musiconhold.c: Merged revisions 201600 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r201600 | russell | 2009-06-18 10:24:31 -0500 (Thu, 18
Jun 2009) | 29 lines Fix memory corruption and leakage related
reloads of non files mode MoH classes. For Music on Hold classes
that are not files mode, meaning that we are executing an
application that will feed us audio data, we use a thread to
monitor the external application and read audio from it. This
thread also makes use of the MoH class object. In the MoH class
destructor, we used pthread_cancel() to ask the thread to exit.
Unfortunately, the code did not wait to ensure that the thread
actually went away. What needed to be done is a pthread_join() to
ensure that the thread fully cleans up before we proceed. By
adding this one line, we resolve two significant problems: 1)
Since the thread was never joined, it never fully goes away. So,
on every reload of non-files mode MoH, an unused thread was
sticking around. 2) There was a race condition here where the
application monitoring thread could still try to access the MoH
class, even though the thread executing the MoH reload has
already destroyed it. (issue #15109) Reported by: jvandal (issue
#15123) Reported by: axisinternet (issue #15195) Reported by:
amorsen (issue AST-208) ........
2009-06-18 15:20 +0000 [r201583] Mark Michelson <mmichelson@digium.com>
* res/res_rtp_asterisk.c, main/rtp_engine.c, channels/chan_sip.c,
include/asterisk/rtp_engine.h: Trunk implementation of setting an
alternate RTP source. This contains the interface by which we can
let an rtp instance know that it might start receiving audio from
a new source. This is similar in nature to revision 197588 of
Asterisk 1.4. Review: https://reviewboard.asterisk.org/r/276
2009-06-18 15:16 +0000 [r201534-201570] David Vossel <dvossel@digium.com>
* channels/chan_sip.c: parsing extension correctly from sip
register lines If a transport type was specified, but no
extension, parsing of the extension would return whatever was
after the transport rather than defaulting to 's'. (closes issue
#15111) Reported by: ffs Patches:
chan_sip.c_register-parser.patch uploaded by ffs (license 730)
Tested by: ffs, dvossel
* configs/iax.conf.sample, CHANGES, channels/chan_iax2.c: Add
rtsavesysname to chan_iax chan_sip has an option to save the
sysname on rtupdate. This patch copies that same logic to
chan_iax. (closes issue #14837) Reported by: barthpbx Patches:
iax2-rtsavesysname.patch uploaded by barthpbx (license 744)
rt_iax.diff uploaded by dvossel (license 671)
2009-06-17 21:31 +0000 [r201531] Tilghman Lesher <tlesher@digium.com>
* apps/app_voicemail.c: Initialize additional variables, to prevent
a possible crash. (closes issue #15186) Reported by: ajohnson
Patches: 20090528__issue15186.diff.txt uploaded by tilghman
(license 14) Tested by: ajohnson
2009-06-17 20:10 +0000 [r201458-201462] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Fix problem with no audio due to ignoring
the SDP. A recent change to our SDP version comparison made audio
not function on some calls. This was because of a test wherein we
were trying to see if an unsigned value was less than 0. This is
a dumb comparison and arguably the compiler should have warned
about it. Alas, though, it slipped past. Now it's fixed by
changing the variable to be a signed type. Found by several
developers. Tested by mnicholson and dbrooks.
* main/channel.c, /: Merged revisions 201450 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r201450 | mmichelson | 2009-06-17 14:59:31 -0500 (Wed, 17 Jun
2009) | 9 lines Change the datastore traversal in
ast_do_masquerade to use a safe list traversal. It is possible
for datastore fixup functions to remove the datastore from the
list and free it. In particular, the queue_transfer_fixup in
app_queue does this. While I don't yet know of this causing any
crashes, it certainly could. Found while discussing a separate
issue with Brian Degenhardt. ........
2009-06-17 20:00 +0000 [r201445-201453] David Vossel <dvossel@digium.com>
* doc/datastores.txt: ast_channel_datastore_alloc is no longer
used. updating datastores.txt to reflect that.
* /, apps/app_mixmonitor.c: Merged revisions 201423 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r201423 | dvossel | 2009-06-17 14:28:12 -0500 (Wed, 17
Jun 2009) | 19 lines StopMixMonitor race condition (not giving up
file immediately) StopMixMonitor only indicates to the MixMonitor
thread to stop writing to the file. It does not guarantee that
the recording's file handle is available to the dialplan
immediately after execution. This results in a race condition. To
resolve this, the filestream pointer is placed in a datastore on
the channel. When StopMixMonitor is called, the datastore is
retrieved from the channel and the filestream is closed
immediately before returning to the dialplan. Documentation
indicating the use of StopMixMonitor to free files has been
updated as well. (closes issue #15259) Reported by: travisghansen
Tested by: dvossel Review:
https://reviewboard.asterisk.org/r/283/ ........
2009-06-17 19:15 +0000 [r201381] David Brooks <dbrooks@digium.com>
* /, channels/chan_sip.c: Merged revisions 201380 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r201380 | dbrooks | 2009-06-17 13:45:50 -0500 (Wed, 17 Jun 2009)
| 9 lines Checks for NULL sip_pvt pointer in
chan_sip.c->acf_channel_read() Zombie channels could be passed,
and chan_sip.c wasn't checking for it. Could crash Asterisk. Now
checking for NULL pointer. (closes issue #15330) Reported by:
okrief Tested by: dbrooks ........
2009-06-17 15:20 +0000 [r201331-201344] David Vossel <dvossel@digium.com>
* channels/chan_sip.c: SIP registry ref count error During a sip
reload, the list of sip_registry objects are supposed to be
traversed, unlinked, and destroyed, but destruction never takes
place due to a ref counting error. This causes a memory leak when
registry items are removed from sip.conf and reloaded. While the
registries are removed from the global list, they are not removed
from the scheduler. Because of this, SIP register attempts
continue to be sent out for the item even though it may no longer
be in the .conf. (closes issue #15295) Reported by: amorsen
Review: https://reviewboard.asterisk.org/r/282/
* channels/chan_iax2.c: update chan_iax to use 64bit feature flags.
(closes issue #15335) Reported by: lmadsen Review:
https://reviewboard.asterisk.org/r/284/
2009-06-17 12:04 +0000 [r201262] Kevin P. Fleming <kpfleming@digium.com>
* /, include/asterisk/linkedlists.h: Merged revisions 201261 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r201261 | kpfleming | 2009-06-17 07:03:25 -0500 (Wed, 17 Jun
2009) | 9 lines Correct AST_LIST_APPEND_LIST behavior when list
to be appended is empty. When the list to be appended is empty,
and the list to be appended to is *not*, AST_LIST_APPEND_LIST
would actually cause the target list to become broken, and no
longer have a pointer to its last entry. This patch fixes the
problem. (reported by Stanislaw Pitucha on the asterisk-dev
mailing list) ........
2009-06-16 22:29 +0000 [r201223] David Vossel <dvossel@digium.com>
* channels/chan_sip.c: fix issue with build_contact introduced by
the "SIP trasnport type issues" commit
2009-06-16 22:11 +0000 [r201190] Sean Bright <sean@malleable.com>
* CREDITS: Update my e-mail address (thanks for the props, russell
:))
2009-06-16 21:10 +0000 [r200985-201139] Kevin P. Fleming <kpfleming@digium.com>
* channels/chan_dahdi.c, channels/chan_sip.c, apps/app_fax.c,
include/asterisk/frame.h: Enable applications to enable/disable
digit and tone detection. Some applications (notably app_fax) do
not need digit detection nor FAX tone detection while they are
running, and if Asterisk is using software DSPs to provide the
detection, this consumes extra CPU cycles that could be better
spent on the actual application. This patch allows applications
to query and control the state of digit and tone detection on a
channel, and modifies app_fax to disable them while the FAX
operations are occurring (and re-enable digit detection
afterwards).
* configure, configure.ac: Explicitly test for 'static weakref'
support. Since we use 'static' weakref symbols, and not all GCC
versions support them, test for that combination explicitly.
* Makefile: When compiling in an Emacs-spawned shell, always print
directory names. This change ensures that Emacs can find the
proper source files when parsing compiler error messages, since
it uses the 'make' output including directory names to do it.
* configure, autoconf/ast_gcc_attribute.m4, configure.ac: Another
minor fix to compiler attribute checking. Defaulting to 'static'
for the function scope was bad... so remove it.
* main/channel.c, main/autoservice.c, main/frame.c, /,
apps/app_meetme.c, main/slinfactory.c,
include/asterisk/linkedlists.h, main/file.c,
include/asterisk/channel.h, include/asterisk/frame.h,
apps/app_chanspy.c, apps/app_mixmonitor.c: Merged revisions
200991 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r200991 | kpfleming | 2009-06-16 12:05:38 -0500 (Tue, 16 Jun
2009) | 11 lines Improve support for media paths that can
generate multiple frames at once. There are various media paths
in Asterisk (codec translators and UDPTL, primarily) that can
generate more than one frame to be generated when the application
calling them expects only a single frame. This patch addresses a
number of those cases, at least the primary ones to solve the
known problems. In addition it removes the broken TRACE_FRAMES
support, fixes a number of bugs in various frame-related API
functions, and cleans up various code paths affected by these
changes. https://reviewboard.asterisk.org/r/175/ ........
* configure, autoconf/ast_gcc_attribute.m4, configure.ac: Fix
problems with new compiler attribute checking in configure
script. The last changes to ast_gcc_attribute.m4 caused some
problems checking for various attributes, because the scope of
the symbol the attribute is applied to can be important; this
patch allows the scope to be specified for the check.
2009-06-16 16:03 +0000 [r200946] David Vossel <dvossel@digium.com>
* channels/chan_sip.c: SIP transport type issues What this patch
addresses: 1. ast_sip_ouraddrfor() by default binds to the UDP
address/port reguardless if the sip->pvt is of type UDP or not.
Now when no remapping is required, ast_sip_ouraddrfor() checks
the sip_pvt's transport type, attempting to set the address and
port to the correct TCP/TLS bindings if necessary. 2. It is not
necessary to send the port number in the Contact header unless
the port is non-standard for the transport type. This patch fixes
this and removes the todo note. 3. In sip_alloc(), the default
dialog built always uses transport type UDP. Now sip_alloc()
looks at the sip_request (if present) and determines what
transport type to use by default. 4. When changing the transport
type of a sip_socket, the file descriptor must be set to -1 and
in some cases the tcptls_session's ref count must be decremented
and set to NULL. I've encountered several issues associated with
this process and have created a function, set_socket_transport(),
to handle the setting of the socket type. (closes issue #13865)
Reported by: st Patches: dont_add_port_if_tls.patch uploaded by
Kristijan (license 753) 13865.patch uploaded by mmichelson
(license 60) tls_port_v5.patch uploaded by vrban (license 756)
transport_issues.diff uploaded by dvossel (license 671) Tested
by: mmichelson, Kristijan, vrban, jmacz, dvossel Review:
https://reviewboard.asterisk.org/r/278/
2009-06-16 15:51 +0000 [r200943] Michiel van Baak <michiel@vanbaak.info>
* apps/app_voicemail.c: add FILE_STORAGE to Voicemail Build Options
Voicemail can only use one storage module at the moment. Because
it's unclear that selecting one of the storage modules in
menuselect will disable filesystem storage we now have a
FILE_STORAGE option that conflicts with the other modules.
(closes issue #15333)
2009-06-16 15:26 +0000 [r200942] Russell Bryant <russell@digium.com>
* CREDITS: Add Sean Bright to CREDITS - Thanks, Sean!
2009-06-16 14:12 +0000 [r200841-200878] Eliel C. Sardanons <eliels@gmail.com>
* /: Recorded merge of revisions 200875 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r200875 | eliel | 2009-06-16 09:25:51 -0400 (Tue, 16 Jun 2009) |
5 lines Show the interface name on error, if it is not found. If
the smdiport specified is not found, show the interface name
instead of '(null)'. ........
* res/res_smdi.c: Show the interface name on error, if it is not
found. If the smdiport specified is not found, show the interface
name instead of '(null)'.
2009-06-16 02:32 +0000 [r200805] Russell Bryant <russell@digium.com>
* main/manager.c: Don't claim a char * is a mansession object.
Since there was only 1 bucket, and no hash function was
specified, the code actually worked perfectly fine. However, in
theory, this was invalid use of the OBJ_POINTER flag, so remove
it so the code provides a better usage example.
2009-06-16 02:24 +0000 [r200799] Moises Silva <moises.silva@gmail.com>
* channels/chan_dahdi.c, configs/chan_dahdi.conf.sample: keep
backwards compatible chan_dahdi with older openr2 versions by not
using the new skip category feature unless supported
2009-06-16 01:28 +0000 [r200764] Kevin P. Fleming <kpfleming@digium.com>
* configure, autoconf/ast_gcc_attribute.m4: Ensure that
configure-script testing for compiler attributes actually works.
The configure script tests for compiler attributes didn't
actually enable enough warnings or provide a proper test harness
to determine whether the compiler supports the attribute in
question or not; this caused gcc 4.1 to report that it supports
'weakref', but it doesn't actually support it in the way that is
needed for our optional API mechanism. The new configure script
test will properly distinguish between full support and partial
support for this attribute, among others.
2009-06-16 01:26 +0000 [r200762] Russell Bryant <russell@digium.com>
* doc/tex/channelvariables.tex: Add missing closure of verbatim
environment.
2009-06-16 01:03 +0000 [r200519-200726] Kevin P. Fleming <kpfleming@digium.com>
* CHANGES: Document the new automatic 'ignoresdpversion' behavior.
Asterisk will now automatically ignore incorrect incoming SDP
version numbers when necessary to complete a T.38 re-INVITE
operation.
* channels/chan_sip.c: Accept T.38 re-INVITE responses with invalid
SDP versions. This commit changes the 'incoming SDP version'
check logic a bit more; when 'ignoresdpversion' is *not* set for
a peer, if we initiate a re-INVITE to switch to T.38, we'll
always accept the peer's SDP response, even if they don't
properly increment the SDP version number as they should. If this
situation occurs, a warning message will be generated suggesting
that the peer's configuration be changed to include the
'ignoresdpversion' configuration option (although ideally they'd
fix their SIP implementation to be RFC compliant). AST-221
* doc/CODING-GUIDELINES, apps/app_read.c, apps/app_page.c,
apps/app_fax.c, apps/app_readexten.c, apps/app_queue.c,
include/asterisk/app.h, apps/app_skel.c, apps/app_minivm.c,
apps/app_macro.c, apps/app_url.c, apps/app_sms.c,
apps/app_externalivr.c, apps/app_stack.c, apps/app_mixmonitor.c,
apps/app_voicemail.c: Last batch of 'static' qualifiers for
module-level global variables. Fix up modules in the 'apps'
directory, and also correct the bad example of enum definitions
in include/asterisk/app.h, which many developers followed (thanks
for reading the documentation!). In addition, add some basic
usage examples of the 'pahole' and 'pglobal' tools to the coding
guidelines.
* res/res_snmp.c, main/devicestate.c, funcs/func_vmcount.c,
res/res_calendar_caldav.c, formats/format_wav_gsm.c,
res/res_jabber.c, main/loader.c, main/cli.c, funcs/func_enum.c,
main/manager.c, res/res_smdi.c, funcs/func_odbc.c,
main/features.c, main/logger.c, main/http.c, pbx/pbx_realtime.c,
main/image.c, main/db.c, cdr/cdr_manager.c,
res/res_calendar_exchange.c, res/res_calendar_icalendar.c,
res/res_config_pgsql.c, funcs/func_lock.c, pbx/pbx_lua.c,
funcs/func_cut.c, include/asterisk/calendar.h,
funcs/func_realtime.c, funcs/func_curl.c, funcs/func_cdr.c,
funcs/func_channel.c, main/file.c, main/event.c, pbx/pbx_dundi.c,
main/xmldoc.c, res/res_calendar.c: More 'static' qualifiers on
module global variables. The 'pglobal' tool is quite handy indeed
:-)
* channels/chan_dahdi.c, channels/chan_misdn.c,
channels/chan_sip.c, channels/chan_skinny.c,
channels/chan_agent.c, channels/chan_h323.c,
channels/chan_iax2.c: Convert a number of global module variables
to 'static'. These modules all contained variables that are
module-global but not system-global, but were not marked
'static'.
* channels/chan_sip.c: Some minor structure size improvements in
sip_pvt and sip_peer. Using the 'pahole' tool, it is now quite
easy to see where structure fields could be organized differently
to keep the compiler from having to add padding to satisfy
alignment requirements. These changes reduced the sizes of
sip_pvt and sip_peer by a few bytes each (on 64-bit platforms),
and also fixed a spelling error in a field name.
* include/asterisk/agi.h, main/Makefile,
include/asterisk/autoconfig.h.in, res/res_smdi.exports,
configure.ac, res/res_agi.exports, include/asterisk/compiler.h,
apps/app_queue.c, res/res_monitor.c,
include/asterisk/optional_api.h, Makefile, res/res_smdi.c,
configure, res/res_agi.c, include/asterisk/monitor.h,
apps/app_stack.c, include/asterisk/smdi.h,
res/res_monitor.exports, apps/app_voicemail.c: Redesigned
'optional API' support. This patch provides a new implementation
of the optional API support defined in asterisk/optional_api.h;
this new version provides solves compatibility issues with the
use of linker version scripts for suppressing global symbols. In
addition, there is now a functional (and tested!) implementation
for Mac OS/X, so module writers no longer need to use special
tests before calling optional API functions. All future
implementations must provide these same semantics, so that module
writers can rely on them.
2009-06-15 15:22 +0000 [r200514] Mark Michelson <mmichelson@digium.com>
* /, channels/chan_sip.c: Merged revisions 200513 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r200513 | mmichelson | 2009-06-15 10:21:46 -0500 (Mon, 15 Jun
2009) | 5 lines Add INFO to our allowed methods so that endpoints
know they may send it to us. AST-223 ........
2009-06-14 06:13 +0000 [r200477] Moises Silva <moises.silva@gmail.com>
* channels/chan_dahdi.c, configs/chan_dahdi.conf.sample,
build_tools/menuselect-deps.in: added openr2 to
menuselect-deps.in, recent commit in menuselect made me realize
this was never done but was working anyways also added support
for skip category request feature of openr2 and updated
chan_dahdi.conf.sample
2009-06-12 19:46 +0000 [r200428-200430] Sean Bright <sean@malleable.com>
* contrib/upstart/asterisk.upstart-0.3.9: Include basic
installation and usage instructions for upstart script.
* contrib/upstart/asterisk.upstart-0.3.9 (added), contrib/upstart
(added): First shot at an upstart script for asterisk on Ubuntu.
This works relatively well (assuming you are using
/var/run/asterisk) as your run directory and upstart 0.3.9. Needs
to be generalized and eventually added to the 'make install'
target for Ubuntu.
2009-06-12 19:07 +0000 [r200290-200361] Mark Michelson <mmichelson@digium.com>
* main/channel.c, /: Merged revisions 200360 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r200360 | mmichelson | 2009-06-12 14:06:41 -0500 (Fri, 12 Jun
2009) | 10 lines Suppress a warning message and give a better
return code when generating inband ringing after a call is
answered. (closes issue #15158) Reported by: madkins Patches:
15158.patch uploaded by mmichelson (license 60) Tested by:
madkins ........
* channels/chan_local.c, apps/app_queue.c: Fix some bad locking
stemming from trying to forward a call to a non-existent
extension from a queue.
* apps/app_queue.c: Fix a potential crash from trying to access a
NULL channel pointer.
2009-06-12 02:20 +0000 [r200254] Sean Bright <sean@malleable.com>
* contrib/init.d/rc.debian.asterisk: Call chgrp instead of chown
when setting run directory group ownership. (issue #13153)
Reported by: pabelanger
2009-06-11 21:17 +0000 [r200146] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Fix a crash due to a potentially NULL
p->options. Thanks to mnicholson for pointing it out.
2009-06-11 15:40 +0000 [r200108] Eliel C. Sardanons <eliels@gmail.com>
* main/channel.c: Release the allocated channel decreasing the
reference counter. When allocating the channel use ao2_ref(-1) to
release it, instead of calling ast_free(). Also avoid freeing
structures inside that channel (on error) if they will be
released by the channel destructor being called if the reference
counter reachs 0.
2009-06-11 12:15 +0000 [r200039] Leif Madsen <lmadsen@digium.com>
* build_tools/make_version_c, build_tools/make_version_h: Fix path
for .flavor and .version (issue #14737) Reported by: davidw
Patches: flavor.patch uploaded by davidw (license 780) Tested by:
davidw
2009-06-10 20:40 +0000 [r200000] Sean Bright <sean@malleable.com>
* sample.call: Remove some trailing whitespace and steal revision
200000.
2009-06-10 20:15 +0000 [r199958] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Only try to use the invite_branch on
outgoing INVITEs with auth credentials. I have added a comment to
the code to help ease understanding of the logic here as well.
2009-06-10 20:00 +0000 [r199957] David Brooks <dbrooks@digium.com>
* main/pbx.c: Fixes the argument order in definition of
new_find_extension(). In the definition of new_find_extension(),
the arguments 'callerid' and 'label' were swapped. The prototype
declaration and all calls to the function are ordered 'callerid'
then 'label', but the function itself was ordered 'label' then
'callerid'. (closes issue #15303) Reported by: JimDickenson
2009-06-10 18:58 +0000 [r199923] Mark Michelson <mmichelson@digium.com>
* main/channel.c: Use ast_channel_unref to instead of ast_free on a
newly created channel. Also I removed an unnecessary free of a
cid_name. This will be freed properly in the channel destructor.
Reported by mnicholson in #asterisk-dev.
2009-06-10 16:10 +0000 [r199857] Sean Bright <sean@malleable.com>
* include/asterisk/utils.h, /: Merged revisions 199856 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r199856 | seanbright | 2009-06-10 12:08:35 -0400 (Wed,
10 Jun 2009) | 2 lines __WORDSIZE is not available on all
platforms, so use sizeof(void *) instead. ........
2009-06-09 20:47 +0000 [r199818] David Vossel <dvossel@digium.com>
* channels/chan_sip.c: CLI NOTIFY sending wrong transport type.
SIP's cli NOTIFY command only used UDP rather than copying the
transport type from the peer. (closes issue #15283) Reported by:
jthurman Patches: sip-notify-tcp-svn199728.patch uploaded by
jthurman (license 614) Tested by: jthurman, dvossel
2009-06-09 18:08 +0000 [r199781] Sean Bright <sean@malleable.com>
* Makefile: Fix all of the parallel build warnings issued when
running make -j#.
2009-06-09 16:22 +0000 [r199743] David Vossel <dvossel@digium.com>
* res/res_timing_pthread.c, include/asterisk/module.h,
res/res_timing_dahdi.c, res/res_timing_timerfd.c, main/loader.c:
module load priority This patch adds the option to give a module
a load priority. The value represents the order in which a
module's load() function is initialized. The lower the value, the
higher the priority. The value is only checked if the
AST_MODFLAG_LOAD_ORDER flag is set. If the AST_MODFLAG_LOAD_ORDER
flag is not set, the value will never be read and the module will
be given the lowest possible priority on load. Since some modules
are reliant on a timing interface, the timing modules have been
given a high load priorty. (closes issue #15191) Reported by:
alecdavis Tested by: dvossel Review:
https://reviewboard.asterisk.org/r/262/
2009-06-08 22:08 +0000 [r199696] Tilghman Lesher <tlesher@digium.com>
* doc/janitor-projects.txt: Add sigaction janitor
2009-06-08 19:33 +0000 [r199630] Sean Bright <sean@malleable.com>
* include/asterisk/utils.h, /: Merged revisions 199626,199628 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r199626 | seanbright | 2009-06-08 15:24:32 -0400 (Mon, 08 Jun
2009) | 21 lines Increase the size of our thread stack on 64 bit
processors. We were setting the stack size for each thread to
240KB regardless of architecture, which meant that in some
scenarios we actually had less available stack space on 64 bit
processors (pointers use 8 bytes instead of 4). So now we
calculate the stack size we reserve based on the platform's
__WORDSIZE, which gives us: 32 bit -> 240KB 64 bit -> 496KB 128
bit -> 1008KB (that's right, we're ready for 128 bit processors)
Patch typed by me but written by several members of
#asterisk-dev, including Kevin, Tilghman, and Qwell. (closes
issue #14932) Reported by: jpiszcz Patches:
06052009_issue14932.patch uploaded by seanbright (license 71)
Tested by: seanbright ........ r199628 | seanbright | 2009-06-08
15:28:33 -0400 (Mon, 08 Jun 2009) | 2 lines Fix a typo in the
stack size calculation just introduced. ........
2009-06-08 17:32 +0000 [r199588] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Fix a deadlock that could occur when setting
rtp stats on SIP calls. (closes issue #15143) Reported by:
cristiandimache Patches: 15143.patch uploaded by mmichelson
(license 60) Tested by: cristiandimache
2009-06-07 19:15 +0000 [r199514-199547] Eliel C. Sardanons <eliels@gmail.com>
* apps/app_osplookup.c: Move OSP* applications static documentation
to XML. Move OSP* applications static documentation to the new
AstXML form. (closes issue #15245) Reported by: eliel Patches:
app_osplookup_static_conversion.txt uploaded by lmadsen (license
10)
* apps/app_externalivr.c: Move application ExternalIVR static
documentation to XML. Move application ExternalIVR static
documentation to the new AstXML form. (issue #15245) Reported by:
eliel Patches: app_externalivr.diff uploaded by eliel (license
64)
2009-06-07 14:55 +0000 [r199479] Russell Bryant <russell@digium.com>
* apps/app_dial.c, apps/app_dahdibarge.c, apps/app_dictate.c,
apps/app_authenticate.c, apps/app_echo.c, apps/app_fax.c,
apps/app_dahdiras.c, apps/app_disa.c, apps/app_alarmreceiver.c,
apps/app_chanisavail.c, apps/app_exec.c, apps/app_db.c,
apps/app_controlplayback.c, apps/app_channelredirect.c,
apps/app_directed_pickup.c, apps/app_dumpchan.c, apps/app_amd.c,
apps/app_confbridge.c, apps/app_directory.c, apps/app_chanspy.c,
apps/app_adsiprog.c: Global var cleanup - constification and
removing unused vars.
2009-06-06 23:28 +0000 [r199374-199446] Eliel C. Sardanons <eliels@gmail.com>
* apps/app_stack.c: Move AGI command 'gosub' static documentation
to XML. Move AGI command 'gosub' statis documentation to the new
AstXML form. (issue #15245) Reported by: eliel Patches:
app_stack_static_conversion.txt uploaded by lmadsen (license 10)
(with minor changes by me)
* res/res_musiconhold.c: Move music on hold related applications
documentation to XML. Move MusicOnHold, SetMusicOnHold,
StartMusicOnHold, StopMusicOnHold static documentation to the new
AstXML form. (issue #15245) Reported by: eliel Patches:
res_musiconhold_static_conversion.txt uploaded by lmadsen
(license 10) (with some fixes and formatting by me)
* res/res_phoneprov.c: Move function PP_EACH_USER and
PP_EACH_EXTENSION documentation to XML. Move function
PP_EACH_USER and PP_EACH_EXTENSION documentation to the new
AstXML form. (issue #15245) Reported by: eliel Patches:
res_phoneprov_static_conversion.txt uploaded by lmadsen (license
10) (with PP_EACH_USER add by me)
* apps/app_meetme.c: Move function MEETME_INFO documentation to
XML. Move function MEETME_INFO static documentation to the new
AstXML form. (issue #15245) Reported by: eliel Patches:
app_meetme_static_conversion.txt uploaded by lmadsen (license 10)
* apps/app_minivm.c: Move function MINIVMACCOUNT and MINIVMCOUNTER
static documentation to XML. Move function MINIVMACCOUNT and
MINIVMCOUNTER statis documentation to the new AstXML form. (issue
#15245) Reported by: eliel Patches:
app_minivm_static_conversion.txt uploaded by lmadsen (license 10)
(with minor changes by me)
* funcs/func_sysinfo.c: Move function SYSINFO documentation to XML.
Move function SYSINFO static documentation to the new AstXML
form. (issue #15245) Reported by: eliel Patches:
func_sysinfo_static_conversion.txt uploaded by lmadsen (license
10)
2009-06-06 21:42 +0000 [r199368-199372] Russell Bryant <russell@digium.com>
* apps/app_jack.c: minor tweak
* apps/app_jack.c: Constify a string and strip trailing whitespace.
* Makefile: Switch from "echo -n" to printf. On my mac, the -n was
just getting printed out.
2009-06-05 21:21 +0000 [r199298] David Vossel <dvossel@digium.com>
* include/asterisk/devicestate.h, /, main/devicestate.c: Merged
revisions 199297 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r199297 | dvossel | 2009-06-05 16:19:56 -0500 (Fri, 05 Jun 2009)
| 14 lines Fixes issue with hints giving unexpected results.
Hints with two or more devices that include ONHOLD gave
unexpected results. (closes issue #15057) Reported by:
p_lindheimer Patches: onhold_trunk.diff uploaded by dvossel
(license 671) pbx.c.1.4.patch uploaded by p (license 558)
devicestate.c.trunk.patch uploaded by p (license 671) Tested by:
p_lindheimer, dvossel Review:
https://reviewboard.asterisk.org/r/254/ ........
2009-06-05 13:51 +0000 [r199227] Mark Michelson <mmichelson@digium.com>
* channels/chan_dahdi.c: Correct "dahdi show channels" output when
specifying a group. Since a DAHDI channel may belong to multiple
groups, we need to use a bitwise and instead of equivalence to
determine whether to display the channel information. (closes
issue #15248) Reported by: gentian Patches: 15248.patch uploaded
by mmichelson (license 60) Tested by: gentian
2009-06-04 19:10 +0000 [r199139] David Vossel <dvossel@digium.com>
* /, channels/chan_iax2.c: Merged revisions 199138 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r199138 | dvossel | 2009-06-04 14:00:15 -0500 (Thu, 04
Jun 2009) | 3 lines Additional updates to AST-2009-001 ........
2009-06-04 16:29 +0000 [r199091] Eliel C. Sardanons <eliels@gmail.com>
* res/res_smdi.c: Move static docs to the new AstXML form. Move
SMDI_MSG and SMDI_MSG_RETRIEVE functions statis documentation to
XML. (issue #15245) Reported by: eliel Patches:
res_smdi_static_conversion.txt uploaded by lmadsen (license 10)
2009-06-04 14:31 +0000 [r199051] Sean Bright <sean@malleable.com>
* /, include/asterisk/_private.h, main/asterisk.c, main/loader.c:
Merged revisions 199022 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r199022 | seanbright | 2009-06-04 10:14:57 -0400 (Thu, 04 Jun
2009) | 40 lines Safely handle AMI connections/reload requests
that occur during startup. During asterisk startup, a lock on the
list of modules is obtained by the primary thread while each
module is initialized. Issue 13778 pointed out a problem with
this approach, however. Because the AMI is loaded before other
modules, it is possible for a module reload to be issued by a
connected client (via Action: Command), causing a deadlock. The
resolution for 13778 was to move initialization of the manager to
happen after the other modules had already been lodaded. While
this fixed this particular issue, it caused a problem for users
(like FreePBX) who call AMI scripts via an #exec in a
configuration file (See issue 15189). The solution I have come up
with is to defer any reload requests that come in until after the
server is fully booted. When a call comes in to ast_module_reload
(from wherever) before we are fully booted, the request is added
to a queue of pending requests. Once we are done booting up, we
then execute these deferred requests in turn. Note that I have
tried to make this a bit more intelligent in that it will not
queue up more than 1 request for the same module to be reloaded,
and if a general reload request comes in ('module reload') the
queue is flushed and we only issue a single deferred reload for
the entire system. As for how this will impact existing
installations - Before 13778, a reload issued before module
initialization was completed would result in a deadlock. After
13778, you simply couldn't connect to the manager during startup
(which causes problems with #exec-that-calls-AMI configuration
files). I believe this is a good general purpose solution that
won't negatively impact existing installations. (closes issue
#15189) (closes issue #13778) Reported by: p_lindheimer Patches:
06032009_15189_deferred_reloads.diff uploaded by seanbright
(license 71) Tested by: p_lindheimer, seanbright Review:
https://reviewboard.asterisk.org/r/272/ ........
2009-06-03 20:30 +0000 [r198824-198954] David Vossel <dvossel@digium.com>
* apps/app_dial.c, main/channel.c, apps/app_queue.c:
ast_call_forward() todo notes and originate flag copy.
* main/channel.c, main/features.c, include/asterisk/channel.h:
Generic call forward api, ast_call_forward() The function
ast_call_forward() forwards a call to an extension specified in
an ast_channel's call_forward string. After an ast_channel is
called, if the channel's call_forward string is set this function
can be used to forward the call to a new channel and terminate
the original one. I have included this api call in both
channel.c's ast_request_and_dial() and feature.c's
feature_request_and_dial(). App_dial and app_queue already
contain call forward logic specific for their application and
options. (closes issue #13630) Reported by: festr Review:
https://reviewboard.asterisk.org/r/271/
* channels/chan_iax2.c: fixes issue with channels not going down
after transfer Iax2 currently does not support native bridging if
the timeoutms value is set. We check for that in iax2_bridge, but
then set timeoutms to 0 by default. If the timeoutms is not
provided it is set to -1. By setting timeoutms to 0 it is
processed causing a bridging retry loop. (closes issue #15216)
Reported by: oxymoron Tested by: dvossel
2009-06-02 13:48 +0000 [r198762-198791] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c, configs/sip.conf.sample: Correct
documentation for the register line, specifically where the
domain should be specified. (closes issue #14367) Reported by:
Nick_Lewis
* main/rtp_engine.c: Fix a bug where we were passing in address
information that should remain unmodified to a function that may
modify it. (closes issue #15243) Reported by: pj
2009-06-01 21:03 +0000 [r198729] Russell Bryant <russell@digium.com>
* channels/iax2-parser.c: Tell the IAX2 parser about more control
frame types.
2009-06-01 20:57 +0000 [r198727] Mark Michelson <mmichelson@digium.com>
* apps/app_dial.c, main/channel.c, include/asterisk/app.h,
main/dial.c, channels/chan_sip.c, apps/app_directed_pickup.c,
main/features.c, apps/app_macro.c, doc/tex/channelvariables.tex,
main/app.c, include/asterisk/channel.h, apps/app_queue.c: Add the
ability to execute connected line interception macros. When
connected line updates are received or generated in the middle of
an application call, it is now possible to execute a macro to
manipulate the connected line data. This way, phone numbers may
be manipulated to be more presentable to users, names may be
changed for...whatever reason, or whatever else needs to be done
may be. Review: https://reviewboard.asterisk.org/r/256 AST-165
2009-06-01 20:33 +0000 [r198725] Tilghman Lesher <tlesher@digium.com>
* funcs/func_math.c: Add INCrement and DECrement functions (closes
issue #15025) Reported by: greenfieldtech Patches:
func_math.c.patch_v4 uploaded by greenfieldtech (license 369)
slightly modified by me Tested by: greenfieldtech, lmadsen
2009-06-01 20:17 +0000 [r198670] Russell Bryant <russell@digium.com>
* include/asterisk/frame.h: Minor whitespace fix.
2009-06-01 19:37 +0000 [r198661] Eliel C. Sardanons <eliels@gmail.com>
* res/res_monitor.c: Moved more static documentation to the new
AstXML form. Moved more static docs to XML (pplications and
manager actions): Monitor, StopMonitor, ChangeMonitor,
PauseMonitor, UnpauseMonitor.
2009-06-01 18:40 +0000 [r198626] Tilghman Lesher <tlesher@digium.com>
* contrib/scripts/meetme.sql: Add information for new meetme
realtime fields
2009-06-01 17:53 +0000 [r198561-198597] Eliel C. Sardanons <eliels@gmail.com>
* main/Makefile: Do not add say.o in a separate line.
* res/res_jabber.c: Move JabberSend manager action from static docs
to the AstXML form.
* res/res_agi.c: Move static documentation of E|Dead|AGI()
application and manager action to XML.
2009-06-01 15:23 +0000 [r198558] David Vossel <dvossel@digium.com>
* main/threadstorage.c: Fixed an issue in the threadstorage cli
functions resulting from the constification of struct
ast_cli_args in r196072.
2009-06-01 14:45 +0000 [r198500-198530] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c: Remove extra lock from app_queue.
* channels/chan_local.c: Remove extra lock from local_indicate in
connected line case. Oh, and this fixes a deadlock I was seeing.
* channels/chan_local.c: Add missing unlock of local pvt.
* channels/chan_agent.c: Remove documentation for the 'exten'
argument to the AGENT function. Since AgentCallbackLogin has been
removed, this should not be documented any more.
2009-06-01 13:31 +0000 [r198498] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: Fix a bug where the Event and Content-Type
headers were added twice to outgoing SIP NOTIFY messages. (closes
issue #15239) Reported by: pj
2009-05-31 17:52 +0000 [r198470] Tilghman Lesher <tlesher@digium.com>
* funcs/func_strings.c: Fix documentation for FIELDQTY.
2009-05-31 02:09 +0000 [r198442] Eliel C. Sardanons <eliels@gmail.com>
* main/Makefile: Filter the say.o object, it is being added later.
2009-05-31 01:40 +0000 [r198438] Russell Bryant <russell@digium.com>
* main/manager.c: Constification and remove some unused code.
2009-05-31 01:22 +0000 [r198437] Eliel C. Sardanons <eliels@gmail.com>
* res/res_timing_dahdi.c: Avoid a crash when res_timing_dahdi is
unloaded but wasn't properly loaded. if dahdi_test_timer() fails,
timing_funcs_handle remains NULL causing a crash when calling
ast_unregister_timing_interface() with a NULL pointer. (closes
issue #15234) Reported by: eliel Patches: timing_dahdi1.diff
uploaded by eliel (license 64)
2009-05-31 01:19 +0000 [r198434] Russell Bryant <russell@digium.com>
* main/channel.c, include/asterisk/channel.h: Constify the
ast_frame arg to ast_queue_frame().
2009-05-30 20:11 +0000 [r198371-198375] Sean Bright <sean@malleable.com>
* res/res_jabber.c: Properly terminate the receive buffer before
sending to iksemel. aji_io_recv takes the maximum number of bytes
to read (instead of the total buffer size), so we have to
subtract 1 from our buffer size. Without this, when we receive
packets that are larger than our buffer, iksemel will choke and
things get wonky. (closes issue #15232) Reported by: lp0 Patches:
05302009_res_jabber.c.patch uploaded by seanbright (license 71)
Tested by: seanbright, lp0
* /, res/res_jabber.c: Merged revisions 198370 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r198370 | seanbright | 2009-05-30 15:36:20 -0400 (Sat, 30 May
2009) | 12 lines Properly terminate AMI JabberSend response
messages. The response message (either Error or Success) needs an
extra trailing \r\n after the fields to inform the client that
the message is complete. (closes issue #14876) Reported by: srt
Patches: 05302009_1.4_res_jabber.c.diff uploaded by seanbright
(license 71) asterisk_14876.patch uploaded by srt (license 378)
trunk-14876-2.diff uploaded by phsultan (license 73) ........
2009-05-30 03:43 +0000 [r198312] Russell Bryant <russell@digium.com>
* res/res_smdi.c, /: Merged revisions 198311 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r198311 | russell | 2009-05-29 22:42:46 -0500 (Fri, 29 May 2009)
| 5 lines Fix a crash that occurred when MWI SMDI messages
expired. (closes issue #14561) Reported by: cmoss28 ........
2009-05-30 03:26 +0000 [r198285] Sean Bright <sean@malleable.com>
* apps/app_dial.c, /: Merged revisions 198251 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r198251 | seanbright | 2009-05-29 22:46:41 -0400 (Fri, 29 May
2009) | 8 lines Treat an empty FORWARD_CONTEXT the same way we
treat a missing one. (closes issue #15056) Reported by:
p_lindheimer Patches: 05292009_bug15056.diff uploaded by
seanbright (license 71) Tested by: p_lindheimer ........
2009-05-30 02:31 +0000 [r198248] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: When removing all packets from a dialog we
also need to free the data if present.
2009-05-30 01:04 +0000 [r198217] Eliel C. Sardanons <eliels@gmail.com>
* configs/agents.conf.sample, channels/chan_agent.c: Remove not
used code in the Agent channel. This code was there because of
the AgentCallbackLogin() application. ->loginchan[] member was
only used by AgentCallbackLogin(). Agent where dumped to astdb if
they where logged in using AgentCallbacklogin() so they are not
being dumper anymore. Review:
https://reviewboard.asterisk.org/r/267/
2009-05-29 23:04 +0000 [r198183-198186] Russell Bryant <russell@digium.com>
* configs/modules.conf.sample: Suggesting that only a single timing
module be loaded is no longer necessary.
* res/res_timing_pthread.c: Improve handling of trying to ACK too
many timer expirations.
2009-05-29 22:21 +0000 [r198182] Terry Wilson <twilson@digium.com>
* res/res_calendar.c: Add a couple of TODO items so I don't forget
2009-05-29 20:06 +0000 [r198146] Russell Bryant <russell@digium.com>
* res/res_timing_pthread.c: Resolve issues with choppy sound when
using res_timing_pthread. The situation that caused this problem
was when continuous mode was being turned on and off while a rate
was set for a timing interface. A very easy way to replicate this
bug was to do a Playback() from behind a Local channel. In this
scenario, a rate gets set on the channel for doing file playback.
At the same time, continuous mode gets turned on and off about
every 20 ms as frames get queued on to the PBX side channel from
the other side of the Local channel. Essentially, this module
treated continuous mode and a set rate as mutually exclusive
states for the timer to be in. When I dug deep enough, I observed
the following pattern: 1) Set timer to tick every 20 ms. 2) Wait
almost 20 ms ... 3) Continuous mode gets turned on for a queued
up frame 4) Continuous mode gets turned off 5) The timer goes
back to its tick per 20 ms. state but starts counting at 0 ms. 6)
Goto step 2. Sometimes, res_timing_pthread would make it 20 ms
and produce a timer tick, but not most of the time. This is what
produced the choppy sound (or sometimes no sound at all). Now,
the module treats continuous mode and a set rate as completely
independent timer modes. They can be enabled and disabled
independently of each other and things work as expected. (closes
issue #14412) Reported by: dome Patches: issue14412.diff.txt
uploaded by russell (license 2) issue14412-1.6.1.0.diff.txt
uploaded by russell (license 2) Tested by: DennisD, russell
2009-05-29 19:46 +0000 [r198139] Eliel C. Sardanons <eliels@gmail.com>
* main/Makefile: Simplify the Makefile and avoid needing to specify
each object file. Instead of specifying every object file, use
make's magic to generate it. This will generate less conflicts in
team branches when a new file is added in trunk. (closes issue
#15226) Reported by: eliel Patches: makefile uploaded by eliel
(license 64) Review: http://reviewboard.asterisk.org/r/269/
2009-05-29 19:19 +0000 [r198088] Jeff Peeler <jpeeler@digium.com>
* channels/chan_dahdi.c, channels/sig_analog.c (added),
channels/sig_analog.h (added), channels/Makefile: New signaling
module to handle analog operations in chan_dahdi This branch
splits all the analog signaling logic out of chan_dahdi.c into
sig_analog.c. Functionality in theory should not change at all.
As noted in the code, there is still some unused code remaining
that will be cleaned up in a later commit. Review:
https://reviewboard.asterisk.org/r/253/
2009-05-29 19:18 +0000 [r198083] Eliel C. Sardanons <eliels@gmail.com>
* CREDITS: Apply anti-spam obfuscation to an email address.
2009-05-29 19:04 +0000 [r198072] Matthew Nicholson <mnicholson@digium.com>
* main/cdr.c, main/channel.c, /, include/asterisk/cdr.h: Merged
revisions 198068 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r198068 | mnicholson | 2009-05-29 13:53:01 -0500 (Fri, 29 May
2009) | 15 lines Use AST_CDR_NOANSWER instead of AST_CDR_NULL as
the default CDR disposition. This change also involves the
addition of an AST_CDR_FLAG_ORIGINATED flag that is used on
originated channels to distinguish: them from dialed channels.
(closes issue #12946) Reported by: meral Patches: null-cdr2.diff
uploaded by mnicholson (license 96) Tested by: mnicholson,
dbrooks (closes issue #15122) Reported by: sum Tested by: sum
........
2009-05-29 18:39 +0000 [r198064] Joshua Colp <jcolp@digium.com>
* main/file.c: Fix a memory leak of the write buffer when writing a
file.
2009-05-29 18:15 +0000 [r198000] Sean Bright <sean@malleable.com>
* Makefile, /: Merged revisions 197998 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r197998 | seanbright | 2009-05-29 14:14:12 -0400 (Fri, 29 May
2009) | 8 lines Fix 'make config' target for Slackware. There was
a missing semi-colon after the echo statement in the Makefile
that was causing problems for some users. Fix suggested by
reporter. (closes issue #15225) Reported by: pdavis ........
2009-05-29 17:51 +0000 [r197996] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: Fix a bug where the default setting did not
perform a remote bridge when it should have.
2009-05-29 16:15 +0000 [r197960] Russell Bryant <russell@digium.com>
* res/res_timing_pthread.c: Trim trailing whitespace so that I can
work on this bug without it bothering me. :-)
2009-05-29 15:48 +0000 [r197959] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: A few fixes to SIP with regards to connected
line updates during transfers. * Set the invitestate to
INV_CALLING when we send a connected line reinvite. This prevents
us from potentially rapid-firing reinvites to a single peer. *
Use the astdb to store a peer's allowed methods. This prevents us
from sending an UPDATE during the interval between startup and
the peer's first registration if the peer does not support the
UPDATE method. * Handle Polycom's method of indicating allowed
methods in REGISTER. Instead of using an Allow header, they place
the allowed methods in a methods= parameter in the Contact
header. ABE-1873
2009-05-29 05:15 +0000 [r197926] Terry Wilson <twilson@digium.com>
* doc/tex/asterisk.tex, doc/tex/calendaring.tex (added): Add some
TeX docs for calendaring. I still need to set up tests to make
sure my examples are completely correct, but I ran out of time
tonight and felt that they at least would give an idea as to how
to use calendaring. I will try to test the examples and do some
cleanup on the docs tomorrow night.
2009-05-28 22:42 +0000 [r197861] Sean Bright <sean@malleable.com>
* include/asterisk/doxygen/releases.h, sounds/Makefile: Update
references to downloads.digium.com to its new URL.
2009-05-28 22:04 +0000 [r197828] Leif Madsen <lmadsen@digium.com>
* apps/app_mixmonitor.c: Update documentation in MixMonitor.
Updated the MixMonitor documentation for the 'b' option so that
it is more obvious that you must not optimize away the Local
channel when using this option. (closes issue #14829) Reported
by: licedey Tested by: mmichelson, licedey, lmadsen
2009-05-28 21:50 +0000 [r197824] Sean Bright <sean@malleable.com>
* doc/CODING-GUIDELINES, doc/asterisk.8, BUGS, doc/backtrace.txt,
doc/tex/mp3.tex, channels/h323/README, main/enum.c,
doc/tex/misdn.tex, include/asterisk/doxyref.h,
contrib/scripts/ast_grab_core, doc/tex/backtrace.tex,
include/asterisk/doxygen/reviewboard.h,
include/asterisk/doxygen/commits.h,
contrib/scripts/asterisk.ldif,
contrib/scripts/asterisk.ldap-schema,
configs/extensions.conf.sample, doc/asterisk.sgml: Update
references to bugs.digium.com and reviewboard.digium.com to the
new URLs.
2009-05-28 20:43 +0000 [r197777] Terry Wilson <twilson@digium.com>
* configs/calendar.conf.sample: Make note of Exchange calendar
support limitations
2009-05-28 20:36 +0000 [r197775] Kevin P. Fleming <kpfleming@digium.com>
* main/utils.c: Ensure that accidental calls to
ast_string_field_free_memory() on embedded stringfield pools are
safe. It is possible for a stringfield manager structure (and
pool) structure to be allocated as part of a larger structure
allocation (using ast_calloc_with_strinfields()); when this is
done, the stringfield pool cannot be separately freed, but users
of the tructure may not be aware (and shouldn't have to be aware)
of whether the pool was embedded. This patch modifies the
behavior so that they can always call
ast_string_field_free_memory() and the function will do the right
thing for both embedded and non-embedded situations.
2009-05-28 20:17 +0000 [r197740] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Treat 405 responses the same way we would a
501. This makes sure that we mark a method as being unallowed if
we receive a 405 response so that we don't continue to try to
send that same type of message.
2009-05-28 19:57 +0000 [r197738] Terry Wilson <twilson@digium.com>
* res/res_calendar.exports (added), res/res_calendar_exchange.c
(added), res/res_calendar_icalendar.c (added),
build_tools/menuselect-deps.in, configure,
include/asterisk/autoconfig.h.in, configure.ac,
configs/calendar.conf.sample (added), res/res_calendar_caldav.c
(added), include/asterisk/calendar.h (added), makeopts.in,
res/res_calendar.c (added): Add Calendaring support for Asterisk
This commit add Calendaring support to Asterisk for iCalendar,
CalDAV, and MS Exchange calendars. Exchange support has only been
tested on Exchange Server 2k3 and does not support forms-based
authentication at this time (patches *very* welcome). Exchange
support is also currently missing the ability to return a list of
a meting's attendees (again, patches are very, very welcome).
Features include: Querying a calendar for events over a specific
time range Checking a calendar's busy status via the dialplan
Writing calendar events via the dialplan (CalDAV and Exchange
only) Handling calendar event notifications through the dialplan
(closes issue #14771) Tested by: lmadsen, twilson, Shivaprakash
Review: https://reviewboard.asterisk.org/r/58
2009-05-28 18:48 +0000 [r197701] Mark Michelson <mmichelson@digium.com>
* channels/chan_local.c: Add missing lock to local_indicate
function for connected line frames.
2009-05-28 18:45 +0000 [r197697] Joshua Colp <jcolp@digium.com>
* channels/chan_iax2.c: Fix a bug where the trunkmtu setting was
not set to the default value of 1240 on load but was on reload.
2009-05-28 16:01 +0000 [r197621] Eliel C. Sardanons <eliels@gmail.com>
* /, channels/chan_sip.c: Merged revisions 197562 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r197562 | eliel | 2009-05-28 11:21:32 -0400 (Thu, 28 May 2009) |
13 lines Use the address we already know when reloading a peer
with nat=yes. If we already have an address for a peer, and we
are reloading the sip configuration, try to use that address to
contact the peer, instead of getting it from the Contact. (closes
issue #15194) Reported by: ibc Patches: sip.patch uploaded by
eliel (license 64) Tested by: manwe ........
2009-05-28 15:35 +0000 [r197616] Tilghman Lesher <tlesher@digium.com>
* channels/chan_dahdi.c, channels/chan_console.c, apps/app_rpt.c,
main/astobj2.c, main/cli.c: Eliminate several needless checks and
fix a few memory leaks (closes issue #14833) Reported by:
contactmayankjain Patches: all_changes.patch uploaded by
contactmayankjain (license 740) slightly modified by me
2009-05-28 15:32 +0000 [r197606] Mark Michelson <mmichelson@digium.com>
* /: Recorded merge of revisions 197588 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r197588 | mmichelson | 2009-05-28 10:27:49 -0500 (Thu, 28 May
2009) | 16 lines Allow for media to arrive from an alternate
source when responding to a reinvite with 491. When we receive a
SIP reinvite, it is possible that we may not be able to process
the reinvite immediately since we have also sent a reinvite out
ourselves. The problem is that whoever sent us the reinvite may
have also sent a reinvite out to another party, and that reinvite
may have succeeded. As a result, even though we are not going to
accept the reinvite we just received, it is important for us to
not have problems if we suddenly start receiving RTP from a new
source. The fix for this is to grab the media source information
from the SDP of the reinvite that we receive. This information is
passed to the RTP layer so that it will know about the alternate
source for media. Review: https://reviewboard.asterisk.org/r/252
........
2009-05-28 15:23 +0000 [r197570] Joshua Colp <jcolp@digium.com>
* main/logger.c: Fix an incorrect call to
ast_string_field_free_memory which caused a crash in the logger.
Since the message structure is allocated using
ast_calloc_with_stringfields we do not need to free the memory
used for the stringfields as it will get freed when the message
structure is.
2009-05-28 14:58 +0000 [r197543] Mark Michelson <mmichelson@digium.com>
* /, include/asterisk/audiohook.h, main/audiohook.c,
apps/app_chanspy.c: Merged revisions 197537 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r197537 | mmichelson | 2009-05-28 09:49:13 -0500 (Thu, 28 May
2009) | 21 lines Add flags to chanspy audiohook so that audio
stays in sync. There are two flags being added to the chanspy
audiohook here. One is the pre-existing
AST_AUDIOHOOK_TRIGGER_SYNC flag. With this set, we ensure that
the read and write slinfactories on the audiohook do not skew
beyond a certain tolerance. In addition, there is a new audiohook
flag added here, AST_AUDIOHOOK_SMALL_QUEUE. With this flag set,
we do not allow for a slinfactory to build up a substantial
amount of audio before flushing it. For this particular issue,
this means that the person spying on the call will hear the
conversations in real time with very little delay in the audio.
(closes issue #13745) Reported by: geoffs Patches: 13745.patch
uploaded by mmichelson (license 60) Tested by: snblitz ........
2009-05-28 14:51 +0000 [r197538] Joshua Colp <jcolp@digium.com>
* main/utils.c: Fix a bug in stringfields where it did not actually
free the pools of memory. (closes issue #15074) Reported by: pj
2009-05-28 14:39 +0000 [r197528-197535] Sean Bright <sean@malleable.com>
* configs/amd.conf.sample, configs/users.conf.sample,
configs/gtalk.conf.sample, configs/rpt.conf.sample,
configs/rtp.conf.sample, configs/cli_aliases.conf.sample,
configs/modules.conf.sample, configs/phone.conf.sample,
configs/extensions.ael.sample, configs/skinny.conf.sample,
configs/ais.conf.sample, configs/meetme.conf.sample,
configs/extensions_minivm.conf.sample, configs/telcordia-1.adsi,
configs/alsa.conf.sample, configs/iax.conf.sample,
configs/followme.conf.sample, configs/mgcp.conf.sample,
configs/sip.conf.sample, configs/extensions.lua.sample,
configs/say.conf.sample, configs/queuerules.conf.sample,
configs/minivm.conf.sample, configs/osp.conf.sample,
configs/chan_dahdi.conf.sample,
configs/cli_permissions.conf.sample, configs/console.conf.sample,
configs/dundi.conf.sample, configs/indications.conf.sample,
configs/oss.conf.sample, configs/queues.conf.sample,
configs/voicemail.conf.sample, configs/usbradio.conf.sample,
configs/cdr.conf.sample, configs/jingle.conf.sample,
configs/misdn.conf.sample, configs/manager.conf.sample,
configs/festival.conf.sample, configs/features.conf.sample,
configs/logger.conf.sample, configs/http.conf.sample,
configs/h323.conf.sample, configs/sla.conf.sample,
configs/phoneprov.conf.sample, configs/res_odbc.conf.sample,
configs/agents.conf.sample, configs/alarmreceiver.conf.sample,
configs/func_odbc.conf.sample, configs/musiconhold.conf.sample,
configs/jabber.conf.sample, configs/extconfig.conf.sample,
configs/res_snmp.conf.sample, configs/iaxprov.conf.sample,
configs/unistim.conf.sample, configs/dnsmgr.conf.sample,
configs/extensions.conf.sample, configs/asterisk.adsi: Remove a
bunch of trailing whitespace in preparation for
reformatting/cleanup. Let's try that again, this time removing
trailing whitespace and not leading whitespace. I can't believe
no one noticed.
* configs/amd.conf.sample, configs/gtalk.conf.sample,
configs/rtp.conf.sample, configs/rpt.conf.sample,
configs/cli_aliases.conf.sample, configs/extensions.ael.sample,
configs/skinny.conf.sample, configs/meetme.conf.sample,
configs/telcordia-1.adsi, configs/alsa.conf.sample,
configs/iax.conf.sample, configs/mgcp.conf.sample,
configs/extensions.lua.sample, configs/sip.conf.sample,
configs/say.conf.sample, configs/minivm.conf.sample,
configs/console.conf.sample, configs/cli_permissions.conf.sample,
configs/chan_dahdi.conf.sample, configs/oss.conf.sample,
configs/queues.conf.sample, configs/jingle.conf.sample,
configs/usbradio.conf.sample, configs/voicemail.conf.sample,
configs/misdn.conf.sample, configs/manager.conf.sample,
configs/features.conf.sample, configs/h323.conf.sample,
configs/sla.conf.sample, configs/res_odbc.conf.sample,
configs/phoneprov.conf.sample, configs/alarmreceiver.conf.sample,
configs/func_odbc.conf.sample, configs/musiconhold.conf.sample,
configs/jabber.conf.sample, configs/unistim.conf.sample,
configs/dnsmgr.conf.sample, configs/extensions.conf.sample,
configs/asterisk.adsi: Remove a bunch of trailing whitespace in
preparation for reformatting/cleanup.
2009-05-28 13:47 +0000 [r197467] Joshua Colp <jcolp@digium.com>
* /, channels/chan_sip.c: Merged revisions 197466 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r197466 | file | 2009-05-28 10:44:58 -0300 (Thu, 28 May 2009) | 8
lines Fix a bug where the flag indicating the presence of rport
would get overwritten by the nat setting. The presence of rport
is now stored as a separate flag. Once the dialog is setup and
authenticated (or it passes through unauthenticated) the proper
nat flag is set. (closes issue #13823) Reported by: dimas
........
2009-05-28 11:25 +0000 [r197406-197431] Gavin Henry <ghenry@suretecsystems.com>
* contrib/scripts/asterisk.ldap-schema,
contrib/scripts/asterisk.ldif: Added AstVoicemailContext Added
AstVoicemailContext (closes issue #15155) Reported by: scramatte
Tested by: suretec
* contrib/scripts/asterisk.ldap-schema,
contrib/scripts/asterisk.ldif: New objectclass AsteriskVoiceMail
and AstAccountCallLimit attribute Added new ObjectClass
AsteriskVoiceMail, and AstAccountCallLimit attribute and cleaned
up formatting and tested with OpenLDAP (closes issue #15155)
Reported by: scramatte Patches: asterisk.schema uploaded by
scramatte (license 796) Tested by: suretec Review: [full review
board URL with trailing slash]
* doc/ldap.txt, configs/res_ldap.conf.sample,
contrib/scripts/asterisk.ldap-schema,
contrib/scripts/asterisk.ldif: closes issue #15156
2009-05-27 23:48 +0000 [r197374] Tilghman Lesher <tlesher@digium.com>
* main/xml.c: Revert commit 192032. This define is needed on Mac OS
X.
2009-05-27 22:42 +0000 [r197338] Russell Bryant <russell@digium.com>
* main/rtp_engine.c: Don't do a pointer comparison before setting
the remote address.
2009-05-27 22:21 +0000 [r197335] Kevin P. Fleming <kpfleming@digium.com>
* include/asterisk/agi.h: Ensure that this header includes
xmldoc.h, since it depends on it.
2009-05-27 20:14 +0000 [r197266] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: Adding some generic handling of error codes
sent to us in replys to requests. Previously they always set
hangupcause 0, which is generally wrong. With this change, we're
setting some generic hangup causes. For 5xx errors, which
indicate some sort of problem with the remote server, we're now
setting CONGESTION. EDVX002
2009-05-27 20:08 +0000 [r197260] Sean Bright <sean@malleable.com>
* Makefile: Use bash explicitly when calling
build_tools/mkpkgconfig from the Makefile. Since we use bashisms
in build_tools/mkpkgconfig, we should call on bash explicitly
when running from the Makefile, otherwise we get errors during a
'make install.' (closes issue #15209) Reported by: seandarcy
2009-05-27 19:20 +0000 [r197209] Tilghman Lesher <tlesher@digium.com>
* /, funcs/func_cut.c: Recorded merge of revisions 197194 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r197194 | tilghman | 2009-05-27 14:09:42 -0500 (Wed, 27 May 2009)
| 5 lines Use a different determinator on whether to print the
delimiter, since leading fields may be blank. (closes issue
#15208) Reported by: ramonpeek Patch by me, though inspired in
part by a patch from ramonpeek ........
2009-05-27 18:25 +0000 [r196948-197189] Sean Bright <sean@malleable.com>
* configs/adtranvofr.conf.sample (removed): Remove a file sample
configuration file that is no longer used.
* configs/chan_dahdi.conf.sample, configs/vpb.conf.sample,
configs/smdi.conf.sample, configs/extensions.conf.sample,
configs/sla.conf.sample: Fix references to /etc/dahdi/system.conf
and /etc/asterisk/chan_dahdi.conf in the sample configuration
files. (closes issue #15207) Reported by: seandarcy
* channels/chan_alsa.c: Display an error message when chan_alsa
fails to load due to a missing or inaccessible configuration
file. Before this change, when chan_alsa failed to load due to a
missing or inaccessible configuration file, no message would be
displayed. With this change, when chan_alsa fails to load due to
a missing or inaccessible configuration file, a message will be
displayed. (closes issue #14760) Reported by: Nick_Lewis Patches:
chan_alsa.c-confload.patch uploaded by Nick (license 657)
* main/xmldoc.c: Reset the terminal to the correct fg/bg after XML
documenation is rendered. (closes issue #15200) Reported by:
ajohnson Patches: 05262009_xmldoc.patch uploaded by seanbright
(license 71) Tested by: ajohnson
2009-05-26 22:40 +0000 [r196946] Russell Bryant <russell@digium.com>
* autoconf/ast_check_osptk.m4 (added), configure,
include/asterisk/autoconfig.h.in, configure.ac: Update configure
script to check for OSP toolkit 3.5.0. (closes issue #14988)
Reported by: tzafrir Patches: configure.ac.diff uploaded by
homesick (license 91) new_ast_check_osptk.m4 uploaded by homesick
(license 91)
2009-05-26 22:38 +0000 [r196907-196945] Sean Bright <sean@malleable.com>
* main/manager.c: Add ActionID to CoreShowChannel event. There is
inconsistency in how we handle manager responses that are lists
of items and, unfortunately, third parties have come to rely on
ActionID being on every event within those lists instead of just
keeping track of the ActionID for the current response. This
change makes CoreShowChannels include the ActionID with each
CoreShowChannel event generated as a result of it being called.
(closes issue #15001) Reported by: sum Patches:
patchactionid2.patch uploaded by sum (license 766)
* main/manager.c: Include startup and reload date in the CoreStatus
manager message. The CoreStartupTime and CoreReloadTime
name/value pairs in the CoreStatus response message only included
the time and not the date. This patch, inspired by the reporter's
patch, adds 2 new fields - CoreStartupDate and CoreReloadDate -
which contain the date portion of these values. (closes issue
#15000) Reported by: sum
2009-05-26 19:50 +0000 [r196893] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c, apps/app_directed_pickup.c: Remove some
redundant or unnecessary connected line-related function calls.
2009-05-26 18:20 +0000 [r196843] Russell Bryant <russell@digium.com>
* /, res/res_convert.c: Merged revisions 196826 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r196826 | russell | 2009-05-26 13:14:36 -0500 (Tue, 26 May 2009)
| 9 lines Resolve a file handle leak. The frames here should have
always been freed. However, out of luck, there was never any
memory leaked. However, after file streams became reference
counted, this code would leak the file stream for the file being
read. (closes issue #15181) Reported by: jkroon ........
2009-05-26 16:38 +0000 [r196725-196792] Sean Bright <sean@malleable.com>
* apps/app_queue.c: Add a missing unref for queues in
handle_statechange.
* main/pbx.c, include/asterisk/pbx.h, res/res_clioriginate.c: Add
new ast_complete_applications function so that we can use it with
the 'channel originate ... application <app>' CLI command. (And
yeah, I cleaned up some whitespace in res_clioriginate.c... big
whoop, wanna fight about it!?)
* cdr/cdr_sqlite3_custom.c: Use a properly allocated channel for
substitution in cdr_sqlite3_custom.
2009-05-26 13:43 +0000 [r196658-196721] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: Fix a bug where the sip unregister CLI
command did not completely unregister the peer. (closes issue
#15118) Reported by: alecdavis Patches:
chan_sip_unregister.diff2.txt uploaded by alecdavis (license 585)
* /, contrib/scripts/safe_asterisk: Merged revisions 196657 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r196657 | file | 2009-05-26 10:06:09 -0300 (Tue, 26 May 2009) | 7
lines Remove some bash specific stuff from safe_asterisk. (closes
issue #10812) Reported by: paravoid Patches:
safe_asterisk_bashism.diff uploaded by tzafrir (license 46)
........
2009-05-26 12:14 +0000 [r196622] Sean Bright <sean@malleable.com>
* cdr/cdr_manager.c: Use a properly allocated channel for
substitution in cdr_manager.
2009-05-24 16:17 +0000 [r196554-196585] Eliel C. Sardanons <eliels@gmail.com>
* res/res_agi.c: Move AGI static documentation to the new AstXML
form. Move AGI commands documentation to XML docs: 'set priority'
'set variable' 'stream file' 'control stream file' 'tdd mode'
'verbose' 'wait for digit' 'speech create' 'speech set' 'speech
destroy' 'speech load grammar' 'speech unload grammar' 'speech
activate grammar' 'speech deactivate grammar' 'speech recognize'
* res/res_agi.c: Move static AGI commands documentation to XML.
Move AGI commands ('say datetime', 'send image', 'send text',
'set autohangup', 'set callerid', 'set context', 'set extension')
documentation to the AstXML form.
2009-05-23 15:16 +0000 [r196520] Sean Bright <sean@malleable.com>
* cdr/cdr_custom.c: Fix errors in cdr_custom that cause reference
errors when non-CDR variable substitution is done. cdr_custom was
creating a ast_channel struct directly and passing it into the
core for variable substition. This was fine as long as the format
string contained only calls to the CDR() function. Doing
something like ${EPOCH} on the other hand tried to lock the
channel, which would fail and throw an error because the passed
channel hadn't been allocated as an ao2 object. So now we create
the dummy channel with ast_channel_alloc, and everything works as
expected.
2009-05-23 13:31 +0000 [r196488] Kevin P. Fleming <kpfleming@digium.com>
* include/asterisk/cli.h: Correct example for CLI autocompletion
(generation) Reported by Atis on #asterisk-dev
2009-05-23 04:27 +0000 [r196456] Moises Silva <moises.silva@gmail.com>
* channels/chan_dahdi.c: set MFCR2_CATEGORY just when starting the
pbx
2009-05-22 21:11 +0000 [r196417] Sean Bright <sean@malleable.com>
* main/asterisk.c: Call ast_stun_init() when we're initializing to
get the 'stun debug set' commands.
2009-05-22 21:09 +0000 [r196416] David Vossel <dvossel@digium.com>
* channels/chan_sip.c, configs/sip.conf.sample: SIP set outbound
transport type from Registration In sip.conf the transport option
allows for the configuration of what transport types (udp, tcp,
and tls) a peer will accept, but only the first type listed was
used for outbound connections. This patch changes this. Now the
default transport type is only used until the peer registers.
When registration takes place the transport type is parsed out of
the Contact header. If the Contact header's transport type is
equal to one that the peer supports, the peer's default transport
type for outbound connections is set to match the Contact
header's type. If the Contact header's transport type is not
present, then the peer's default transport type is set to match
the one the peer registered with. When a peer unregisters or the
registration expires, the default transport type for that peer is
reset. (closes issue #12282) Reported by: rjain Patches:
reg_patch_1.diff uploaded by dvossel (license 671) Tested by:
dvossel (closes issue #14727) Reported by: pj Patches:
reg_patch_3.diff uploaded by dvossel (license 671) Tested by: pj,
dvossel Review: https://reviewboard.asterisk.org/r/249/
2009-05-22 20:01 +0000 [r196381] Sean Bright <sean@malleable.com>
* channels/chan_gtalk.c: Don't crash if an RTP instance can't be
created. This could occur when an invalid bindaddr was specified
in gtalk.conf.
2009-05-22 19:38 +0000 [r196308-196377] Eliel C. Sardanons <eliels@gmail.com>
* apps/app_minivm.c: Unregister every registered application by
MiniVM. The MinivmMWI application was not being unregistered on
unload and we were not able to load again the module or reload
it. (closes issue #15174) Reported by: junky Patches:
unregister_minivm_mwi.diff uploaded by junky (license 177)
* res/res_agi.c: Moved static documentation to the AstXML form.
Moved AGI commands static documentation to XML docs ('say alpha',
'say digits', 'say number', 'say phonetic', 'say date' and 'say
time').
* main/pbx.c, channels/chan_sip.c, apps/app_meetme.c,
channels/chan_agent.c, apps/app_queue.c, channels/chan_iax2.c,
include/asterisk/manager.h, channels/chan_dahdi.c,
main/manager.c, channels/chan_skinny.c, main/features.c,
res/res_agi.c, include/asterisk/xmldoc.h, include/asterisk/pbx.h,
apps/app_senddtmf.c, doc/appdocsxml.dtd, main/db.c,
main/xmldoc.c, apps/app_voicemail.c: Implement a new element in
AstXML for AMI actions documentation. A new xml element was
created to manage the AMI actions documentation, using AstXML. To
register a manager action using XML documentation it is now
possible using ast_manager_register_xml(). The CLI command
'manager show command' can be used to show the parsed
documentation. Example manager xml documentation: <manager
name="ami action name" language="en_US"> <synopsis> AMI action
synopsis. </synopsis> <syntax> <xi:include
xpointer="xpointer(...)" /> <-- for ActionID <parameter
name="header1" required="true"> <para>Description</para>
</parameter> ... </syntax> <description> <para>AMI action
description</para> </description> <see-also> ... </see-also>
</manager>
2009-05-22 16:53 +0000 [r196272] Tilghman Lesher <tlesher@digium.com>
* main/astmm.c: Two more minor fixes due to constification
2009-05-22 16:51 +0000 [r196270] Sean Bright <sean@malleable.com>
* res/res_agi.c: Fix res_agi compilation after the const-ify the
world merge. Since we are dealing with a 'const char * const'
now, we have to create a temporary copy of the string to work on
rather than the original. Fix inspired by reporter. Reviewed by
everyone-and-their-mother in #asterisk-dev. (closes issue #15184)
Reported by: andrew
2009-05-22 16:50 +0000 [r196268] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: s/it's/its/
2009-05-22 16:20 +0000 [r196246] Russell Bryant <russell@digium.com>
* channels/chan_dahdi.c: resolve compiler warning
2009-05-22 16:10 +0000 [r196227] Sean Bright <sean@malleable.com>
* channels/chan_dahdi.c, main/pbx.c, res/res_jabber.c,
res/res_monitor.c: Fix build under dev mode and remove some casts
that are no longer necessary as a result of the const-ify the
world patch.
2009-05-22 15:07 +0000 [r196187-196188] Richard Mudgett <rmudgett@digium.com>
* apps/app_mp3.c: Fix constify the world compile problem.
* channels/chan_misdn.c: Make chan_misdn compile.
2009-05-22 13:56 +0000 [r196117] Joshua Colp <jcolp@digium.com>
* channels/chan_misdn.c, /: Merged revisions 196116 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r196116 | file | 2009-05-22 10:54:17 -0300 (Fri, 22 May
2009) | 5 lines Fix a bug where using immediate with mISDN caused
a cause code of 16 to get sent back instead of 1 if the 's'
extension did not exist. (closes issue #12286) Reported by:
lmamane ........
2009-05-22 13:34 +0000 [r196114] Eliel C. Sardanons <eliels@gmail.com>
* main/pbx.c: Avoid using prototypes when not necessary (it is
already defined in the header file). Make log_match_char_tree()
static to main/pbx.c (only used there).
2009-05-21 21:13 +0000 [r196072] Kevin P. Fleming <kpfleming@digium.com>
* apps/app_dahdibarge.c, main/frame.c, apps/app_record.c,
apps/app_playtones.c, funcs/func_strings.c,
include/asterisk/extconf.h, apps/app_alarmreceiver.c,
apps/app_chanisavail.c, apps/app_ices.c, apps/app_exec.c,
channels/chan_iax2.c, main/astobj2.c, channels/chan_dahdi.c,
channels/chan_skinny.c, apps/app_dumpchan.c, pbx/pbx_ael.c,
main/pbx.c, channels/vcodecs.c, apps/app_softhangup.c,
apps/app_morsecode.c, apps/app_talkdetect.c,
channels/iax2-parser.c, apps/app_db.c, apps/app_speech_utils.c,
apps/app_sendtext.c, pbx/pbx_config.c, apps/app_mixmonitor.c,
main/asterisk.c, res/res_odbc.c, apps/app_voicemail.c,
apps/app_dictate.c, apps/app_authenticate.c,
apps/app_readexten.c, apps/app_userevent.c, res/res_jabber.c,
include/asterisk/abstract_jb.h, main/channel.c,
apps/app_setcallerid.c, apps/app_osplookup.c, funcs/func_odbc.c,
apps/app_mp3.c, apps/app_minivm.c, apps/app_directory.c,
apps/app_rpt.c, channels/chan_mgcp.c, apps/app_adsiprog.c,
apps/app_read.c, channels/chan_sip.c,
include/asterisk/taskprocessor.h, include/asterisk/cli.h,
apps/app_originate.c, utils/conf2ael.c,
apps/app_channelredirect.c, apps/app_forkcdr.c,
main/abstract_jb.c, channels/misdn/chan_misdn_config.h,
apps/app_sms.c, utils/extconf.c, funcs/func_devstate.c,
apps/app_stack.c, apps/app_verbose.c, main/dsp.c, main/udptl.c,
include/asterisk/agi.h, cdr/cdr_sqlite3_custom.c,
apps/app_readfile.c, apps/app_sayunixtime.c, apps/app_test.c,
include/asterisk/speech.h, cdr/cdr_adaptive_odbc.c,
apps/app_image.c, main/taskprocessor.c, main/loader.c,
main/cli.c, apps/app_skel.c, include/asterisk/module.h,
main/features.c, apps/app_amd.c, channels/chan_alsa.c,
apps/app_url.c, apps/app_externalivr.c, formats/format_gsm.c,
apps/app_milliwatt.c, res/res_speech.c, main/ast_expr2.fl,
apps/app_dial.c, include/asterisk/utils.h, apps/app_page.c,
apps/app_privacy.c, apps/app_fax.c, apps/app_echo.c,
channels/chan_agent.c, apps/app_dahdiras.c, apps/app_disa.c,
pbx/dundi-parser.c, apps/app_transfer.c, res/res_monitor.c,
apps/app_playback.c, include/asterisk/app.h,
channels/chan_misdn.c, apps/app_waitforring.c,
include/asterisk/image.h, apps/app_macro.c,
apps/app_zapateller.c, apps/app_chanspy.c, apps/app_cdr.c,
channels/chan_unistim.c, apps/app_meetme.c, main/utils.c,
res/res_musiconhold.c, apps/app_followme.c,
channels/misdn_config.c, apps/app_controlplayback.c, main/ulaw.c,
main/cdr.c, main/manager.c, channels/console_gui.c,
cdr/cdr_sqlite.c, res/res_agi.c, main/app.c,
apps/app_confbridge.c, main/image.c, apps/app_ivrdemo.c,
apps/app_parkandannounce.c, res/res_clioriginate.c,
apps/app_jack.c, apps/app_while.c, res/res_rtp_asterisk.c,
apps/app_nbscat.c, apps/app_festival.c, res/res_limit.c,
apps/app_waitforsilence.c, apps/app_waituntil.c,
channels/chan_console.c, apps/app_queue.c, apps/app_system.c,
apps/app_getcpeid.c, channels/chan_oss.c,
include/asterisk/features.h, apps/app_flash.c,
apps/app_directed_pickup.c, channels/chan_nbs.c,
include/asterisk/strings.h, include/asterisk/pbx.h,
apps/app_senddtmf.c: Const-ify the world (or at least a good part
of it) This patch adds 'const' tags to a number of Asterisk APIs
where they are appropriate (where the API already demanded that
the function argument not be modified, but the compiler was not
informed of that fact). The list includes: - CLI command handlers
- CLI command handler arguments - AGI command handlers - AGI
command handler arguments - Dialplan application handler
arguments - Speech engine API function arguments In addition,
various file-scope and function-scope constant arrays got 'const'
and/or 'static' qualifiers where they were missing. Review:
https://reviewboard.asterisk.org/r/251/
2009-05-21 19:11 +0000 [r195995] David Vossel <dvossel@digium.com>
* /, channels/chan_iax2.c: Merged revisions 195991 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r195991 | dvossel | 2009-05-21 14:04:56 -0500 (Thu, 21
May 2009) | 14 lines Sign problem calculating timestamp for iax
frame leads to no audio on the receiving peer. There are rare
cases in which a frame's delivery timestamp is slightly less than
the iax2_pvt's offset. This causes the pvt's timestamp to be a
small negative number, but since the timestamp value is unsigned
it looks like a huge positive number. This patch checks for this
negative case and sets the ms to zero. A similar check is already
done right below this one in the 'else' statement. (closes issue
#15032) Reported by: guillecabeza Patches:
chan_iax2.c.patch_timestamp uploaded by guillecabeza (license
380) Tested by: guillecabeza (closes issue #14216) Reported by:
Andrey Sofronov ........
2009-05-21 19:06 +0000 [r195992] Mark Michelson <mmichelson@digium.com>
* main/features.c: Pass connected line updates along during a
bridge.
2009-05-21 17:15 +0000 [r195949] Sean Bright <sean@malleable.com>
* configs/cdr_custom.conf.sample: Rework the cdr_custom.conf.sample
header a bit to reflect the changes in functionality (allowing
multiple mappings).
2009-05-21 15:33 +0000 [r195882] Matthew Nicholson <mnicholson@digium.com>
* main/cdr.c, /, include/asterisk/cdr.h: Merged revisions 195881
via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r195881 | mnicholson | 2009-05-21 10:25:50 -0500 (Thu, 21 May
2009) | 13 lines This commit prevents cdr records with
AST_CDR_FLAG_ANSLOCKED and AST_CDR_FLAG_LOCKED from being updated
in certain cases. This is accomplished by adding two functions to
update the answer time and disposition of calls that checks for
the proper lock flags. These functions are used in the
ast_bridge_call() function so that ForkCDR(A) calls are
respected. This patch also modifies the way ast_bridge_call()
chooses the cdr record to base the bridged_cdr on. Previously the
first unlocked cdr record would be chosen, now instead the first
cdr record is chosen and forked cdr records are moved to the
bridge_cdr. This allows the original cdr record and any forked
cdr records to be properly updated with answer and end times.
(closes issue #13797) Reported by: sh0t Tested by: sh0t (closes
issue #14744) Reported by: deepesh ........
2009-05-20 23:30 +0000 [r195839] Tilghman Lesher <tlesher@digium.com>
* apps/app_stack.c: If a variable had a blank value upon the
initial setting, then it would do nothing. Identified by Dmitry
Andrianov via private email, fixed by me.
2009-05-20 20:45 +0000 [r195763-195798] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Get rid of some duplicated code and correct
a connected line error. When receiving a 200 OK response to an
INVITE, it was possible to transmit two connected line updates
instead of a single one. Furthermore, the second did not have the
proper information present. Now the two have been combined into a
single update and the correct information is presented.
* apps/app_dial.c: Plug a memory leak in app_dial. Since we may
have copied connected line info into the chanlist struct prior to
placing an outbound call, we need to be sure to free the
allocated data when we hang the call up.
2009-05-20 17:33 +0000 [r195636-195698] Joshua Colp <jcolp@digium.com>
* /, main/features.c: Merged revisions 195688 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r195688 | file | 2009-05-20 14:30:25 -0300 (Wed, 20 May 2009) | 5
lines Fix some code that wrongly assumed a pointer would always
be non-NULL when dealing with CDRs after a bridge. (closes issue
#15079) Reported by: barryf ........
* /, apps/app_meetme.c: Merged revisions 195635 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r195635 | file | 2009-05-20 14:14:00 -0300 (Wed, 20 May 2009) | 5
lines Fix a bug where the MeetMe option 'D' did not actually
prompt for the pin. (closes issue #15050) Reported by: pmhaddad
........
2009-05-19 20:59 +0000 [r195589] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c, configs/sip.conf.sample: Add basic support
for handling connected line-related UPDATE requests. SIP purists
may want to look the other way... When COLP/CONP support for SIP
was committed, there was a condition under which Asterisk may
transmit a SIP UPDATE in order to communicate the change in
connected line information. The issue here is that while we could
send a SIP UPDATE message, we were not prepared to receive such
an UPDATE and would always responde with a 501 when we received
an UPDATE. The situation was a bit rough. We really want to be
able to receive UPDATEs having to do with connected line changes,
but the amount of effort involved in properly supporting RFC 3311
was staggering. This commit represents a compromise. First, it
was decided that it is important to only send a SIP UPDATE to an
endpoint that is able to handle one. So, now we have added
parsing of the Allow header into SIP. We store the allowed
methods on SIP peers so that when we communicate with them, we
already will know what we can and cannot send to them. We will
parse the peer's allowed methods when he registers with us. If
the peer is not the type to register with us, but the qualify
option is enabled, then we will use the response to the OPTIONS
request we send the peer to determine the peer's allowed methods.
When the peer's registration expires, or when qualify deems the
peer to be unreachable, we clear the allowed methods from the
peer. For an actual call, we will copy the peer's allowed methods
to the sip_pvt representing the call leg. If we are communicating
with an endpoint which is not a peer, then we will just parse the
Allow header from the first message we receive during the call
and store the information in the sip_pvt. If, during
communication with a peer, we receive a 501 response, then we
will make sure to save the fact that we cannot use that method
when communicating with that peer. Now, with all that
infrastructure in place, the only actual place we use this
information currently is when attempting to send a connected line
change using an UPDATE request. If we cannot send the change
immediately using an UPDATE, we will set the SIP_NEEDREINVITE
flag so that we can send a REINVITE as soon as it is allowed. The
second part of the changes here is for Asterisk to accept UPDATE
requests that have connected line changes. Since we are not fully
supporting RFC 3311, Asterisk will NOT place the UPDATE method in
Allow headers it sends. Instead, if you are communicating with
what you know to be another Asterisk box, you may set the
rpid_update parameter in sip.conf so that we will send UPDATEs to
that Asterisk box. When we send a connected line update, we set a
custom header called "X-Asterisk-rpid-update." On the receiving
end, if Asterisk receives an UPDATE that does not have the
"X-Asterisk-rpid-update" header present, then Asterisk will
respond with a 501 since media-changing UPDATEs are not
supported. We should never get such UPDATEs, since as was stated
earlier, Asterisk does not put UPDATE in its Allow header. If the
custom header is present in the received UPDATE, though, then we
will check the incoming request for connected line updates and
queue the update on the channel where the change occurred.
ABE-1840 ABE-1822
2009-05-19 20:16 +0000 [r195521] Tilghman Lesher <tlesher@digium.com>
* /, apps/app_voicemail.c: Merged revisions 195520 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r195520 | tilghman | 2009-05-19 15:12:20 -0500 (Tue, 19
May 2009) | 7 lines Ensure thread keys are initialized before
attempting to access them. (closes issue #14889) Reported by:
jaroth Patches: app_voicemail.c.patch uploaded by msirota
(license 758) Tested by: msirota, BlargMaN ........
2009-05-19 14:43 +0000 [r195449] Joshua Colp <jcolp@digium.com>
* /, channels/chan_sip.c: Merged revisions 195448 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r195448 | file | 2009-05-19 11:41:45 -0300 (Tue, 19 May 2009) | 7
lines Fix a bug where direct RTP setup would partially occur even
when disabled if the calling channel was answered. (issue #13545)
Reported by: davidw (issue #14244) Reported by: mbnwa ........
2009-05-18 20:52 +0000 [r195370] Tilghman Lesher <tlesher@digium.com>
* res/res_smdi.c, /, include/asterisk/monitor.h, apps/app_queue.c,
include/asterisk/smdi.h, res/res_monitor.c, apps/app_voicemail.c:
Recorded merge of revisions 195366 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r195366 | tilghman | 2009-05-18 15:24:13 -0500 (Mon, 18 May 2009)
| 8 lines Add a similar dependency on SMDI for voicemail as
already exists for ADSI. (closes issue #14846) Reported by: pj
Patches: 20090413__bug14846__1.4.diff.txt uploaded by tilghman
(license 14) 20090507__issue14846__1.6.0.diff.txt uploaded by
tilghman (license 14) 20090507__issue14846__1.6.1.diff.txt
uploaded by tilghman (license 14) ........
2009-05-18 20:49 +0000 [r195365-195369] Eliel C. Sardanons <eliels@gmail.com>
* main/manager.c: Fix the CLI command 'manager show command'
documentation and functionality. The CLI command 'manager show
command' supports passing multiple action names in the same line,
but it was not allowing that because of a incorrect check in the
argumentes counter. Also the documentation was updated to show
that this usage of the command is possible.
* main/manager.c: Rollback commit 195367. The CLI command 'manager
show command' supports passing multiple AMI actions at a time.
The issue with this command was in another place.
* main/manager.c: Avoid autocompleting passed the action name
argument in the CLI command. When running the autocomplete of the
CLI command 'manager show command <action>' it was autocompleting
everything else after the <action> argument, giving an error,
because this command doesn't support multiple AMI action names at
a time.
* res/res_agi.c: Move AGI documentation from static to the XML
form. Move the AGI commands 'receive text', 'receive char' and
'record' static documentation to XML docs.
2009-05-18 19:17 +0000 [r195320] Tilghman Lesher <tlesher@digium.com>
* main/asterisk.c: Move the spawn of astcanary down, until after
the call to daemon(3). This avoids possible conflicts with the
internal implementation of daemon(3). (closes issue #15093)
Reported by: tzafrir Patches: 20090513__issue15093__2.diff.txt
uploaded by tilghman (license 14) Tested by: tzafrir
2009-05-18 18:58 +0000 [r195316] Mark Michelson <mmichelson@digium.com>
* apps/app_externalivr.c: Fix externalivr's setvariable command so
that it properly sets multiple variables. The command had a for
loop that was guaranteed to only execute once since the
continuation operation of the loop would set the input buffer
NULL. I rewrote the loop so that its operation was more obvious,
and it would set multiple variables correctly. I also reduced
stack space required for the function, constified the input
string, and modified the function so that it would not modify the
input string while I was at it. (closes issue #15114) Reported
by: chris-mac Patches: 15114.patch uploaded by mmichelson
(license 60) Tested by: chris-mac
2009-05-18 17:08 +0000 [r195279] Sean Bright <sean@malleable.com>
* cdr/cdr_custom.c: Remove some unused code.
2009-05-18 16:29 +0000 [r195266] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c: The facilityenable parameter does not have
anything to do with pritimer parameters.
2009-05-18 15:55 +0000 [r195210] Sean Bright <sean@malleable.com>
* cdr/cdr_custom.c: Const-ify a string, fix a log message, and use
the correct signature for the load_module function.
2009-05-18 15:53 +0000 [r195207] Joshua Colp <jcolp@digium.com>
* main/frame.c, /: Merged revisions 195206 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r195206 | file | 2009-05-18 12:51:22 -0300 (Mon, 18 May 2009) | 7
lines Fix a typo which caused loss of audio when using G729 in
some scenarios with a smoother present. (closes issue #15105)
Reported by: bamby Patches: process-vad-correctly.diff uploaded
by bamby (license 430) ........
2009-05-18 14:54 +0000 [r195165] Sean Bright <sean@malleable.com>
* configs/cdr_custom.conf.sample, CHANGES, cdr/cdr_custom.c: Allow
cdr_custom to write to multiple files instead of just one. Up to
now, cdr_custom would only accept a single filename/format from
cdr_custom.conf. This change allows you to specify multiple
filename & format directives.
2009-05-18 14:45 +0000 [r195162] Eliel C. Sardanons <eliels@gmail.com>
* apps/app_dial.c, main/pbx.c, apps/app_macro.c: Warn about the use
of the application WaitExten() within a Macro(). Update
applications documentation to warn the user about the use of the
WaitExten() application within a Macro(). Recommend the use of
Read() instead. (closes issue #14444) Reported by: ewieling
2009-05-18 13:56 +0000 [r195089-195096] Joshua Colp <jcolp@digium.com>
* main/rtp_engine.c, /: Merged revisions 195095 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r195095 | file | 2009-05-18 10:53:39 -0300 (Mon, 18 May 2009) | 5
lines Fix a bug where the codecs of the called party leg were not
properly sent back to the caller call leg when reinvited. (closes
issue #13569) Reported by: bkw918 ........
* channels/chan_sip.c: Fix a bug where specifying an empty
outboundproxy would cause packets to get sent to ourself. (closes
issue #15106) Reported by: timeshell
2009-05-18 13:30 +0000 [r195075] Eliel C. Sardanons <eliels@gmail.com>
* main/xml.c: Do not avoid loading the XML documentation if not
XInclude substitution is done.
2009-05-18 12:59 +0000 [r195021] Russell Bryant <russell@digium.com>
* /: Recorded merge of revisions 195020 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r195020 | russell | 2009-05-18 07:57:46 -0500 (Mon, 18 May 2009)
| 5 lines Don't try to unlock a bogus channel. (closes issue
#15144) Reported by: cristiandimache ........
2009-05-16 20:01 +0000 [r194945-194982] Eliel C. Sardanons <eliels@gmail.com>
* Makefile, main/xml.c, doc/appdocsxml.dtd: Allow to include
sections of other parts of the xml documentation. Avoid
duplicating xml documentation by allowing to include other parts
of the xml documentation using XInclude. Example: <xi:include
xpointer="xpointer(/docs/function[@name='CHANNEL']/synopsis)" />
(Insert this line to include the synopsis of the CHANNEL function
xml documentation). It is also possible to include documentation
from other files in the 'documentation/' directory using the
href="" attribute inside a xinclude element. (closes issue
#15107) Reported by: lmadsen (issue #14444) Reported by: ewieling
* main/pbx.c: Fix a missing unlock in case of error, and a missing
free(). Always free the allocated memory for a string field,
because we are always using it (not only when xmldocs are
enabled). Also if there is an error allocating memory for the
string field remember to unlock the list of registered
applications, before returning.
2009-05-15 22:44 +0000 [r194833-194874] David Vossel <dvossel@digium.com>
* /, channels/chan_iax2.c: Merged revisions 194873 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r194873 | dvossel | 2009-05-15 17:43:13 -0500 (Fri, 15
May 2009) | 17 lines IAX2 REGAUTH loop IAX was not sending REGREJ
to terminate invalid registrations. Instead it sent another
REGAUTH if the authentication challenge failed. This caused a
loop of REGREQ and REGAUTH frames. (Related to Security fix
AST-2009-001) (closes issue #14867) Reported by: aragon Tested
by: dvossel (closes issue #14717) Reported by: mobeck Patches:
regauth_loop_update_patch.diff uploaded by dvossel (license 671)
Tested by: dvossel ........
* channels/iax2-parser.h, /, channels/iax2.h, channels/chan_iax2.c,
channels/iax2-parser.c: Merged revisions 194557,194685 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r194557 | dvossel | 2009-05-14 17:59:43 -0500 (Thu, 14 May 2009)
| 10 lines IAX2 "Ghost" Channels There is a bug tracker issue
where people are reporting "Ghost" channels in their 'iax2 show
channels' output. The confusion is caused by channels being
listed as "(NONE)" with format "unknown". These are not channels
of coarse. They are usually just pending registration or poke
requests, but it is confusing output. To help make sense of this
I have added two columns to 'iax2 show channels'. One shows the
first message which started the transaction, and the second shows
the last message sent by either side of the call. This helps
diagnose why the entry exists and why it may not go away. (closes
issue #14207) Reported by: clive18 Review:
https://reviewboard.asterisk.org/r/246/ ........ r194685 |
dvossel | 2009-05-15 10:40:37 -0500 (Fri, 15 May 2009) | 6 lines
Update to previous IAX2 "Ghost" Channels patch. Fixed some
comments made on reviewboard for the previous patch. (issue
#14207) ........
2009-05-15 18:43 +0000 [r194714-194765] Russell Bryant <russell@digium.com>
* /, configs/logger.conf.sample: Merged revisions 194764 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r194764 | russell | 2009-05-15 13:43:18 -0500 (Fri, 15 May 2009)
| 2 lines Fix some spelling fail. ........
* codecs/g722/g722_encode.c, codecs/g722/g722_decode.c: Shuttle
some bits around to address some gain issues with G.722. (closes
AST-209)
* codecs/Makefile, codecs/g722/Makefile (removed): Further simplify
codec_g722 build.
* codecs/Makefile: Actually force running make for g722.
2009-05-15 13:43 +0000 [r194649] Michiel van Baak <michiel@vanbaak.info>
* CREDITS: add eliel
2009-05-15 13:23 +0000 [r194635] Eliel C. Sardanons <eliels@gmail.com>
* doc/appdocsxml.dtd, main/xmldoc.c: Allow to specify an enumlist
inside an enum. It was not possible to use an enumlist inside an
enum: <enumlist> <enum name="aa"> <enumlist> ... </enumlist>
</enum> </enumlist> Now we will be able to insert as many levels
as we want. (closes issue #15112) Reported by: lmadsen
2009-05-15 13:13 +0000 [r194520-194610] Kevin P. Fleming <kpfleming@digium.com>
* include/asterisk/logger.h, tests/test_logger.c (added),
main/logger.c: Add ability for modules to dynamically register
logger levels This patch adds the ability for modules to
dynamically create logger levels for their own use; these are
named levels just like the built-in levels, and can be directed
to any destination that the logger can send any level to, by
including their names in logger.conf. Review:
https://reviewboard.asterisk.org/r/244/
* /: Merged revisions 194509 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r194509 | kpfleming | 2009-05-14 17:23:49 -0500 (Thu, 14 May
2009) | 1 line Update URL to Reviewboard ........
2009-05-14 22:20 +0000 [r194496] Mark Michelson <mmichelson@digium.com>
* /, channels/chan_sip.c: Merged revisions 194484 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r194484 | mmichelson | 2009-05-14 17:17:55 -0500 (Thu, 14 May
2009) | 24 lines Fix a race condition where a reinvite could
trigger a 482 response. The loop detection/spiral detection code
in chan_sip used the owner channel's state as a criterion for
determining if the incoming INVITE is a looped request. The
problem with this is that the INVITE-handling code happens in a
different thread than the thread that marks the owner channel as
being up. As a result, if a reinvite were to come in very
quickly, say from another Asterisk on the same LAN, it was
possible for the reinvite to arrive before the owner channel had
been set to the up state. This patch corrects the problem by
using the invitestate of the sip_pvt instead, since that can be
guaranteed to be set correctly by the time the reinvite arrives.
Since there is a switch statement further in the INVITE-handling
code, the AST_STATE_RINGING state also checks the invitestate of
the sip_pvt in case we should actually be treating the channel as
if it were up already. (closes issue #12215) Reported by: jpyle
Patches: 12215_confirmed.patch uploaded by mmichelson (license
60) Tested by: lmadsen ........
2009-05-14 22:03 +0000 [r194479] Richard Mudgett <rmudgett@digium.com>
* channels/misdn/isdn_lib.h, channels/chan_misdn.c,
channels/misdn/chan_misdn_config.h,
channels/misdn/isdn_msg_parser.c, configs/misdn.conf.sample,
CHANGES, channels/misdn/isdn_lib.c, channels/misdn_config.c: Add
outgoing_colp misdn.conf port parameter. Select what to do with
outgoing COLP information on this port. 0 - Send out COLP
information unaltered. (default) 1 - Force COLP to restricted on
all outgoing COLP information. 2 - Do not send COLP information.
outgoing_colp=0 Also fixed sending the EctInform message so it
always has the required redirectionNumber parameter when the
status is active. JIRA ABE-1853
2009-05-14 21:24 +0000 [r194477] Russell Bryant <russell@digium.com>
* main/features.c: Fix a typo where an equality check should be an
assignment. (closes issue #15103) Reported by: lmsteffan Patches:
transfer_crash.patch uploaded by lmsteffan (license 779)
2009-05-14 17:05 +0000 [r194434] Joshua Colp <jcolp@digium.com>
* apps/app_meetme.c: Fix a bug where the 'T' option to Meetme did
not work. (closes issue #15031) Reported by: Stochastic (closes
issue #13801) Reported by: justdave
2009-05-14 16:22 +0000 [r194430] Tilghman Lesher <tlesher@digium.com>
* main/pbx.c: If the timing ended on a zero, then we would loop
forever. (closes issue #14983) Reported by: teox Patches:
20090513__issue14983.diff.txt uploaded by tilghman (license 14)
Tested by: teox
2009-05-13 15:02 +0000 [r194283] Eliel C. Sardanons <eliels@gmail.com>
* main/manager.c: Do not lock the 'sessions' container, lock the
allocated 'session'. There was a typo in the structure being
locked, and we were locking the 'sessions' container instead of
the 'session' structure thar we are modifying. Reported by
seanbright on #asterisk-dev, thanks!
2009-05-13 13:39 +0000 [r194209] Joshua Colp <jcolp@digium.com>
* res/res_rtp_asterisk.c, /: Merged revisions 194208 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r194208 | file | 2009-05-13 10:38:01 -0300 (Wed, 13 May
2009) | 11 lines Fix RFC2833 issues with DTMF getting duplicated
and with duration wrapping over. (closes issue #14815) Reported
by: geoff2010 Patches: v1-14815.patch uploaded by dimas (license
88) Tested by: geoff2010, file, dimas, ZX81, moliveras (closes
issue #14460) Reported by: moliveras Tested by: moliveras
........
2009-05-13 00:52 +0000 [r194101-194138] Tilghman Lesher <tlesher@digium.com>
* main/pbx.c, /: Merged revisions 194137 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r194137 | tilghman | 2009-05-12 19:52:03 -0500 (Tue, 12 May 2009)
| 7 lines Fix logic for how to proceed with a single digit
extension. (closes issue #15091) Reported by: andrew Patches:
20090512__issue15091.diff.txt uploaded by tilghman (license 14)
Tested by: andrew ........
* main/pbx.c, main/logger.c: Two fixes found while debugging with
ast_backtrace(): 1) If MALLOC_DEBUG is used when concurrently
using ast_backtrace, the free() used in that routine will trigger
an error, because the memory was allocated internally to libc,
where we could not intercept that call to wrap it. Therefore,
it's not memory we knew about, and the free is reported as an
error. 2) Now that channels are objects, the old hack of
initializing a channel to all zeroes no longer works, since we
may try to call something like ast_channel_lock() within a
function on that reference. In that case, it's reported as an
error, because the pointer isn't an object reference.
2009-05-12 22:49 +0000 [r194060] Eliel C. Sardanons <eliels@gmail.com>
* main/manager.c: Fix a crash when logging out from the AMI and
avoid astobj2 warning messages. When the user logout the session
was being destroyed twice and the file descriptor was being
closed twice. The sessions reference counter wasn't used in a
proper way. The 'mansession' structure was being treated as an
astobj2 and we were calling ao2_lock/ao2_unlock causing astobj2
report a warning message and not locking the structure. Also we
were using an ugly naming convention 'destroy_session',
'session_destroy', 'free_session', ... all this "duplicated" code
was merged. (closes issue #14974) Reported by: pj Patches:
manager.diff2 uploaded by eliel (license 64) Tested by: dhubbard,
eliel, mnicholson (closes issue #15088) Reported by: eliel
Review: http://reviewboard.asterisk.org/r/248/
2009-05-12 22:32 +0000 [r194057] Matthew Nicholson <mnicholson@digium.com>
* /, apps/app_queue.c: Merged revisions 194028 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r194028 | mnicholson | 2009-05-12 17:15:45 -0500 (Tue, 12 May
2009) | 16 lines This change modifies app_queue to properly
generate CDR records in failure situations. This involves setting
a proper cdr disposition coresponding to the given failure
condition and ensuring the proper information is stored in the
cdr record. (closes issue #13691) Reported by: dferrer Tested by:
mnicholson (closes issue #13637) Reported by: atis Tested by:
atis ........
2009-05-12 20:40 +0000 [r193956] Tilghman Lesher <tlesher@digium.com>
* /, apps/app_voicemail.c: Merged revisions 193955 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r193955 | tilghman | 2009-05-12 15:39:21 -0500 (Tue, 12
May 2009) | 6 lines Avoid initializing routines if the
authentication fails. Fixes a crash (RR) issue. (closes issue
#14508) Reported by: tiziano Patches:
20090221_2_wrongmailbox.diff.txt uploaded by tiziano (license
377) ........
2009-05-12 20:28 +0000 [r193954] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Update spiral support in trunk and 1.6.X to
match what is in 1.4. In 1.4, a SIP spiral is treated the same
way as a call forward. This works much better than what is
currently in trunk and 1.6.X. The code in trunk and 1.6.X did not
create a new call to the recipient of the spiral, instead trying
to continue the same call. In addition to just being plain wrong,
this also had the side effect of only being able to spiral calls
to other SIP channels. With this in place, as long as call
forwards are honored, SIP spirals will work properly. This means
that it will work for outbound calls made by the Queue, Dial, and
Page applications. For originated calls and spool calls, however,
the spiral will not work properly until a generic call forward
mechanism is introduced into Asterisk. (relates to issue #13630)
2009-05-12 17:29 +0000 [r193870] Tilghman Lesher <tlesher@digium.com>
* apps/app_voicemail.c: Convert a THREADSTORAGE object into a
simple malloc'd object (as suggested by Russell on -dev)
2009-05-12 13:59 +0000 [r193832] Kevin P. Fleming <kpfleming@digium.com>
* apps/app_dial.c, main/pbx.c, apps/app_meetme.c, apps/app_page.c,
main/devicestate.c, apps/app_queue.c, apps/app_transfer.c,
apps/app_playback.c, apps/app_controlplayback.c, main/term.c,
channels/chan_dahdi.c, channels/chan_misdn.c, funcs/func_curl.c,
apps/app_sendtext.c, apps/app_directed_pickup.c,
channels/console_gui.c, main/features.c, apps/app_confbridge.c,
apps/app_externalivr.c, apps/app_chanspy.c,
apps/app_mixmonitor.c, apps/app_stack.c, res/res_odbc.c,
apps/app_voicemail.c: add 'const' qualifiers in various places
where they should have been
2009-05-11 23:04 +0000 [r193756-193757] Tilghman Lesher <tlesher@digium.com>
* apps/app_voicemail.c: Found and fixed a memory leak
* /: Recorded merge of revisions 193755 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r193755 | tilghman | 2009-05-11 17:48:20 -0500 (Mon, 11 May 2009)
| 18 lines Move 300 bytes around on the stack, to make more room
for an extension buffer. This allows more concurrent extensions
to be copied for a single voicemail, without creating a
possibility of upsetting existing users, where a dialplan could
run out of stack space where it had run fine before.
Alternatively, we could have allocated off the heap, but that is
a larger change and would have increased the chance for
instability introduced by this change. This is really solved
starting in 1.6.0.11, as the use of an ast_str buffer allows an
unlimited number of extensions (up to available memory). We
additionally create a new warning message when the buffer length
is exceeded, permitting administrators to see an issue after the
fact, whereas previously the list was silently truncated. (closes
issue #14739) Reported by: p_lindheimer Patches:
20090417__bug14739.diff.txt uploaded by tilghman (license 14)
Tested by: p_lindheimer ........
2009-05-11 22:04 +0000 [r193718] Russell Bryant <russell@digium.com>
* res/res_timing_timerfd.c: Fix some timer state corruption. In
res_timer_timerfd, handle the case that set_rate gets called
while a timer is still in continuous mode. In this case, we want
to remember the configured rate, but not actually set it until
continuous mode has been disabled. Thanks to dvossel for finding
and helping to debug the problem. (closes issue #15080) Reported
by: dvossel Tested by: dvossel
2009-05-11 19:32 +0000 [r193678] Tilghman Lesher <tlesher@digium.com>
* apps/app_voicemail.c: Don't nullify an ast_str pointer. (closes
issue #15061) Reported by: alecdavis
2009-05-11 19:11 +0000 [r193614] Richard Mudgett <rmudgett@digium.com>
* channels/chan_misdn.c, /: Merged revisions 193613 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r193613 | rmudgett | 2009-05-11 14:09:00 -0500 (Mon, 11
May 2009) | 12 lines Sent wrong message to clear a call we
started if the other end has not responed yet. In the state
MISDN_CALLING (i.e. SETUP was sent but no answer has arrived
yet), it is not allowed to clear the call with RELEASE_COMPLETE.
It must be cleared with DISCONNECT. A RELEASE_COMPLETE is only
allowed as an answer to a SETUP. (See Q.931 ch. 5.3.2, 5.3.2.a,
5.3.2.b) Patches: chan-misdn-ccstate7.patch uploaded by customer.
JIRA ABE-1862 ........
2009-05-11 18:01 +0000 [r193545] Leif Madsen <lmadsen@digium.com>
* /, funcs/func_channel.c: Recorded merge of revisions 193544 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r193544 | lmadsen | 2009-05-11 13:35:17 -0400 (Mon, 11 May 2009)
| 7 lines Document CHANNEL(transfercapability) in CLI
documentation. (issue #15073) Reported by: pkempgen Patches:
20090511__issue15073.diff.txt uploaded by tilghman (license 14)
........
2009-05-10 17:07 +0000 [r193502] Joshua Colp <jcolp@digium.com>
* main/bridging.c: Fix a bug where receiving a control frame of
subclass -1 would cause certain channels to get hung up.
2009-05-09 11:33 +0000 [r193459-193461] Russell Bryant <russell@digium.com>
* include/asterisk/event.h: Minor documentation update for
ast_event_queue().
* main/channel.c: Declare private data as static.
2009-05-08 20:32 +0000 [r193387] David Vossel <dvossel@digium.com>
* channels/chan_sip.c: TCP not matching valid peer. find_peer()
does not find a valid peer when using pvt->recv as the
sockaddr_in argument. Because of the way TCP works, the port
number in pvt->recv is not what we're looking for at all. There
is currently only one place that find_peer searches for a peer
using the sockaddr_in argument. If the peer is not found after
using pvt->recv (works for UDP since the port number will be
correct), a temp sockaddr_in struct is made using the Contact
header in the sip_request. This has the correct port number in
it. Review: http://reviewboard.digium.com/r/236/
2009-05-08 19:50 +0000 [r193349] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c: Reset the members' call counts when resetting
queue statistics. This helps to prevent odd scenarios where a
queue will claim to have taken 0 calls, but the members appear to
have taken a non-zero amount. (closes issue #15068) Reported by:
sum Patches: patchreset.patch uploaded by sum (license 766)
Tested by: sum
2009-05-08 15:18 +0000 [r193274] Sean Bright <sean@malleable.com>
* funcs/func_devstate.c: Fix the spelling of UNAVAILABLE in
func_devstate CLI completion.
2009-05-08 14:52 +0000 [r193263] David Vossel <dvossel@digium.com>
* /, channels/misdn_config.c: Merged revisions 193262 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r193262 | dvossel | 2009-05-08 09:51:09 -0500 (Fri, 08
May 2009) | 9 lines "misdn show config" segfaults asterisk, if no
MSN lists (closes issue #14976) Reported by: alecdavis Patches:
misdn_config.diff.txt uploaded by alecdavis (license 585) Tested
by: alecdavis, FabienToune ........
2009-05-08 14:06 +0000 [r193194] Kevin P. Fleming <kpfleming@digium.com>
* /, main/logger.c, configs/logger.conf.sample: Merged revisions
193193 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r193193 | kpfleming | 2009-05-08 09:03:28 -0500 (Fri, 08 May
2009) | 7 lines Make absolute paths for logger channels work
properly (Note: This is not a new feature, it was previously
undocumented and broken.) The Asterisk logger has a feature to
support absolute pathnames for logger channels, but the code
implementing the feature was broken. This has been fixed, and the
absolute path feature is now documented in the sample
logger.conf. ........
2009-05-07 23:42 +0000 [r193120] Tilghman Lesher <tlesher@digium.com>
* main/pbx.c, /: Merged revisions 193119 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r193119 | tilghman | 2009-05-07 18:41:11 -0500 (Thu, 07 May 2009)
| 19 lines Fix Background within a Macro for FreePBX. If the
single digit DTMF is an extension in the specified context, then
go there and signal no DTMF. Otherwise, we should exit with that
DTMF. If we're in Macro, we'll exit and seek that DTMF as the
beginning of an extension in the Macro's calling context. If
we're not in Macro, then we'll simply seek that extension in the
calling context. Previously, someone complained about the
behavior as it related to the interior of a Gosub routine, and
the fix (#14011) inadvertently broke FreePBX (#14940). This
change should fix both of these situations, but with the possible
incompatibility that if a single digit extension does not exist
(but a longer extension COULD have matched), it would have
previously gone immediately to the "i" extension, but will now
need to wait for a timeout. (closes issue #14940) Reported by:
p_lindheimer Patches: 20090420__bug14940.diff.txt uploaded by
tilghman (license 14) Tested by: p_lindheimer ........
2009-05-07 22:24 +0000 [r193077] Richard Mudgett <rmudgett@digium.com>
* channels/chan_misdn.c, /: Merged revisions 193050 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r193050 | rmudgett | 2009-05-07 17:17:06 -0500 (Thu, 07
May 2009) | 5 lines Give a more helpful message when an incoming
call's dialed extension does not match. Added the dialed
extension and context to the chan_misdn messages warning that the
dialed number cannot be matched in the dialplan. ........
2009-05-07 17:51 +0000 [r192933-193006] Tilghman Lesher <tlesher@digium.com>
* funcs/func_odbc.c: Second result should not contain data from the
first result. (closes issue #15039) Reported by: jims Patches:
20090506__issue15039.diff.txt uploaded by tilghman (license 14)
Tested by: jims
* channels/chan_unistim.c: Send DTMF frame before playing back
audio. (closes issue #14858) Reported by: barryf Patches:
20090507__bug14858.diff.txt uploaded by tilghman (license 14)
* /, channels/chan_sip.c: Merged revisions 192932 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r192932 | tilghman | 2009-05-07 11:29:08 -0500 (Thu, 07 May 2009)
| 10 lines Eliminate repetition of fullcontact during
reconstruction. If the fullcontact field appears in both the
sippeers and the sipregs table, then during reconstruction of the
field, it will otherwise be doubled. (closes issue #14754)
Reported by: Alexei Gradinari Patches:
20090506__bug14754.diff.txt uploaded by tilghman (license 14)
Tested by: lmadsen ........
2009-05-06 22:17 +0000 [r192853-192861] Jeff Peeler <jpeeler@digium.com>
* /, main/features.c: Merged revisions 192858 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r192858 | jpeeler | 2009-05-06 17:15:19 -0500 (Wed, 06 May 2009)
| 10 lines Make ParkedCall application stop execution of the
dialplan after hang up Just changed park_exec to always return
non-zero. I really wasn't entirely sure at first if this was a
bug. Decided it was since it would be surprising when not using
ParkedCall in the dialplan to hang up and have dialplan execution
continue. (closes issue #14555) Reported by: francesco_r ........
* main/pbx.c: If no extension was found in the pattern tree, don't
crash.
2009-05-06 17:38 +0000 [r192808] Joshua Colp <jcolp@digium.com>
* channels/chan_iax2.c: Fix a bug where a timer would be created
but not acknowledged. This scenario crept up if chan_iax2 was
loaded with no configuration file present. It would create a
timer and tell it to go at an interval but the thread that
normally acknowledges it would not be created because no
configuration file was present. The timer will now be closed if
no configuration file is present. (closes issue #15014) Reported
by: madkins
2009-05-06 16:28 +0000 [r192772] Tilghman Lesher <tlesher@digium.com>
* main/say.c, doc/lang/urdu.ods (added): Add numbers in Urdu, the
national language of Pakistan (closes issue #15034) Reported by:
nasirq Patches: ast_say_number_full_ur-patch.c uploaded by nasirq
(license 772) urdu.ods uploaded by nasirq (license 772)
2009-05-06 16:09 +0000 [r192634-192736] Joshua Colp <jcolp@digium.com>
* res/res_clialiases.c: Make the code that prevents an infinite
loop from happening into a case insensitive check. (thanks eliel)
* res/res_clialiases.c: Fix an infinite loop with tab completion of
CLI aliases that reference themselves. (closes issue #15020)
Reported by: junky
* /, channels/chan_sip.c: Merged revisions 192633 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r192633 | file | 2009-05-06 10:30:51 -0300 (Wed, 06 May 2009) | 7
lines Update some old logic to stop both begin and end DTMF
frames from reaching the core if rfc2833 is not enabled. (closes
issue #15036) Reported by: dimas Patches: v1-15036.patch uploaded
by dimas (license 88) ........
2009-05-05 20:54 +0000 [r192590] Richard Mudgett <rmudgett@digium.com>
* apps/app_dial.c, channels/chan_sip.c, apps/app_directed_pickup.c,
main/features.c, apps/app_queue.c: Fixed crashes from issue8824
review board channel locking changes. The local struct
ast_party_connected_line connected_caller variable was
uninitialized when the copy function was called.
2009-05-05 19:57 +0000 [r192525] Sean Bright <sean@malleable.com>
* /, static-http/astman.js: Merged revisions 192524 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r192524 | seanbright | 2009-05-05 15:56:11 -0400 (Tue,
05 May 2009) | 11 lines Fix Javascript error when using astman.js
in Internet Explorer. Internet Explorer (tested with 7.0) does
not like trailing commas on constructs like object initializers,
so get rid of them to avoid some errors. (closes issue #15026)
Reported by: rajnishgiri Patches: bug15026.patch uploaded by
seanbright (license 71) Tested by: seanbright ........
2009-05-05 18:23 +0000 [r192430-192462] Joshua Colp <jcolp@digium.com>
* /, main/features.c: Merged revisions 192454 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r192454 | file | 2009-05-05 15:22:27 -0300 (Tue, 05 May 2009) | 8
lines Fix an incorrect assumption that certain values on the
channel will always exist when they may not. The CDR code
involved with bridges wrongly assumed that the currently
executing application and data values will always exist. It is
possible for this to be false when call forwarding is involved.
(closes issue #14984) Reported by: gincantalupo ........
* /, apps/app_followme.c: Merged revisions 192429 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r192429 | file | 2009-05-05 14:43:30 -0300 (Tue, 05 May 2009) | 5
lines Fix a bug where the followme application would continue
trying numbers after the caller hung up. (closes issue #13624)
Reported by: sgenyuk ........
2009-05-05 17:33 +0000 [r192427] Matthew Fredrickson <creslin@digium.com>
* channels/chan_dahdi.c: Revert CPC patch for now, until I decide
whether or not it all should be merged into libss7/1.0 (It's
still in the bug13495 branch and in libss7/trunk)
2009-05-05 14:22 +0000 [r192387] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: Fix a bug with setting t38pt_udptl at the
user or peer level. If an incoming call authenticated as a user
or peer and t38pt_udptl was not set to yes in general then no
UDPTL session would be present and any T38 related things would
fail. This commit changes it so that if after authenticating T38
is enabled but no UDPTL session is present one will be created.
(issue AST-215)
2009-05-05 14:17 +0000 [r192279-192362] Kevin P. Fleming <kpfleming@digium.com>
* main/utils.c, include/asterisk/stringfields.h: Add a more
efficient way of allocating structures that use stringfields This
commit adds an API call that can be used to allocate a structure
along with this stringfield storage in a single allocation.
* main/utils.c, main/astobj2.c, include/asterisk/stringfields.h:
Correct some flaws in the memory accounting code for stringfields
and ao2 objects Under some conditions, the memory allocation for
stringfields and ao2 objects would not have supplied valid
file/function names for MALLOC_DEBUG tracking, so this commit
corrects that.
* main/channel.c, include/asterisk/astobj2.h,
include/asterisk/datastore.h, include/asterisk/channel.h,
main/astobj2.c, main/datastore.c: Properly account for memory
allocated for channels and datastores As in previous commits,
when channels are allocated (with ast_channel_alloc) or
datastores are allocated (with ast_datastore_alloc) properly
account for the memory being owned by the caller, instead of the
allocator function itself.
* main/utils.c, include/asterisk/stringfields.h: Ensure that string
pools allocated to hold stringfields are properly accounted in
MALLOC_DEBUG mode This commit modifies the stringfield pool
allocator to remember the 'owner' of the stringfield manager the
pool is being allocated for, and ensures that pools allocated in
the future when fields are populated are owned by that
file/function.
2009-05-04 22:44 +0000 [r192214] David Vossel <dvossel@digium.com>
* /, channels/chan_iax2.c: Merged revisions 192213 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r192213 | dvossel | 2009-05-04 17:37:31 -0500 (Mon, 04
May 2009) | 11 lines global mohinterpret setting is ignored
mohinterpret and mohsuggest global variables were not copied over
during build_users and build_peers. (closes issue #14728)
Reported by: dimas Patches: v1-14728.patch uploaded by dimas
(license 88) Tested by: dimas, dvossel ........
2009-05-04 19:29 +0000 [r192132-192171] Tilghman Lesher <tlesher@digium.com>
* include/asterisk/autoconfig.h.in, res/res_agi.c: Restore
'asyncagi break' command to 1.6.1 and higher. (closes issue
#14985) Reported by: nikkk Patches: 20090428__bug14985.diff.txt
uploaded by tilghman (license 14)
20090429__bug14985__1.6.1.diff.txt uploaded by tilghman (license
14) Tested by: nikkk
* autoconf/ast_ext_tool_check.m4: Pass libraries in LIBS, not
LDFLAGS. (closes issue #14671) Reported by: Chainsaw Patches:
asterisk-1.6.0.6-toolcheck-libs-not-ldflags.patch uploaded by
Chainsaw (license 723)
2009-05-04 17:42 +0000 [r192096] Leif Madsen <lmadsen@digium.com>
* apps/app_forkcdr.c: Commit documentation changes related to issue
#14801. (issue #14801)
2009-05-04 16:24 +0000 [r192059] Kevin P. Fleming <kpfleming@digium.com>
* include/asterisk/astobj2.h, main/astobj2.c: Ensure that astobj2
memory allocations are properly accounted for when MALLOC_DEBUG
is used This commit ensures that all astobj2 allocated objects
are properly accounted for in MALLOC_DEBUG mode by passing down
the file/function/line information from the module/function that
actually called the astobj2 allocation function.
2009-05-04 15:35 +0000 [r192032] Eliel C. Sardanons <eliels@gmail.com>
* main/xml.c: Do not re-define _POSIX_C_SOURCE if it was already
defined.
2009-05-04 12:52 +0000 [r191919-191997] Kevin P. Fleming <kpfleming@digium.com>
* tests/test_skel.c, tests/test_sched.c: Minor changes in test
modules Correct command description in test_sched.c and include
asterisk/cli.h in test_skel.c, since it's highly unlikely that a
test module will *not* want to provide CLI commands to execute
the tests
* configs/modules.conf.sample: Ensure that by default only one
console channel driver is loaded This configuration file was
changed to ensure that only one console channel driver (chan_oss)
is loaded by default, but the change would only work if
chan_console was not built. Now it will work as expected; if
chan_alsa or chan_console are built and installed, they will not
be loaded unless explicity requested.
* include/asterisk/event.h, include/asterisk/event_defs.h,
main/event.c: Add 'bitflags'-style information elements to event
framework This patch add a new payload type for information
elements, a set of bit flags. The payload is transported as a
32-bit unsigned integer but when matching is performed between
events and subscribers, the matching is done by using a bitwise
AND instead of numeric value comparison. Review:
http://reviewboard.asterisk.org/r/242/
2009-05-03 14:05 +0000 [r191848-191884] Russell Bryant <russell@digium.com>
* Makefile: Remove unnecessary compiler flag
* main/event.c: Do a bit of code cleanup. - convert handling of IE
PLTYPEs to switch statements - add braces to various small blocks
- remove a bit of trailing whitespace - remove a couple of
unnecessary ast_strdupa() uses
2009-05-02 19:02 +0000 [r191775-191785] Kevin P. Fleming <kpfleming@digium.com>
* include/asterisk/logger.h, main/manager.c, pbx/pbx_spool.c,
main/logger.c, apps/app_sms.c, CHANGES, apps/app_verbose.c,
configs/logger.conf.sample: Remove rarely-used
event_log/LOG_EVENT support In discussions today at the Europe
Asterisk Developer Meet-Up, we determined that the event_log was
used in only 9 places in the entire tree, and really was not
needed at all. The users have been converted to use LOG_NOTICE,
or the messages have been removed since other messages were
already in place that provided the same information.
* main/logger.c: Fix an error in queue_log file rotation
optimization code This code was copy-and-pasted without properly
changing references to event_rotate into queue_rotate, so under
some conditions the log rotation would rotate queue_log even
though it was not necessary.
2009-05-02 16:43 +0000 [r191700-191739] Sean Bright <sean@malleable.com>
* channels/chan_dahdi.c: Conditional include ioctl's to change EC
policy based on DAHDI caps. This feels like a sane change
(wouldn't compile without this addition), but I'm not intimately
familiar with this code.
* main/asterisk.c: Update copyright year to 2009
2009-05-01 20:01 +0000 [r191494-191560] Tilghman Lesher <tlesher@digium.com>
* /, channels/chan_sip.c: Merged revisions 191559 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r191559 | tilghman | 2009-05-01 15:00:23 -0500 (Fri, 01 May 2009)
| 6 lines SIP Response 410 maps to cause code 22 (or 23), not 1.
(closes issue #14993) Reported by: BigJimmy Patches: causepatch
uploaded by BigJimmy (license 371) ........
* channels/chan_iax2.c: Set debug message back to DEBUG level.
(closes issue #15007) Reported by: hulber
2009-05-01 18:09 +0000 [r191489] Jeff Peeler <jpeeler@digium.com>
* main/channel.c, /: Merged revisions 191488 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r191488 | jpeeler | 2009-05-01 12:40:46 -0500 (Fri, 01 May 2009)
| 9 lines Fix DTMF not being sent to other side after a partial
feature match This fixes a regression from commit 176701. The
issue was that ast_generic_bridge never exited after the feature
digit timeout had elapsed, which prevented the queued DTMF from
being sent to the other side. This issue was reported to me
directly. ........
2009-05-01 14:58 +0000 [r191419] Joshua Colp <jcolp@digium.com>
* main/audiohook.c: Drop my IRC nickname.
2009-05-01 09:50 +0000 [r191418] TransNexus OSP Development <support@transnexus.com>
* configs/osp.conf.sample, apps/app_osplookup.c: Made security
features optional.
2009-04-30 21:42 +0000 [r191411] Kevin P. Fleming <kpfleming@digium.com>
* channels/chan_dahdi.c, configure,
include/asterisk/autoconfig.h.in, configure.ac, CHANGES: Add
buffer and echo canceller control to CHANNEL() dialplan function
for DAHDI channels Adds ability for CHANNEL() dialplan function,
when used on DAHDI channels, to temporarily change the number of
buffers and/or the buffer policy, and also to enable, disable, or
switch the echo canceller between FAX/data and voice modes.
2009-04-30 17:40 +0000 [r191367] Tilghman Lesher <tlesher@digium.com>
* configure, include/asterisk/autoconfig.h.in, configure.ac,
main/asterisk.c: Detect eaccess (or euidaccess) before using it.
Reported by Andrew Lindh via the -dev list.
2009-04-30 09:11 +0000 [r191300-191332] TransNexus OSP Development <support@transnexus.com>
* apps/app_osplookup.c: Added routing number support.
* apps/app_osplookup.c: Fixed not report source network ID and not
export destination network ID issues.
2009-04-30 06:47 +0000 [r191219-191283] Tilghman Lesher <tlesher@digium.com>
* main/asterisk.c: Change working directory to / under certain
conditions. If backgrounding and no core will be produced, then
changing the directory won't break anything; likewise, if the CWD
isn't accessible by the current user, then a core wasn't possible
anyway. (closes issue #14831) Reported by: chris-mac Patches:
20090428__bug14831.diff.txt uploaded by tilghman (license 14)
20090430__bug14831.diff.txt uploaded by tilghman (license 14)
Tested by: chris-mac
* /: Recorded merge of revisions 191220 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r191220 | tilghman | 2009-04-29 18:10:54 -0500 (Wed, 29 Apr 2009)
| 2 lines Allow H.323 to compile with FDLEAK checking enabled.
........
* channels/h323/ast_h323.cxx, channels/chan_h323.c: Make H.323
compile with FDLEAK detection code enabled
2009-04-29 22:56 +0000 [r191213] Jeff Peeler <jpeeler@digium.com>
* res/res_phoneprov.c: fix typos
2009-04-29 22:23 +0000 [r191211] Tilghman Lesher <tlesher@digium.com>
* main/pbx.c: Part of the merge did not happen correctly, which
resulted in a compile error
2009-04-29 21:13 +0000 [r191177] David Vossel <dvossel@digium.com>
* main/tcptls.c, configs/sip.conf.sample,
include/asterisk/tcptls.h, CHANGES: SIP option to specify
outbound TLS/SSL client protocol. chan_sip allows for outbound
TLS connections, but does not allow the user to specify what
protocol to use (default was SSLv2, and still is if this new
option is not specified). This patch lets the user pick the
SSL/TLS client method for outbound connections in sip. (closes
issue #14770) Reported by: TheOldSaint (closes issue #14768)
Reported by: TheOldSaint Review:
http://reviewboard.digium.com/r/240/
2009-04-29 21:07 +0000 [r191175] Richard Mudgett <rmudgett@digium.com>
* channels/chan_misdn.c, CHANGES: Outgoing PTP redirected calls did
not wait for the COLR from the redirected-to party. For outgoing
PTP redirected calls, you now need to use the inhibit(i) option
on all of the REDIRECTING statements before dialing the
redirected-to party. You still have to set the
REDIRECTING(to-xxx,i) and the REDIRECTING(from-xxx,i) values. The
PTP call will update the redirecting-to presentation when it
becomes available and queue the redirecting update to the calling
channel.
2009-04-29 18:53 +0000 [r191140] Tilghman Lesher <tlesher@digium.com>
* tests/test_substitution.c (added), funcs/func_base64.c,
funcs/func_rand.c, funcs/func_speex.c, funcs/func_md5.c,
funcs/func_module.c, include/asterisk/autoconfig.h.in,
funcs/func_env.c, funcs/func_strings.c, res/res_phoneprov.c,
funcs/func_sysinfo.c, funcs/func_vmcount.c, funcs/func_sha1.c,
funcs/func_logic.c, apps/app_exec.c, funcs/func_groupcount.c,
configure, funcs/func_aes.c, main/ast_expr2f.c, res/res_agi.c,
apps/app_minivm.c, include/asterisk/ast_expr.h, cdr/cdr_custom.c,
main/strings.c, main/pbx.c, funcs/func_dialplan.c,
funcs/func_db.c, funcs/func_timeout.c, funcs/func_lock.c,
funcs/func_cut.c, funcs/func_extstate.c, res/res_config_curl.c,
funcs/func_curl.c, funcs/func_blacklist.c, apps/app_macro.c,
include/asterisk/pbx.h, funcs/func_callerid.c,
apps/app_voicemail.c: Merge str_substitution branch. This branch
adds additional methods to dialplan functions, whereby the result
buffers are now dynamic buffers, which can be expanded to the
size of any result. No longer are variable substitutions limited
to 4095 bytes of data. In addition, the common case of needing
buffers much smaller than that will enable substitution to only
take up the amount of memory actually needed. The existing
variable substitution routines are still available, but users of
those API calls should transition to using the dynamic-buffer
APIs. Reviewboard: http://reviewboard.digium.com/r/174/
2009-04-29 18:32 +0000 [r191136] David Brooks <dbrooks@digium.com>
* pbx/pbx_config.c: Removing crufty code that is no longer
necessary. Code cleanup.
2009-04-29 14:39 +0000 [r191028] David Vossel <dvossel@digium.com>
* main/tcptls.c, main/manager.c, channels/chan_sip.c, main/http.c,
configs/manager.conf.sample, include/asterisk/tcptls.h, CHANGES,
configs/http.conf.sample: Consistent SSL/TLS options across conf
files ast_tls_read_conf() is a new api call for handling SSL/TLS
options across all conf files. Before this change, SSL/TLS
options were not consistent. http.conf and manager.conf required
the 'ssl' prefix while sip.conf used options with the 'tls'
prefix. While the options had different names in different conf
files, they all did the exact same thing. Now, instead of mixing
'ssl' or 'tls' prefixes to do the same thing depending on what
conf file you're in, all SSL/TLS options use the 'tls' prefix.
For example. 'sslenable' in http.conf and manager.conf is now
'tlsenable' which matches what already existed in sip.conf. Since
this has the potential to break backwards compatibility, previous
options containing the 'ssl' prefix still work, but they are no
longer documented in the sample.conf files. The change is noted
in the CHANGES file though. Review:
http://reviewboard.digium.com/r/237/
2009-04-29 08:58 +0000 [r190989-190993] Russell Bryant <russell@digium.com>
* main/indications.c: Log an error message if indications.conf is
not found. (closes issue #14990) Reported by: tzafrir Patches:
indications_err.diff uploaded by tzafrir (license 46)
* apps/app_queue.c: Fix app_queue XML documentation. I think it
would behoove us to force "make validate-docs" to be run after
the XML documentation has been generated if dev-mode is enabled.
(closes issue #14989) Reported by: tzafrir Patches:
app_queue_xml.diff uploaded by tzafrir (license 46)
* main/rtp_engine.c, include/asterisk/channel.h: Resolve Solaris
build issues and add some API documentation. (issue #14981)
Reported by: snuffy
2009-04-28 22:07 +0000 [r190946-190947] Matthew Fredrickson <creslin@digium.com>
* channels/chan_dahdi.c: Add support setting CPC from channel
variable
* channels/chan_dahdi.c: Make sure that we do not clear the down
flag on the BRI during PTMP link transients
2009-04-28 17:31 +0000 [r190904] Tilghman Lesher <tlesher@digium.com>
* doc/tex/cdrdriver.tex: UniqueID column has a maximum size of 150
2009-04-28 14:15 +0000 [r190861-190865] Kevin P. Fleming <kpfleming@digium.com>
* Makefile: Build XML documention from *only* the source files that
have docs in them Change the build process so that
doc/core-en_US.xml is dependent solely on the source files that
have documentation in them, not on all source files.
* Makefile.rules: Remove Makefile rules for bison and flex sources
We never, ever want these files to processed automatically,
because we store the output files in Subversion and users should
never need to rebuild them.
2009-04-28 09:10 +0000 [r190830] TransNexus OSP Development <support@transnexus.com>
* apps/app_osplookup.c: Updated for OSP Toolkit 3.5.
2009-04-27 21:22 +0000 [r190735-190797] Richard Mudgett <rmudgett@digium.com>
* main/channel.c: Fix a small memory leak on error in
ast_channel_alloc().
* channels/misdn/isdn_lib.h, channels/chan_misdn.c, CHANGES,
channels/misdn/isdn_lib.c, funcs/func_redirecting.c: Make PTP
DivertingLegInformation3 message behavior closer to the
specifications. * Wait for a DivertingLegInformation3 message
after receiving a DivertingLegInformation1 message to complete
the redirecting-to information before queuing a redirecting
update to the other channel. * A DivertingLegInformation2 message
should be responded to with a DivertingLegInformation3 when the
COLR is determined. If the call could or does experience another
redirection, you should manually determine the COLR to send to
the switch by setting REDIRECTING(to-pres) to the COLR and
setting REDIRECTING(to-num) = ${EXTEN}. * A
DivertingLegInformation2 message must have an original called
number if the redirection count is greater than one. Since
Asterisk does not keep track of this information, we can only
indicate that the number is not available due to interworking.
2009-04-27 19:34 +0000 [r190726] Tilghman Lesher <tlesher@digium.com>
* main/pbx.c: Don't warn on pipe in the System call. (closes issue
#14979) Reported by: pj
2009-04-27 19:30 +0000 [r190725] Kevin P. Fleming <kpfleming@digium.com>
* /, configure, include/asterisk/autoconfig.h.in: Merged revisions
190721 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r190721 | kpfleming | 2009-04-27 14:29:46 -0500 (Mon, 27 Apr
2009) | 7 lines Fix 'inconsistent line endings' when autoconf
2.63 is used Attempt to make configure script regeneration 'safe'
using autoconf 2.63, which embeds a bare CR into the script, thus
making Subversion complain about inconsistent line endings This
commit changes the MIME type of the configure script to be
'binary' thus making Subversion no longer inspect line endings,
and as a bonus 'svn diff' will no longer try to generate diff
output for it, which is not generally useful anyway. ........
2009-04-27 19:08 +0000 [r190663] Russell Bryant <russell@digium.com>
* res/res_smdi.c, /: Merged revisions 190661-190662 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r190661 | russell | 2009-04-27 14:00:54 -0500 (Mon, 27
Apr 2009) | 9 lines Resolve a crash in res_smdi when used with
chan_dahdi. When chan_dahdi goes to get an SMDI message, it
provides no search criteria. It just grabs the next message that
arrives. This code was written with the SMDI dialplan functions
in mind, since that is now the preferred method of using SMDI.
However, this broke support of it being used from chan_dahdi.
(closes AST-212) ........ r190662 | russell | 2009-04-27 14:03:59
-0500 (Mon, 27 Apr 2009) | 2 lines Fix a typo from 190661.
........
2009-04-27 16:37 +0000 [r190622-190626] Mark Michelson <mmichelson@digium.com>
* doc/tex/channelvariables.tex, apps/app_queue.c: Allow for a
position to be specified when entering a queue. This would allow
for one to add a caller to a specific place in the queue instead
of just placing the caller in the back every time. To help
facilitate some interesting manipulations, a new channel variable
called QUEUEPOSITION has been added. When a caller is removed
from a queue, his position in that queue is stored in the
QUEUEPOSITION variable. One such strategy an administrator can
employ is to allow for the removal of a caller from one queue
followed by the insertion of the same caller into a separate
queue in the same position. Review:
http://reviewboard.digium.com/r/189
* apps/app_queue.c: Update warning message to not have pipes and
contain all options.
2009-04-27 15:18 +0000 [r190586] Joshua Colp <jcolp@digium.com>
* main/manager.c: Fix a bug where we tried to send events out when
no sessions container was present. This commit stops a warning
message (user_data is NULL) from getting output when manager
events get sent before manager is initialized. This happens
because manager is initialized *after* modules are loaded and the
act of loading modules triggers manager events. (issue #14974)
Reported by: pj
2009-04-27 14:46 +0000 [r190577] Mark Michelson <mmichelson@digium.com>
* configs/sip.conf.sample: Remove nonexistent option from
sip.conf.sample. The option to choose which connected line header
to use is not 'rpid_header' but 'sendrpid'
2009-04-24 21:22 +0000 [r190545] David Vossel <dvossel@digium.com>
* main/tcptls.c, main/manager.c, channels/chan_sip.c, main/http.c,
configs/manager.conf.sample, configs/sip.conf.sample,
include/asterisk/tcptls.h, CHANGES, configs/http.conf.sample:
TLS/SSL private key option Adds option to specify a private key
.pem file when configuring TLS or SSL in AMI, HTTP, and SIP.
Before this, the certificate file was used for both the public
and private key. It is possible for this file to hold both, but
most configurations allow for a separate private key file to be
specified. Clarified in .conf files how these options are to be
used. The current conf files do not explain how the private key
is handled at all, so without knowledge of Asterisk's TLS
implementation, it would be hard to know for sure what was going
on or how to set it up. Review:
http://reviewboard.digium.com/r/234/
2009-04-24 17:59 +0000 [r190516-190517] Richard Mudgett <rmudgett@digium.com>
* channels/chan_misdn.c, funcs/func_connectedline.c: There is no
need to use the struct ast_party_connected_line.source update
values. The messages sent by a technology when a connected line
update is received are best determined by the current call state
of the channel. The struct ast_party_connected_line.source value
is really only useful as a possible tracing aid.
* include/asterisk/channel.h: Update comment.
2009-04-24 15:26 +0000 [r190423-190484] Russell Bryant <russell@digium.com>
* include/asterisk/channel.h: Add \since tag for new API calls.
* channels/chan_misdn.c: Fix a build error.
* channels/chan_unistim.c, channels/chan_local.c,
apps/app_dahdiscan.c (removed), main/devicestate.c,
main/autochan.c (added), funcs/func_logic.c,
channels/chan_gtalk.c, channels/chan_iax2.c, main/cli.c,
main/channel.c, build_tools/cflags.xml, channels/chan_dahdi.c,
main/manager.c, funcs/func_odbc.c, apps/app_minivm.c,
main/features.c, res/res_agi.c, main/logger.c,
channels/chan_mgcp.c, res/res_clioriginate.c, main/pbx.c,
channels/chan_sip.c, include/asterisk/autochan.h (added),
channels/chan_bridge.c, main/Makefile, apps/app_softhangup.c,
channels/chan_agent.c, UPGRADE.txt, include/asterisk/channel.h,
CHANGES, funcs/func_global.c, res/res_monitor.c,
apps/app_channelredirect.c, channels/chan_misdn.c,
apps/app_directed_pickup.c, funcs/func_channel.c,
res/snmp/agent.c, include/asterisk/lock.h, apps/app_senddtmf.c,
apps/app_mixmonitor.c, apps/app_chanspy.c, apps/app_voicemail.c:
Convert the ast_channel data structure over to the astobj2
framework. There is a lot that could be said about this, but the
patch is a big improvement for performance, stability, code
maintainability, and ease of future code development. The channel
list is no longer an unsorted linked list. The main container for
channels is an astobj2 hash table. All of the code related to
searching for channels or iterating active channels has been
rewritten. Let n be the number of active channels. Iterating the
channel list has gone from O(n^2) to O(n). Searching for a
channel by name went from O(n) to O(1). Searching for a channel
by extension is still O(n), but uses a new method for doing so,
which is more efficient. The ast_channel object is now a
reference counted object. The benefits here are plentiful. Some
benefits directly related to issues in the previous code include:
1) When threads other than the channel thread owning a channel
wanted access to a channel, it had to hold the lock on it to
ensure that it didn't go away. This is no longer a requirement.
Holding a reference is sufficient. 2) There are places that now
require less dealing with channel locks. 3) There are places
where channel locks are held for much shorter periods of time. 4)
There are places where dealing with more than one channel at a
time becomes _MUCH_ easier. ChanSpy is a great example of this.
Writing code in the future that deals with multiple channels will
be much easier. Some additional information regarding channel
locking and reference count handling can be found in channel.h,
where a new section has been added that discusses some of the
rules associated with it. Mark Michelson also assisted with the
development of this patch. He did the conversion of ChanSpy and
introduced a new API, ast_autochan, which makes it much easier to
deal with holding on to a channel pointer for an extended period
of time and having it get automatically updated if the channel
gets masqueraded. Mark was also a huge help in the code review
process. Thanks to David Vossel for his assistance with this
branch, as well. David did the conversion of the DAHDIScan
application by making it become a wrapper for ChanSpy internally.
The changes come from the
svn/asterisk/team/russell/ast_channel_ao2 branch. Review:
http://reviewboard.digium.com/r/203/
2009-04-24 13:49 +0000 [r190421] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: Fix nat setting on RTP instances. (closes
issue #14827) Reported by: pj
2009-04-23 21:13 +0000 [r190357] Russell Bryant <russell@digium.com>
* /, channels/chan_sip.c: Merged revisions 190356 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r190356 | russell | 2009-04-23 16:07:07 -0500 (Thu, 23 Apr 2009)
| 2 lines Remove a bogus ast_channel_unlock(). ........
2009-04-23 20:42 +0000 [r190349-190352] Tilghman Lesher <tlesher@digium.com>
* main/pbx.c: Labels are sometimes (most of the time?) NULL for
extensions. (closes issue #14895) Reported by: chris-mac Patches:
20090423__bug14895__2.diff.txt uploaded by tilghman (license 14)
Tested by: lmadsen
* include/asterisk/http.h, include/asterisk/utils.h,
main/manager.c, res/res_phoneprov.c, main/http.c, main/utils.c,
res/res_http_post.c, main/astobj2.c: Support HTTP digest
authentication for the http manager interface. (closes issue
#10961) Reported by: ys Patches: digest_auth_r148468_v5.diff
uploaded by ys (license 281) SVN branch
http://svn.digium.com/svn/asterisk/team/group/manager_http_auth
Tested by: ys, twilson, tilghman Review:
http://reviewboard.digium.com/r/223/ Reviewed by:
tilghman,russellb,mmichelson
2009-04-23 19:15 +0000 [r190287] Joshua Colp <jcolp@digium.com>
* channels/chan_local.c, /: Merged revisions 190286 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r190286 | file | 2009-04-23 16:13:18 -0300 (Thu, 23 Apr
2009) | 6 lines Fix a bug in chan_local glare hangup detection.
If both sides of a Local channel were hung up at around the same
time it was possible for one thread to destroy the local private
structure and have the other thread immediately try to remove the
already freed structure from the local channel list. ........
2009-04-23 17:45 +0000 [r190250] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c: Fix reversed behavior of leavewhenempty option
in queues.conf. (closes issue #14650) Reported by: alecdavis
Patches: 14650.patch uploaded by mmichelson (license 60) Tested
by: mmichelson, lmadsen
2009-04-23 16:55 +0000 [r190217] Joshua Colp <jcolp@digium.com>
* apps/app_directed_pickup.c: Fix a double free issue with the
Pickup dialplan application. As part of the pickup process the
connected line information is updated. Part of this process does
a shallow copy of the target channel's connected line information
to a local structure. Once complete the structure contents are
freed. As a result any information in the target channel's
connected line information structure is no longer valid. This
change will now set the contents back to a clean state so that
the freeing of the target channel's connected line information
structure when the channel is destroyed will no longer try to
double free things. (closes issue #14839) Reported by: lmsteffan
2009-04-23 00:44 +0000 [r190154] Terry Wilson <twilson@digium.com>
* funcs/func_strings.c: Fix example that could fail in certain
circumstances
2009-04-22 21:38 +0000 [r190093] Tilghman Lesher <tlesher@digium.com>
* configure, include/asterisk/autoconfig.h.in, configure.ac,
include/asterisk/lock.h: Merged revisions 190092 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r190092 | tilghman | 2009-04-22 16:35:03 -0500 (Wed, 22
Apr 2009) | 7 lines Detect availability of
pthread_rwlock_timedwrlock() before using it. (closes issue
#14930) Reported by: tilghman Patches:
20090420__bug14930.diff.txt uploaded by tilghman (license 14)
Tested by: mvanbaak, tilghman ........
2009-04-22 21:15 +0000 [r190057] Jeff Peeler <jpeeler@digium.com>
* funcs/func_groupcount.c, main/app.c, include/asterisk/channel.h,
main/cli.c: Fix building of chan_h323 with gcc-3.3 There seems to
be a bug with old versions of g++ that doesn't allow a structure
member to use the name list. Rename list member to group_list in
ast_group_info and change the few places it is used. (closes
issue #14790) Reported by: stuarth
2009-04-22 20:07 +0000 [r190000] Terry Wilson <twilson@digium.com>
* funcs/func_strings.c: Add funcs for manipulating delimited lists
in the dialplan Adds PUSH and POP for appending to and
retrieving/removing from the end of a list and UNSHIFT and SHIFT
for insert to and retrieiving/ removing from the beginning of a
list. Review: http://reviewboard.digium.com/r/230
2009-04-22 19:23 +0000 [r189993] Jeff Peeler <jpeeler@digium.com>
* channels/h323/ast_h323.cxx, channels/chan_h323.c,
channels/h323/chan_h323.h: Make chan_h323 respect packetization
settings and fix small reload issue. Previously, packetization
settings were ignored and now they are not. A new config option
'autoframing' has been added to mirror the way chan_sip handles
it. Turning on the autoframing option (available both as a global
option or per peer) overrides the local settings with the remote
packetization settings. Testing was performed with varying
packetization levels with the following codecs: ulaw, alaw, gsm,
and g729. Also, an unrelated config reload issue has been fixed
in the case of the config file not changing. (closes issue
#12415) Reported by: pj Patches:
2009012200_h323packetization.diff.txt uploaded by mvanbaak
(license 7), modified by me
2009-04-22 16:56 +0000 [r189951] Russell Bryant <russell@digium.com>
* main/features.c: Fix call parking callback. Pipes -> Commas.
2009-04-22 16:01 +0000 [r189911] Tilghman Lesher <tlesher@digium.com>
* channels/chan_unistim.c: Do not continue to receive DTMF, when
the channel is hungup and about to be destroyed. (closes issue
#14858) Reported by: barryf Patches: 20090421__bug14858.diff.txt
uploaded by tilghman (license 14) Tested by: barryf
2009-04-22 14:30 +0000 [r189850] Michiel van Baak <michiel@vanbaak.info>
* /, contrib/scripts/get_ilbc_source.sh: Merged revisions 189849
via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r189849 | mvanbaak | 2009-04-22 16:29:28 +0200 (Wed, 22 Apr 2009)
| 12 lines replace sed with tr to remove \r from downloaded file
On some systems, sed does not recognize \r in the pattern the way
it was used here. Use tr instead because this works the same
across systems. (closes issue #14936) Reported by: leobrown
Patches: 2009042201_14936.diff.txt uploaded by mvanbaak (license
7) Tested by: leobrown, mvanbaak ........
2009-04-22 06:33 +0000 [r189813] Tilghman Lesher <tlesher@digium.com>
* configure, configure.ac: Detect liblua on SuSE, and add libm for
linking for Fedora. (Reported via the -dev list, Subject:
Compiling Asterisk with LUA)
2009-04-21 20:28 +0000 [r189771] David Vossel <dvossel@digium.com>
* channels/chan_sip.c: Fixes segfault when switching UDP to TCP in
sip.conf after reload. If transport in sip.conf is switched from
UDP to TCP, Asterisk segfaults right after issuing a sip reload.
The problem is the socket type is changed to TCP but the fd may
still be present for UDP. Later, when the TCP session should be
created or set using an existing one, it isn't because the old
file descriptor is still present. Now every time transport is
changed during a sip.conf reload, the file descriptor is set to
-1, signifying it must be created or found. (closes issue #14727)
Reported by: pj Tested by: dvossel Review:
http://reviewboard.digium.com/r/229/
2009-04-21 17:44 +0000 [r189735] Richard Mudgett <rmudgett@digium.com>
* channels/misdn/isdn_lib_intern.h, channels/misdn/isdn_lib.h,
channels/chan_misdn.c, channels/misdn/chan_misdn_config.h,
channels/misdn/ie.c, channels/misdn/isdn_msg_parser.c,
configs/misdn.conf.sample, CHANGES, channels/misdn/isdn_lib.c,
channels/misdn_config.c: Added CCBS/CCNR Party A support and
enhanced COLP support. This change adds the following features to
chan_misdn: * CCBS/CCNR Party A support for PTMP and PTP modes. *
Enhances COLP support for call diversion and explicit call
transfer. These enhanced features require a modified version of
mISDN. The latest modified mISDN v1.1.x based version is
available at: http://svn.digium.com/svn/thirdparty/mISDN/trunk
http://svn.digium.com/svn/thirdparty/mISDNuser/trunk Taged
versions of the modified mISDN code are available under:
http://svn.digium.com/svn/thirdparty/mISDN/tags
http://svn.digium.com/svn/thirdparty/mISDNuser/tags Review:
http://reviewboard.digium.com/r/218/ Merged from
team/rmudgett/misdn_facility branch.
2009-04-21 15:54 +0000 [r189629-189665] Doug Bailey <dbailey@digium.com>
* utils/muted.c, /: Merged revisions 189664 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r189664 | dbailey | 2009-04-21 10:52:13 -0500 (Tue, 21 Apr 2009)
| 2 lines Remove daemon call on systems that do not support
forking. ........
* /, configure, include/asterisk/autoconfig.h.in,
include/asterisk/compat.h, configure.ac: Merged revisions 189601
via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r189601 | dbailey | 2009-04-21 09:00:55 -0500 (Tue, 21 Apr 2009)
| 3 lines Add check in configure script to check for GLOB_NOMAGIC
and GLOB_BRACE in glob.h This allows config.c to compile when
linked against uclibc that does not support these parameters
........
2009-04-20 22:10 +0000 [r189539] Tilghman Lesher <tlesher@digium.com>
* main/stdtime/localtime.c: Use nanosleep instead of poll. This is
not just because mmichelson suggested it, but also because Mac OS
X puked on my poll().
2009-04-20 21:29 +0000 [r189495-189516] Terry Wilson <twilson@digium.com>
* apps/app_dial.c, /: Merged revisions 189465 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r189465 | twilson | 2009-04-20 16:10:27 -0500 (Mon, 20 Apr 2009)
| 2 lines Update CDR appropriately when AST_CAUSE_NO_ANSWER is
set ........
* apps/app_dial.c, /: Merged revisions 189463 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r189463 | twilson | 2009-04-20 16:00:52 -0500 (Mon, 20 Apr 2009)
| 2 lines Don't treat a NOANSWER like a CHANUNAVAIL ........
2009-04-20 21:09 +0000 [r189464] Sean Bright <sean@malleable.com>
* /, res/ael/ael.tab.c, res/ael/ael.y: Merged revisions 189462 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r189462 | seanbright | 2009-04-20 16:58:39 -0400 (Mon, 20 Apr
2009) | 13 lines Properly handle @s within hints in AEL. AEL was
not handling the case of a device hint containing an @ symbol,
which caused parking hints (e.g. hint(park:exten@context)) to
error out the parser. This patch makes AEL treat the @ the same
way it treats colon and ampersand now, meaning the characters are
included in verbatim. (closes issue #14941) Reported by: bpgoldsb
Patches: bug14941.patch uploaded by seanbright (license 71)
Tested by: bpgoldsb ........
2009-04-20 19:28 +0000 [r189419] Doug Bailey <dbailey@digium.com>
* main/manager.c, /, main/db1-ast/recno/rec_open.c,
channels/chan_iax2.c: Merged revisions 189391 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r189391 | dbailey | 2009-04-20 14:10:56 -0500 (Mon, 20 Apr 2009)
| 4 lines Clean up problem with manager implementation of mmap
where it was not testing against MAP_FAILED response. Got rid of
shadowed variable used in processign the mmap results. Change
test of mmap results to compare against MAP_FAILED ........
2009-04-20 17:05 +0000 [r189350] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: Fix a bug with non-UDP connections that
caused dialogs to not get freed. This issue crept up because of a
reference count issue on non-UDP based dialogs. The dialog
reference count was increased when transmitting a packet reliably
but never decreased. This caused the dialog structure to hang
around despite being unlinked from the dialogs container. (closes
issue #14919) Reported by: vrban
2009-04-20 14:05 +0000 [r189278] Mark Michelson <mmichelson@digium.com>
* main/channel.c, /: Merged revisions 189277 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r189277 | mmichelson | 2009-04-20 09:04:41 -0500 (Mon, 20 Apr
2009) | 12 lines Move the check for chan->fdno == -1 to after the
zombie/hangup check. Many users were finding that their hung up
channels were staying up and causing 100% CPU usage. (issue
#14723) Reported by: seadweller Patches: 14723_1-4-tip.patch
uploaded by mmichelson (license 60) Tested by: falves11, bamby
........
2009-04-18 01:28 +0000 [r189204] David Vossel <dvossel@digium.com>
* /, channels/chan_agent.c: Merged revisions 189203 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r189203 | dvossel | 2009-04-17 20:27:19 -0500 (Fri, 17
Apr 2009) | 12 lines Fixed autologoff in agents.conf not working
when agent logs in via AgentLogin app An agent logs in by calling
an extension that calls the AgentLogin app. In agents.conf
ackcall=always is set, so when they get a call they have the
choice to either acknowledge it or ignore it. autologoff=10 is
set as well, so if the agent ignores the call over 10sec one may
assume that the agent should be logged out (and in this case
hungup on as well), but this was not happening. (closes issue
#14091) Reported by: evandro Patches: autologoff.diff uploaded by
dvossel (license 671) Review:
http://reviewboard.digium.com/r/225/ ........
2009-04-17 21:48 +0000 [r189137] Richard Mudgett <rmudgett@digium.com>
* channels/chan_misdn.c, /, channels/misdn/isdn_lib.c: Merged
revisions 188833,189134 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r188833 | rmudgett | 2009-04-16 16:37:58 -0500 (Thu, 16 Apr 2009)
| 4 lines Only disable mISDN DSP if Asterisk DSP is enabled.
Leave jitter setting alone. JIRA ABE-1835 ........ r189134 |
rmudgett | 2009-04-17 16:27:55 -0500 (Fri, 17 Apr 2009) | 4 lines
Modifed/added some debug messages. JIRA ABE-1835 ........
2009-04-17 20:20 +0000 [r189097] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Prevent a crash when SIP blonde transferring
an unbridged call. If one attempts to use the attended transfer
button on a SIP phone to transfer an unbridged call (such as a
call to an IVR) but hangs up while the target of the transfer is
still ringing, we need to not crash. The problem was that
ast_hangup was called from outside the channel thread. AST-211
2009-04-17 19:36 +0000 [r189077] Sean Bright <sean@malleable.com>
* main/asterisk.c: Fix copy/paste error with 'transmit silence'
flag.
2009-04-17 15:44 +0000 [r189010] Matthew Nicholson <mnicholson@digium.com>
* main/pbx.c, /: Merged revisions 189009 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r189009 | mnicholson | 2009-04-17 10:43:09 -0500 (Fri, 17 Apr
2009) | 5 lines Make Busy() application set the CDR disposition
to BUSY. (closes issue #14306) Reported by: cristiandimache
........
2009-04-17 14:44 +0000 [r188947] Joshua Colp <jcolp@digium.com>
* /, channels/chan_sip.c: Merged revisions 188946 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r188946 | file | 2009-04-17 11:41:25 -0300 (Fri, 17 Apr 2009) |
15 lines Fix a bug where a value used to create the channel name
was bogus. This commit fixes the scenario where an incoming call
is authenticated using a peer entry. Previously the channel name
was created using either the username setting from the sip.conf
entry or the IP address that the call came from. Now the channel
name will be created using the peer name itself. This commit will
not change the way the channel name is generated for users or
friends. (closes issue #14256) Reported by: Nick_Lewis Patches:
chan_sip.c-chname.patch uploaded by Nick (license 657) Tested by:
Nick_Lewis, file ........
2009-04-17 14:33 +0000 [r188942] Mark Michelson <mmichelson@digium.com>
* main/pbx.c: Fix a spacing issue that I claimed I would when I
committed this code. Nothing major though.
2009-04-17 14:26 +0000 [r188938] Joshua Colp <jcolp@digium.com>
* channels/chan_dahdi.c, /: Merged revisions 188937 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r188937 | file | 2009-04-17 11:25:57 -0300 (Fri, 17 Apr
2009) | 4 lines Fix a situation where the DAHDI channel private
structure lock was not unlocked when it should have been. (issue
AST-210) ........
2009-04-17 13:29 +0000 [r188901] Mark Michelson <mmichelson@digium.com>
* main/pbx.c: Several fixes to the extenpatternmatchnew logic. 1.
Differentiate between literal characters in an extension and
characters that should be treated as a pattern match. Prior to
these fixes, an extension such as NNN would be treated as a
pattern, rather than a literal string of N's. 2. Fixed the logic
used when matching an extension with a bracketed expression, such
as 2[5-7]6. 3. Removed all areas of code that were executed when
NOT_NOW was #defined. The code in these areas had the potential
to crash, for one thing, and the actual intent of these blocks
seemed counterproductive. 4. Fixed many many coding guidelines
problems I encountered while looking through the corresponding
code. 5. Added failure cases and warning messages for when
duplicate extensions are encountered. 6. Miscellaneous fixes to
incorrect or redundant statements. (closes issue #14615) Reported
by: steinwej Tested by: mmichelson Review:
http://reviewboard.digium.com/r/194/
2009-04-16 21:57 +0000 [r188774-188836] Tilghman Lesher <tlesher@digium.com>
* /, channels/chan_sip.c: Merged revisions 188835 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r188835 | tilghman | 2009-04-16 16:41:13 -0500 (Thu, 16 Apr 2009)
| 7 lines Only update realtime, if global option rtupdate !=
false (closes issue #14885) Reported by: deepesh Patches:
20090413__bug14885.diff.txt uploaded by tilghman (license 14)
Tested by: deepesh ........
* /, apps/app_voicemail.c: Merged revisions 188773 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r188773 | tilghman | 2009-04-16 16:02:29 -0500 (Thu, 16
Apr 2009) | 4 lines Umask should not be exported into global
namespace. (closes issue #14912) Reported by: jcapp ........
2009-04-16 19:30 +0000 [r188742] David Vossel <dvossel@digium.com>
* channels/chan_sip.c: SIP state notify reorganization What I've
done here is simply break up how a state NOTIFY is built.
Originally both the XML and sip header information were built
within the same function. While this does work, it does not allow
for the creation of multipart/related message bodies that can
contain multiple XML entries with only one sip header. Now a
separate function builds the XML for each notify. This patch also
makes maintaining and modifying state notifications in the future
much less of a pain. Review: http://reviewboard.digium.com/r/224/
2009-04-16 13:42 +0000 [r188705] Joshua Colp <jcolp@digium.com>
* channels/chan_dahdi.c: Fix a bug with the dahdi_setoption
callback in chan_dahdi. This function incorrectly reported
success even if the option was unsupported. This was exposed by
the options to change the underlying channel format. The function
now returns a failure if the option is unsupported.
2009-04-15 22:10 +0000 [r188647] David Vossel <dvossel@digium.com>
* channels/chan_dahdi.c, /: Merged revisions 188646 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r188646 | dvossel | 2009-04-15 17:08:40 -0500 (Wed, 15
Apr 2009) | 12 lines National prefix inserted even when caller ID
not available When the caller ID is restricted, the expected
behavior is for the caller id to be blank. In chan_dahdi, the
national prefix is placed onto the callers number even if its
restricted (empty) causing the caller id to be the national
prefix rather than blank. (closes issue #13207) Reported by:
shawkris Patches: national_prefix.diff uploaded by dvossel
(license 671) Review: http://reviewboard.digium.com/r/220/
........
2009-04-15 20:17 +0000 [r188544-188585] Mark Michelson <mmichelson@digium.com>
* /, main/file.c: Merged revisions 188582 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r188582 | mmichelson | 2009-04-15 15:04:20 -0500 (Wed, 15 Apr
2009) | 7 lines Update ast_readvideo_callback to match
ast_readaudio_callback. This fixes potential refcount errors that
may occur on ast_filestreams. AST-208 ........
* apps/app_dial.c: Make the cancellation of the dial timeout on a
call forward optional. This introduces the 'z' option to
app_dial. With it set, a call forward will cancel any timeout
originally set for this instance of the Dial application. AST-207
2009-04-15 14:57 +0000 [r188515] Jeff Peeler <jpeeler@digium.com>
* channels/chan_dahdi.c: Don't try to do anything in
pri_check_restart with service messages unless libpri supports
it.
2009-04-14 23:28 +0000 [r188470] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c: Fix a couple of queue member reference leaks.
2009-04-14 17:40 +0000 [r188413] Joshua Colp <jcolp@digium.com>
* res/res_rtp_asterisk.c: Fix an incorrect clock rate when sending
T140 text. (closes issue #14029) Reported by: epicac
2009-04-14 16:49 +0000 [r188342-188378] Jeff Peeler <jpeeler@digium.com>
* channels/chan_dahdi.c, CHANGES: change some capitalization
* channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, configure,
include/asterisk/autoconfig.h.in, configure.ac, CHANGES: Add
service maintenance message support This is the companion commit
to libpri r732. Service messages are now supported for switch
types 4ess/5ess. A new option service_message_support has been
added to chan_dahdi.conf and is noted in the sample config file.
The service message support is turned off by default. The current
implementation relies on AstDB to keep track of channel state,
which allows the statuses to be preserved across Asterisk
restarts. Below is a description of the storage format. The state
and reason for the service state are in the form
<state>:<reason>, where: <state> ::= { 'O' } // 'O' – Out Of
Service <reason> ::= { '0' | '1' | '2' | '3' }, where: '0' – No
reason (backwards compatibility) '1' – NEAR END '2' – FAR END '3'
– both NEAR and FAR END The new CLI commands to handle channel
service state are: pri service disable channel <chan> pri service
enable channel <chan> Many people contributed to the development
of this functionality. Because I entered at the very end I do not
know the exact history. Special thanks to all who moved the bug
forward one way or another: cmaj, PCadach, markster, mattf,
drmac, MikeJ, serge-v, murf, kanelbullar, Seb7, tilghman,
lmadsen, and especially dhubbard (he answered lots of my
questions and did a large portion of the work) (closes issue
#3450) Reported by: cmaj
2009-04-14 14:22 +0000 [r188283-188284] Olle Johansson <oej@edvina.net>
* doc/manager_1_1.txt: New actions should go under "New Actions",
not "new events"
* main/xmldoc.c, apps/app_jack.c: Making sure we have references to
external libraries. Note: Update h.323 with the recent changes
too
2009-04-14 13:14 +0000 [r188247] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: Fix a bug with the change I made yesterday
to outbound proxy support. Per discussion with oej on IRC we need
the actual IP address, not the outbound proxy IP address, in the
sa field. This change matches the already existing code for all
other uses of the outbound proxy setting.
2009-04-14 05:45 +0000 [r188206-188210] Tilghman Lesher <tlesher@digium.com>
* main/pbx.c: As suggested by Russell, warn users when their
dialplan arguments contain pipes, but not commas.
* utils/smsq.c: Application delimiter is ',', not '|'. (closes
issue #14881) Reported by: stegro Patches: smsq.patch uploaded by
stegro (license 752)
2009-04-13 19:31 +0000 [r188102] Mark Michelson <mmichelson@digium.com>
* res/res_musiconhold.c: Fix another crash related to cached
realtime music on hold. This was another off-by-one problem
caused by moh_register.
2009-04-13 16:28 +0000 [r188067] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: Fix a bug where using an outbound proxy
would cause the local address to be 127.0.0.1. Copy the outbound
proxy IP address into the SIP dialog structure as the IP address
we will be sending to. This has to be done because the logic that
determines what local IP address to use in the SIP messages is
not aware of an outbound proxy being in place. It only knows what
IP address we are sending to. (closes issue #12006) Reported by:
mnicholson
2009-04-13 14:17 +0000 [r188032] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c: Set all queue variables on both the caller and
member channels. This allows for the variables to be accessed if
a member macro is run. Thanks to Grigoriy Puzankin for bringing
this up on the -dev list.
2009-04-10 20:26 +0000 [r187906] Jeff Peeler <jpeeler@digium.com>
* channels/Makefile: Fix module embedding for chan_h323. Include
libchanh323.a in the modules.link file so that all the symbols
can be resolved at link time. (closes issue #11966) Reported by:
dome Patches: issue_11966.patch uploaded by kpfleming (license
421) Tested by: jpeeler
2009-04-10 18:56 +0000 [r187830] Mark Michelson <mmichelson@digium.com>
* channels/chan_local.c: Indicating connected line or redirecting
updates were missing a call to lock the local_pvt.
2009-04-10 18:14 +0000 [r187772-187773] Joshua Colp <jcolp@digium.com>
* res/res_rtp_asterisk.c, main/rtp_engine.c: Change how we set the
local and remote address. The code will now only change the
address and port. It will not overwrite any other values.
* channels/chan_jingle.c, channels/chan_unistim.c,
res/res_rtp_asterisk.c, main/rtp_engine.c, channels/chan_sip.c,
channels/chan_skinny.c, channels/chan_h323.c,
channels/chan_gtalk.c, channels/chan_mgcp.c: Fix some
uninitialized memory notices that appeared under valgrind.
2009-04-10 17:32 +0000 [r187770] Mark Michelson <mmichelson@digium.com>
* apps/app_dial.c: Make sure tc is unlocked before calling ast_call
since calling a Local channel could result in a deadlock.
2009-04-10 17:29 +0000 [r187764] Tilghman Lesher <tlesher@digium.com>
* contrib/scripts/realtime_pgsql.sql, /,
contrib/scripts/sip-friends.sql: Merged revisions 187763 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r187763 | tilghman | 2009-04-10 12:28:46 -0500 (Fri, 10 Apr 2009)
| 2 lines Add lastms column to the contributed table designs
........
2009-04-10 16:51 +0000 [r187721] Kevin P. Fleming <kpfleming@digium.com>
* build_tools/embed_modules.xml: clean up some patterns for files
to remove add embedding support for bridge and test modules
2009-04-10 16:26 +0000 [r187680-187714] Mark Michelson <mmichelson@digium.com>
* channels/chan_local.c: ast_strdup failures aren't really failures
if the original value was NULL.
* main/channel.c: Don't let ast_channel_alloc fail if explicitly
passed NULL cid_name or cid_number. This also fixes a small
memory leak.
2009-04-10 16:00 +0000 [r187675] Russell Bryant <russell@digium.com>
* tests/test_heap.c, tests/test_sched.c: Disable test modules by
default.
2009-04-10 15:59 +0000 [r187674] Tilghman Lesher <tlesher@digium.com>
* channels/chan_sip.c: Ensure pvt is not NULL before dereferencing
it. (closes issue #14784) Reported by: pj
2009-04-10 15:49 +0000 [r187673] David Vossel <dvossel@digium.com>
* apps/app_dial.c: Even more changes concerning r187426. Revised
where locks are placed yet once again. ast_call() should not be
called with a channel locked. could cause deadlock issues with
local channels.
2009-04-10 15:11 +0000 [r187636] Kevin P. Fleming <kpfleming@digium.com>
* include/asterisk/logger.h, main/logger.c, apps/app_verbose.c,
configs/logger.conf.sample: revert addition of LOG_SECURITY log
channel; after further discussion, a much better solution will be
used
2009-04-10 14:53 +0000 [r187634-187635] Richard Mudgett <rmudgett@digium.com>
* channels/misdn/isdn_lib.h, channels/chan_misdn.c,
channels/misdn/isdn_lib.c: Miscellaneous minor changes to
chan_misdn. * Miscellaneous spacing and comment changes. * Minor
code rearangements. * Miscellaneous doxygen comments.
* channels/chan_misdn.c: Make chan_misdn_log() avoid generating the
log message if logging is disabled.
2009-04-10 03:55 +0000 [r187599] Tilghman Lesher <tlesher@digium.com>
* main/channel.c, main/pbx.c, main/manager.c,
include/asterisk/linkedlists.h, main/features.c, main/http.c,
main/app.c, include/asterisk/lock.h, main/audiohook.c,
main/bridging.c: Modify headers and macros, according to
Russell's suggestions on the -dev list
2009-04-09 21:06 +0000 [r187560] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c, configs/sip.conf.sample: Add a new option,
mwi_from, to sip.conf. This allows for you to change the From
header for outgoing MWI NOTIFY requests. Prior to this, the best
you could do was to set a callerid in the general section of
sip.conf. The problem was that this was used for all outbound
requests, not just MWI NOTIFY requests. AST-201
2009-04-09 20:40 +0000 [r187556] David Vossel <dvossel@digium.com>
* apps/app_dial.c: More changes concerning r187426. Revised where
locks are placed.
2009-04-09 19:10 +0000 [r187491] Jeff Peeler <jpeeler@digium.com>
* apps/app_dial.c, main/pbx.c, include/asterisk/pbx.h, CHANGES: Add
ability for dialplan execution to continue when caller hangs up.
The F option to app_dial has been modified to accept no
parameters and perform the above functionality. I don't see
anywhere else that is doing function overloading, but this really
is the best place for this operation because: - It makes it close
to the 'g' option in the argument list which provides similar
functionality. - The existing code to support the current F
option provides a very convienient location to add this new
feature. (closes issue #12381) Reported by: michael-fig
2009-04-09 18:58 +0000 [r187488] Mark Michelson <mmichelson@digium.com>
* /, channels/chan_sip.c: Merged revisions 187484 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r187484 | mmichelson | 2009-04-09 13:51:20 -0500 (Thu, 09 Apr
2009) | 18 lines Handle a SIP race condition (reinvite before an
ACK) properly. RFC 5047 explains the proper course of action to
take if a reINVITE is received before the ACK from a previous
invite transaction. What we are to do is to treat the reINVITE as
if it were both an ACK and a reINVITE and process it normally.
Later, when we receive the ACK we had been expecting, we will
ignore it since its CSeq is less than the current iseqno of the
sip_pvt representing this dialog. (closes issue #13849) Reported
by: klaus3000 Patches: 13849_v2.patch uploaded by mmichelson
(license 60) Tested by: mmichelson, klaus3000 ........
2009-04-09 18:40 +0000 [r187483] Tilghman Lesher <tlesher@digium.com>
* main/manager.c, /, include/asterisk/linkedlists.h,
include/asterisk/lock.h: Merged revisions 187428 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r187428 | tilghman | 2009-04-09 13:08:20 -0500 (Thu, 09
Apr 2009) | 8 lines Race condition between ast_cli_command() and
'module unload' could cause a deadlock. Add lock timeouts to
avoid this potential deadlock. (closes issue #14705) Reported by:
jamessan Patches: 20090320__bug14705.diff.txt uploaded by
tilghman (license 14) Tested by: jamessan ........
2009-04-09 17:39 +0000 [r187426] David Vossel <dvossel@digium.com>
* apps/app_dial.c: Fixes deadlock caused by calling get_cid_name
with chan locked. get_cid_name should not be called with a
channel lock. get_cid_name calls ast_get_hint which eventually
calls pbx_find_extension. pbx_find_extension starts and stops
autoservice which should not be done with a channel lock, so
get_cid_name should not be called with one.
2009-04-09 17:34 +0000 [r187421-187424] Mark Michelson <mmichelson@digium.com>
* res/res_musiconhold.c: Use safe macro practices even though they
really aren't necessary.
* res/res_musiconhold.c: Fix a crash in res_musiconhold when using
cached realtime moh. The moh_register function links an mohclass
and then immediately unrefs the class since the container now has
a reference. The problem with using realtime music on hold is
that the class is allocated, registered, and started in one fell
swoop. The refcounting logic resulted in the count being off by
one. The same problem did not happen when using a static config
because the allocation and registration of an mohclass is a
separate operation from starting moh. This also did not affect
non-cached realtime moh because the classes are not registered at
all. I also have modified res_musiconhold to use the _t_ variants
of the ao2_ functions so that more info can be gleaned when
attempting to trace the refcounts. I found this to be incredibly
helpful for debugging this issue and there's no good reason to
remove it. (closes issue #14661) Reported by: sum
2009-04-09 17:20 +0000 [r187363-187381] Tilghman Lesher <tlesher@digium.com>
* channels/chan_sip.c: Allow '/' in username portion of register;
this is a regression. (closes issue #14668) Reported by: Netview
* /, channels/chan_sip.c, apps/app_sendtext.c: Merged revisions
187362 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r187362 | tilghman | 2009-04-09 11:38:37 -0500 (Thu, 09 Apr 2009)
| 3 lines Permit zero-length text messages in SIP. (Related to an
issue posted to the -users list, subject "AEL2, BASE64_DECODE and
hexadecimal") ........
2009-04-09 16:27 +0000 [r187360-187361] Joshua Colp <jcolp@digium.com>
* channels/chan_iax2.c: Do not try to send the format read/format
write/make compatible options over IAX2.
* main/channel.c, channels/chan_sip.c, include/asterisk/frame.h:
Add support for allowing the channel driver to handle
transcoding. This was accomplished using a set of options and the
setoption channel callback. The core calls into the channel
driver using these options and the channel driver either returns
success or failure.
2009-04-09 04:59 +0000 [r187302] Tilghman Lesher <tlesher@digium.com>
* agi/Makefile, build_tools/cflags.xml, utils/Makefile,
include/asterisk.h, /, main/Makefile, main/file.c, main/astfd.c
(added), main/asterisk.c: Merged revisions 187300-187301 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r187300 | tilghman | 2009-04-08 23:31:38 -0500 (Wed, 08 Apr 2009)
| 3 lines Add debugging mode for diagnosing file descriptor
leaks. (Related to issue #14625) ........ r187301 | tilghman |
2009-04-08 23:32:40 -0500 (Wed, 08 Apr 2009) | 2 lines Oops,
missed this file in the last commit. ........
2009-04-09 02:44 +0000 [r187269] Kevin P. Fleming <kpfleming@digium.com>
* include/asterisk/logger.h, main/logger.c, apps/app_verbose.c,
configs/logger.conf.sample: add a dedicated log channel for
modules to be able report security-related events, so that they
can be fed into external processes for analysis and possible
mitigation efforts (inspired by this evening's Toronto Asterisk
Users Group meeting and previous dicussions amongst various
community members)
2009-04-08 21:00 +0000 [r187211] Jeff Peeler <jpeeler@digium.com>
* main/channel.c, main/features.c, include/asterisk/channel.h: Add
timer for features so that backup bridge config can go away The
biggest change done here was elimination of the backup_config for
use with features. Previously, the bridging code upon detecting a
feature would set the start time of the bridge to the start time
of the feature. Then after the feature had either expired or
timed out the start time would be reset to the true bridge start
time from the backup_config. Now, the time differences are
calculated with respect to the newly added feature_start_time
timeval instead. There should be no behavior changes from the
previous functionality aside from the bridge timing being
unaffected by either valid or partial feature matches. Previously
the timing would be increased by the length of time configured
for featuredigittimeout, which was probably never noticed.
(closes issue #14503) Reported by: KNK Tested by: jpeeler Review:
http://reviewboard.digium.com/r/179/
2009-04-08 20:39 +0000 [r187210] Tilghman Lesher <tlesher@digium.com>
* /: Recorded merge of revisions 187209 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r187209 | tilghman | 2009-04-08 15:39:13 -0500 (Wed, 08 Apr 2009)
| 4 lines Backport resolution for file descriptor leak in 1.6.0
to 1.4. This fixes short reads in http manager sessions, such as
those done by the ast-gui branch. (Fixes AST-198) ........
2009-04-08 19:59 +0000 [r187179] Russell Bryant <russell@digium.com>
* include/asterisk/doxyref.h,
include/asterisk/doxygen/reviewboard.h (added): Add documentation
for reviewboard usage and guidelines.
2009-04-08 18:12 +0000 [r187108] Joshua Colp <jcolp@digium.com>
* main/rtp_engine.c: Fix a bug where we would native bridge when we
did not want to.
2009-04-08 17:51 +0000 [r187105] Russell Bryant <russell@digium.com>
* channels/chan_sip.c: Remove duplicate prototype for temp_peer().
2009-04-08 17:08 +0000 [r187050] Tilghman Lesher <tlesher@digium.com>
* funcs/func_odbc.c: If the first column is empty, output a
delimiter anyway. (closes issue #14848) Reported by: john8675309
Patches: 20090408__bug14848.diff.txt uploaded by tilghman
(license 14) Tested by: john8675309
2009-04-08 16:52 +0000 [r187046] Mark Michelson <mmichelson@digium.com>
* /, res/res_musiconhold.c: Merged revisions 187045 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r187045 | mmichelson | 2009-04-08 11:52:03 -0500 (Wed,
08 Apr 2009) | 10 lines Fix a small logical error when loading
moh classes. We were unconditionally incrementing the number of
mohclasses registered. However, we should actually only increment
if the call to moh_register was successful. While this probably
has never caused problems, I noticed it and decided to fix it
anyway. ........
2009-04-08 16:27 +0000 [r187036] Joshua Colp <jcolp@digium.com>
* res/res_rtp_asterisk.c, main/rtp_engine.c: Turn a warning message
into a debug message and do not treat two situations as errors
when they are not.
2009-04-08 15:27 +0000 [r186985] Mark Michelson <mmichelson@digium.com>
* main/channel.c, /: Merged revisions 186984 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r186984 | mmichelson | 2009-04-08 10:26:46 -0500 (Wed, 08 Apr
2009) | 24 lines Make a couple of changes with regards to a new
message printed in ast_read(). "ast_read() called with no
recorded file descriptor" is a new message added after a bug was
discovered. Unfortunately, it seems there are a bunch of places
that potentially make such calls to ast_read() and trigger this
error message to be displayed. This commit does two things to
help to make this message appear less. First, the message has
been downgraded to a debug level message if dev mode is not
enabled. The message means a lot more to developers than it does
to end users, and so developers should take an effort to be sure
to call ast_read only when a channel is ready to be read from.
However, since this doesn't actually cause an error in operation
and is not something a user can easily fix, we should not spam
their console with these messages. Second, the message has been
moved to after the check for any pending masquerades. ast_read()
being called with no recorded file descriptor should not
interfere with a masquerade taking place. This could be seen as a
simple way of resolving issue #14723. However, I still want to
try to clear out the existing ways of triggering this message,
since I feel that would be a better resolution for the issue.
........
2009-04-08 13:38 +0000 [r186928-186957] Russell Bryant <russell@digium.com>
* include/asterisk/doxygen/releases.h: Add some additional notes on
release numbering.
* Makefile, include/asterisk/doxygen/releases.h (added),
include/asterisk/doxyref.h, contrib/asterisk-ng-doxygen,
include/asterisk/doxygen (added),
include/asterisk/doxygen/commits.h (added),
include/asterisk/doxygen/licensing.h (added), main/asterisk.c:
Start splitting up miscellaneous doxygen documentation into
separate files. doxyref.h was created to hold miscellaneous
documentation that was not specific to a part of the code. This
file has grown quite a bit so I decided to start splitting parts
of it out into new files. Now, you can drop a new file into
include/asterisk/doxygen/ and it will be processed by doxygen.
* channels/chan_sip.c: Update some comments and resolve potential
memory corruption in chan_sip. While browsing chan_sip the other
day, I noticed this dangerous code in dialog_needdestroy(). This
function is an ao2_callback. It is absolutely _not_ okay to
unlock the container from within this function. It's also not
clear why it was useful. Given that it could cause memory
corruption, I have removed it. There was also a TODO comment left
describing a potential implementation of an improvement to the
needdestroy handling. I'm not convinced that what was described
is the best choice here, so I have briefly described the way that
this function is used today that could be improved.
2009-04-08 05:06 +0000 [r186899] Tilghman Lesher <tlesher@digium.com>
* channels/chan_sip.c: Add lastms to the require API call.
2009-04-08 00:09 +0000 [r186833-186842] Mark Michelson <mmichelson@digium.com>
* /, formats/format_wav.c, formats/format_wav_gsm.c: Merged
revisions 186841 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r186841 | mmichelson | 2009-04-07 19:09:04 -0500 (Tue, 07 Apr
2009) | 8 lines Fix a few typos of the word "frequency." (closes
issue #14842) Reported by: jvandal Patches: frequency-typo.diff
uploaded by jvandal (license 413) ........
* channels/chan_sip.c: Fix bad merge from fix for issue 13867.
(closes issue #14686) Reported by: davidw
* main/channel.c, /: Merged revisions 186832 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r186832 | mmichelson | 2009-04-07 18:49:49 -0500 (Tue, 07 Apr
2009) | 8 lines Set the AST_FEATURE_WARNING_ACTIVE flag when a
p2p bridge returns AST_BRIDGE_RETRY. Without this flag set,
warning sounds will not be properly played to either party of the
bridge. (closes issue #14845) Reported by: adomjan ........
2009-04-07 22:23 +0000 [r186799] Tilghman Lesher <tlesher@digium.com>
* /, apps/app_macro.c: Merged revisions 186775 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r186775 | tilghman | 2009-04-07 17:16:50 -0500 (Tue, 07 Apr 2009)
| 3 lines Fix Macro documentation to match current (and intended)
behavior. (See -dev mailing list) ........
2009-04-07 20:46 +0000 [r186720] Mark Michelson <mmichelson@digium.com>
* main/manager.c, /: Merged revisions 186719 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r186719 | mmichelson | 2009-04-07 15:43:49 -0500 (Tue, 07 Apr
2009) | 6 lines Ensure that \r\n is printed after the ActionID in
an OriginateResponse. (closes issue #14847) Reported by: kobaz
........
2009-04-06 23:11 +0000 [r186624-186687] Joshua Colp <jcolp@digium.com>
* res/res_rtp_asterisk.c: Fix a log message getting output when it
should not have been.
* channels/chan_sip.c: Fix problem when authenticating a non-RTP
dialog.
* channels/chan_sip.c, doc/tex/channelvariables.tex, CHANGES: Add
support for changing the outbound codec on a SIP call using a
dialplan variable. This adds a dialplan variable
(SIP_CODEC_OUTBOUND) which controls the codec offered for an
outgoing SIP call. This is much like the SIP_CODEC dialplan
variable and has the same restrictions. The codec set must be one
that is configured for the call. (closes issue #13243) Reported
by: samdell3 Patches: 13243.diff uploaded by file (license 11)
2009-04-06 16:06 +0000 [r186620] Mark Michelson <mmichelson@digium.com>
* funcs/func_connectedline.c (added), funcs/func_redirecting.c
(added): Silly svn. These files didn't get merged over in the
merge of the issue8824 branch.
2009-04-06 13:23 +0000 [r186563] Joshua Colp <jcolp@digium.com>
* main/rtp_engine.c: Pass the correct value to sizeof when copying
address information. (issue #14827) Reported by: pj Patches:
14827.diff uploaded by file (license 11) Tested by: pj
2009-04-04 00:13 +0000 [r186537] Richard Mudgett <rmudgett@digium.com>
* /: Remove merged branch properties accidentally merged to trunk.
2009-04-03 22:41 +0000 [r186525] Mark Michelson <mmichelson@digium.com>
* channels/chan_unistim.c, channels/misdn/isdn_lib_intern.h,
channels/chan_local.c, main/rtp_engine.c, /,
channels/misdn/isdn_msg_parser.c, channels/chan_iax2.c,
channels/misdn/isdn_lib.c, channels/misdn_config.c,
include/asterisk/callerid.h, main/channel.c, main/dial.c,
channels/misdn/isdn_lib.h, channels/chan_dahdi.c,
channels/chan_phone.c, channels/chan_skinny.c, main/features.c,
configs/sip.conf.sample, include/asterisk/frame.h,
include/asterisk/rtp_engine.h, channels/chan_mgcp.c,
apps/app_dial.c, res/res_rtp_asterisk.c, main/stun.c,
channels/chan_sip.c, channels/chan_agent.c,
configs/misdn.conf.sample, include/asterisk/channel.h, CHANGES,
apps/app_queue.c, channels/chan_misdn.c,
apps/app_directed_pickup.c, channels/misdn/chan_misdn_config.h,
channels/chan_h323.c, main/callerid.c, include/asterisk/stun.h:
This commit introduces COLP/CONP and Redirecting party
information into Asterisk. The channel drivers which have been
most heavily tested with these enhancements are chan_sip and
chan_misdn. Further work is being done to add Q.SIG support and
will be introduced in a later commit. chan_skinny has code added
to it here, but according to user pj, the support on chan_skinny
is not working as of now. This will be fixed in a later commit. A
special thanks goes out to bugtracker user gareth for getting the
ball rolling and providing the initial support for this work.
Without his initial work on this, this would not have been nearly
as painless as it was. This functionality has been tested by
Digium's product quality department, as well as a customer site
running thousands of calls every day. In addition, many many many
many bugtracker users have tested this, too. (closes issue #8824)
Reported by: gareth Review: http://reviewboard.digium.com/r/201
2009-04-03 20:20 +0000 [r186461] Kevin P. Fleming <kpfleming@digium.com>
* channels/chan_dahdi.c, /: Merged revisions 186458 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r186458 | kpfleming | 2009-04-03 15:19:20 -0500 (Fri, 03
Apr 2009) | 5 lines Fix a bug where DAHDI/Zaptel channels would
not properly switch formats when requested Don't offer
AST_FORMAT_SLINEAR on DAHDI/Zaptel channels... while it could
provide a slight performance benefit, the translation core in
Asterisk has some flaws when a channel driver offers multiple raw
formats. this fix is much simpler than fixing the translation
core to solve that issue (although that will be done later).
........
2009-04-03 19:59 +0000 [r186444-186447] Tilghman Lesher <tlesher@digium.com>
* /, apps/app_voicemail.c: Merged revisions 186445 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r186445 | tilghman | 2009-04-03 14:56:48 -0500 (Fri, 03
Apr 2009) | 2 lines Found a conflict in the last commit, due to
multiple targets ........
* /, configs/voicemail.conf.sample, apps/app_voicemail.c: Merged
revisions 186415 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r186415 | tilghman | 2009-04-03 14:06:58 -0500 (Fri, 03 Apr 2009)
| 7 lines Distinguish in a sent email between simple sends and
forwards. (closes issue #11678) Reported by: jamessan Patches:
20090330__bug11678.diff.txt uploaded by tilghman (license 14)
Tested by: tilghman, lmadsen ........
2009-04-03 16:47 +0000 [r186382] Joshua Colp <jcolp@digium.com>
* main/channel.c, channels/chan_sip.c, channels/chan_iax2.c,
include/asterisk/frame.h: Add better support for relaying success
or failure of the ast_transfer() API call. This API call now
waits for a special frame from the underlying channel driver to
indicate success or failure. This allows the return value to
truly convey whether the transfer worked or not. In the case of
the Transfer() dialplan application this means the value of the
TRANSFERSTATUS dialplan variable is actually true. (closes issue
#12713) Reported by: davidw Tested by: file
2009-04-03 16:29 +0000 [r186379] David Vossel <dvossel@digium.com>
* main/audiohook.c: audio_audiohook_write_list() did not correctly
update sample size after ast_translate.
audio_audiohook_write_list() did not take into account that the
sample size may change after translation depending on if the
original frame is is 8khz or 16khz. the sample size is now
updated after translating to reflect this possibility. This
caused the audio on the receiving end to sound terrible. Thanks
to jcolp and mmichelson for helping me work this out. (issue
AST-197)
2009-04-03 15:52 +0000 [r186321] Joshua Colp <jcolp@digium.com>
* include/asterisk/crypto.h, /: Merged revisions 186320 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r186320 | file | 2009-04-03 12:48:56 -0300 (Fri, 03 Apr 2009) | 5
lines Fix a problem with the crypto variable definitions not
actually being defined properly. (closes issue #14804) Reported
by: jvandal ........
2009-04-03 15:18 +0000 [r186297] Tilghman Lesher <tlesher@digium.com>
* main/stdtime/localtime.c: Compatibility fix for glibc 2.4 (Closes
issue #14820) Reported by: phsultan
2009-04-03 14:32 +0000 [r186286] Mark Michelson <mmichelson@digium.com>
* apps/app_voicemail.c: Fix the ability to retrieve voicemail
messages from IMAP. A recent change made interactive vm_states no
longer get added to the list of vm_states and instead get stored
in thread-local storage. In trunk and all the 1.6.X branches, the
problem is that when we search for messages in a voicemail box,
we would attempt to update the appropriate vm_state struct by
directly searching in the list of vm_states instead of using the
get_vm_state_by_imap_user function. This meant we could not find
the interactive vm_state that we wanted. (closes issue #14685)
Reported by: BlargMaN Patches: 14685.patch uploaded by mmichelson
(license 60) Tested by: BlargMaN, qualleyiv, mmichelson
2009-04-03 02:03 +0000 [r186230] Russell Bryant <russell@digium.com>
* /, cdr/cdr_radius.c: Merged revisions 186229 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r186229 | russell | 2009-04-02 20:57:44 -0500 (Thu, 02 Apr 2009)
| 21 lines Fix a memory leak in cdr_radius. I came across this
while doing some testing of my ast_channel_ao2 branch. After
running a test overnight that generated over 5 million calls,
Asterisk had taken up about 1 GB of my system memory. So, I
re-ran the test with MALLOC_DEBUG turned on. However, it showed
no leaks in Asterisk during the test, even though Asterisk was
still consuming it somehow. Instead, I turned to valgrind, which
when run with --leak-check=full, told me exactly where the leak
came from, which was from allocations inside the radiusclient-ng
library. This explains why MALLOC_DEBUG did not report it. After
a bit of analysis, I found that we were leaking a little bit of
memory every time a CDR record was passed to cdr_radius. I don't
actually have a radius server set up to receive CDR records.
However, I always have my development systems compile and install
all modules. In addition to making sure there are not build
errors across modules, always loading modules helps find bugs
like this, too, so it is strongly recommend for all developers.
........
2009-04-02 21:56 +0000 [r186175] Mark Michelson <mmichelson@digium.com>
* /, configs/features.conf.sample: Merged revisions 186174 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r186174 | mmichelson | 2009-04-02 16:55:34 -0500 (Thu, 02 Apr
2009) | 5 lines Fix instructions in one-step parking comment to
make more sense. Changed a capital K to a lowercase k. ........
2009-04-02 17:26 +0000 [r186101] Kevin P. Fleming <kpfleming@digium.com>
* channels/chan_dahdi.c, /: Merged revisions 186081 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r186081 | kpfleming | 2009-04-02 12:21:29 -0500 (Thu, 02
Apr 2009) | 3 lines ensure that the buffer passed to
DAHDI_SET_BUFINFO is fully initialized ........
2009-04-02 17:20 +0000 [r186078] Joshua Colp <jcolp@digium.com>
* res/res_rtp_asterisk.c (added), channels/chan_unistim.c,
apps/app_dial.c, main/stun.c (added), main/rtp_engine.c (added),
channels/chan_local.c, channels/chan_sip.c,
channels/chan_bridge.c, main/Makefile, channels/chan_agent.c,
include/asterisk/rtp.h (removed), UPGRADE.txt,
channels/chan_gtalk.c, include/asterisk/_private.h, main/rtp.c
(removed), main/loader.c, channels/chan_jingle.c,
channels/chan_skinny.c, channels/chan_h323.c,
configs/sip.conf.sample, include/asterisk/stun.h (added),
include/asterisk/rtp_engine.h (added), main/asterisk.c,
channels/chan_mgcp.c: Merge in the RTP engine API. This API
provides a generic way for multiple RTP stacks to be integrated
into Asterisk. Right now there is only one present,
res_rtp_asterisk, which is the existing Asterisk RTP stack.
Functionality wise this commit performs the same as previously.
API documentation can be viewed in the rtp_engine.h header file.
Review: http://reviewboard.digium.com/r/209/
2009-04-02 17:10 +0000 [r186021-186060] Tilghman Lesher <tlesher@digium.com>
* /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions
186059 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
................ r186059 | tilghman | 2009-04-02 12:09:13 -0500
(Thu, 02 Apr 2009) | 9 lines Merged revisions 186056 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.2
........ r186056 | tilghman | 2009-04-02 12:02:18 -0500 (Thu, 02
Apr 2009) | 2 lines Fix for AST-2009-003 ........
................
* main/strings.c: Missed a common case for needing to extend the
buffer. (closes issue #14716) Reported by: sum Patches:
20090402__bug14716.diff.txt uploaded by tilghman (license 14)
Tested by: sum
2009-04-02 13:51 +0000 [r185953] Kevin P. Fleming <kpfleming@digium.com>
* channels/chan_dahdi.c, /: Merged revisions 185952 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r185952 | kpfleming | 2009-04-02 08:43:43 -0500 (Thu, 02
Apr 2009) | 5 lines the DAHDI_GETCONF, DAHDI_SETCONF and
DAHDI_GET_PARAMS ioctls were recently corrected to show that they
do, in fact, read data from userspace as part of their work. due
to this fix, valgrind now reports a number of cases where
chan_dahdi passed an uninitialized (or partially) buffer to these
ioctls, which could lead to unexpected behavior. this patch
corrects chan_dahdi to ensure that buffers passed to these ioctls
are always fully initialized. ........
2009-04-01 20:13 +0000 [r185912] Tilghman Lesher <tlesher@digium.com>
* include/asterisk/res_odbc.h, include/asterisk.h, main/strings.c,
main/manager.c, main/tdd.c, include/asterisk/astobj2.h,
main/ast_expr2f.c, include/asterisk/pbx.h,
include/asterisk/strings.h, main/taskprocessor.c, res/res_odbc.c:
Merge changes from str_substitution that are unrelated to that
branch. Included is a small bugfix to an ast_str helper, but most
of these changes are simply doxygen fixes.
2009-04-01 19:03 +0000 [r185846] David Vossel <dvossel@digium.com>
* /, channels/chan_sip.c: Merged revisions 185845 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r185845 | dvossel | 2009-04-01 14:02:00 -0500 (Wed, 01 Apr 2009)
| 10 lines Fixes issue with dropped calles due to re-Invite glare
and re-Invites never executing after a 491 Acknowledgement for
491 responses were never being processed because it didn't match
our pending invite's seqno. Since the ACK was never processed,
the 491 frame would continue to be retransmitted until eventually
the call was dropped due to max retries. Now during a pending
invite, if we receive another invite, we send an 491 and hold on
to that glare invite's seqno in the "glareinvite" variable for
that sip_pvt struct. When ACK's are received, we first check to
see if it is in response to our pending invite, if not we check
to see if it is in response to a glare invite. In this case, it
is in response to the glare invite and must be dealt with or the
call is dropped. I've changed the wait time for resending the
re-Invite after receving a 491 response to comply with RFC 3261.
Before this patch the scheduled re-Invite would only change a
flag indicating that the re-Invite should be sent out, now it
actually sends it out as well. (closes issue #12013) Reported by:
alx Review: http://reviewboard.digium.com/r/213/ ........
2009-04-01 13:59 +0000 [r185777] Mark Michelson <mmichelson@digium.com>
* main/manager.c: Address Russell's comments regarding rev 185704.
Use ast_debug and ast_softhangup_nolock.
2009-04-01 13:48 +0000 [r185741-185772] Russell Bryant <russell@digium.com>
* main/channel.c, /: Merged revisions 185771 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r185771 | russell | 2009-04-01 08:47:30 -0500 (Wed, 01 Apr 2009)
| 6 lines Fix a case where DTMF could bypass audiohooks. This
change fixes a situation where an audiohook that wants DTMF would
not actually get it. This is in the code path where we end DTMF
digit length emulation while handling a NULL frame. ........
* include/asterisk/stringfields.h: Fix dev-mode build on my box.
2009-04-01 00:39 +0000 [r185704] Mark Michelson <mmichelson@digium.com>
* main/manager.c, CHANGES: Allow the AMI Hangup command to accept a
Cause header. (closes issue #14695) Reported by: mneuhauser
Patches: cause-for-hangup-manager-action.patch uploaded by
mneuhauser (license 425)
2009-03-31 22:35 +0000 [r185664] Kevin P. Fleming <kpfleming@digium.com>
* utils: ignore copied (generated) file
2009-03-31 22:12 +0000 [r185600-185604] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c: Fix trunk's compilation.
* /, apps/app_queue.c: Merged revisions 185599 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r185599 | mmichelson | 2009-03-31 17:00:01 -0500 (Tue, 31 Mar
2009) | 6 lines Fix crash that would occur if an empty member was
specified in queues.conf. (closes issue #14796) Reported by: pida
........
2009-03-31 21:29 +0000 [r185581] Kevin P. Fleming <kpfleming@digium.com>
* main/utils.c, include/asterisk/stringfields.h: Optimizations to
the stringfields API This patch provides a number of
optimizations to the stringfields API, focused around saving (not
wasting) memory whenever possible. Thanks to Mark Michelson for
inspiring this work and coming up with the first two
optimizations that are represented here: Changes: - Cleanup of
some code, fix incorrect doxygen comments - When a field is
emptied or replaced with a new allocation, decrease the amount of
'active' space in the pool it was held in; if that pool reaches
zero active space, and is not the current pool, then free it as
it is no longer in use - When allocating a pool, try to allocate
a size that will fit in a 'standard' malloc() allocation without
wasting space - When allocating space for a field, store the
amount of space in the two bytes immediately preceding the field;
this eliminates the need to call strlen() on the field when
overwriting it, and more importantly it 'remembers' the amount of
space the field has available, even if a shorter string has been
stored in it since it was allocated - Don't automatically double
the size of each successive pool allocated; it's wasteful
http://reviewboard.digium.com/r/165/
2009-03-31 19:46 +0000 [r185469] Mark Michelson <mmichelson@digium.com>
* /, apps/app_voicemail.c: Merged revisions 185468 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r185468 | mmichelson | 2009-03-31 14:45:30 -0500 (Tue,
31 Mar 2009) | 8 lines Fix Russian voicemail intro to say the
word "messages" properly. (closes issue #14736) Reported by:
chappell Patches: voicemail_no_messages.diff uploaded by chappell
(license 8) ........
2009-03-31 19:07 +0000 [r185432] Russell Bryant <russell@digium.com>
* channels/chan_iax2.c: Improve performance of the code handling
the frame queue in chan_iax2. In my tests that exercised full
frame handling in chan_iax2, the version with these changes took
30% to 40% of the CPU time compared to the same test of Asterisk
trunk before these modifications. While doing some profiling for
<http://reviewboard.digium.com/r/205/>, one function that caught
my eye was network_thread() in chan_iax2.c. After the things that
I was working on there, it was the next target for analysis and
optimization. I used oprofile's source annotation functionality
and found that the loop traversing the frame queue in
network_thread() was to blame for the excessive CPU cycle
consumption. The frame_queue in chan_iax2 previously held all
frames that either were pending transmission or had been
transmitted and are still pending acknowledgment. In
network_thread(), the previous code would go back through the
main for loop after reading a single incoming frame or after
being signaled because a frame had been queued up for initial
transmission. In each iteration of the loop, it traverses the
entire frame queue looking for frames that need to be
transmitted. On a busy server, this could easily be quite a few
entries. This patch is actually quite simple. The frame_queue has
become only a list of frames pending acknowledgment. Frames that
need to be transmitted are queued up to a dedicated transmit
thread via the taskprocessor API. As a result, the code in
network_thread() becomes much simpler, as its only job is to read
incoming frames. In addition to the previously described changes,
this patch includes some additional changes to the frame_queue.
Instead of one big frame_queue, now there is a list per call
number to further reduce wasted list traversals. The biggest
impact of this change is in socket_process(). For additional
details on testing and test results, see the review request.
Review: http://reviewboard.digium.com/r/212/
2009-03-31 16:46 +0000 [r185363] David Brooks <dbrooks@digium.com>
* /, channels/chan_gtalk.c: Merged revisions 185362 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r185362 | dbrooks | 2009-03-31 11:37:12 -0500 (Tue, 31
Mar 2009) | 35 lines Fix incorrect parsing in chan_gtalk when
xmpp contains extra whitespaces To drill into the xmpp to find
the capabilities between channels, chan_gtalk calls iks_child()
and iks_next(). iks_child() and iks_next() are functions in the
iksemel xml parsing library that traverse xml nodes. The bug here
is that both iks_child() and iks_next() will return the next
iks_struct node *regardless* of type. chan_gtalk expects the next
node to be of type IKS_TAG, which in most cases, it is, but in
this case (a call being made from the Empathy IM client), there
exists iks_struct nodes which are not IKS_TAG data (they are
extraneous whitespaces), and chan_gtalk doesn't handle that case,
so capabilities don't match, and a call cannot be made.
iks_first_tag() and iks_next_tag(), on the other hand, will not
return the very next iks_struct, but will check to see if the
next iks_struct is of type IKS_TAG. If it isn't, it will be
skipped, and the next struct of type IKS_TAG it finds will be
returned. This assures that chan_gtalk will find the iks_struct
it is looking for. This fix simply changes all calls to
iks_child() and iks_next() to become calls to iks_first_tag() and
iks_next_tag(), which resolves the capability matching. The
following is a payload listing from Empathy, which, due to the
extraneous whitespace, will not be parsed correctly by iksemel:
<iq from='dbrooksjab@235-22-24-10/Telepathy'
to='astjab@235-22-24-10/asterisk' type='set' id='542757715704'>
<session xmlns='http://www.google.com/session'
initiator='dbrooksjab@235-22-24-10/Telepathy' type='initiate'
id='1837267342'> <description
xmlns='http://www.google.com/session/phone'> <payload-type
clockrate='16000' name='speex' id='96'/> <payload-type
clockrate='8000' name='PCMA' id='8'/> <payload-type
clockrate='8000' name='PCMU' id='0'/> <payload-type
clockrate='90000' name='MPA' id='97'/> <payload-type
clockrate='16000' name='SIREN' id='98'/> <payload-type
clockrate='8000' name='telephone-event' id='99'/> </description>
</session> </iq> Review: http://reviewboard.digium.com/r/181/
........
2009-03-31 14:53 +0000 [r185261] Russell Bryant <russell@digium.com>
* apps/app_queue.c: Don't free() an astobj2 object. (closes issue
#14672) Reported by: makoto
2009-03-31 14:07 +0000 [r185197] Joshua Colp <jcolp@digium.com>
* /, main/audiohook.c: Merged revisions 185196 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r185196 | file | 2009-03-31 11:06:39 -0300 (Tue, 31 Mar 2009) | 8
lines Fix crash when moving audiohooks between channels. Handle
the scenario where we are called to move audiohooks between
channels and the source channel does not actually have any on it.
(closes issue #14734) Reported by: corruptor ........
2009-03-30 20:42 +0000 [r185122-185123] Richard Mudgett <rmudgett@digium.com>
* /, configs/misdn.conf.sample, channels/misdn_config.c: Merged
revisions 185121 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r185121 | rmudgett | 2009-03-30 15:40:11 -0500 (Mon, 30 Mar 2009)
| 1 line Update the channel allocation method documentation.
........
* /, channels/misdn/isdn_lib.c: Merged revisions 185120 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r185120 | rmudgett | 2009-03-30 15:38:11 -0500 (Mon, 30 Mar 2009)
| 19 lines Make chan_misdn BRI TE side normally defer channel
selection to the NT side. Channel allocation collisions are not
handled by chan_misdn very well. This patch simply avoids the
problem for BRI only. For PRI, allocation collisions are still
possible but less likely since there are simply more channels
available and each end could use a different allocation strategy.
misdn.conf options available: te_choose_channel - Use to force
the TE side to allocate channels. method - Specify the channel
allocation strategy. (closes issue #13488) Reported by:
Christian_Pinedo Patches: isdn_lib.patch.txt uploaded by crich
Tested by: crich, siepkes, festr ........
2009-03-30 16:26 +0000 [r185072] Mark Michelson <mmichelson@digium.com>
* /, apps/app_queue.c: Merged revisions 185031 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r185031 | mmichelson | 2009-03-30 11:17:35 -0500 (Mon, 30 Mar
2009) | 39 lines Fix queue weight behavior so that calls in
low-weight queues are not inappropriately blocked. (This is
copied and pasted from the review request I made for this patch)
Asterisk has some odd behavior when queue weights are used. The
current logic used when potentially calling a queue member is: If
the member we are going to call is part of another queue and
_that other queue has any callers in it_ and has a higher weight
than the queue we are calling from, then don't try to contact
that member. The issue here is what I have marked with
underscores. If the higher-weighted queue has any callers in it
at all, then the queue member will be unreachable from the
lower-weighted queue. This has the potential to be really really
bad if using a queue strategy, such as leastrecent or
fewestcalls, with the potential to call the same member
repeatedly. The fix proposed by garychen on issue 13220 is very
simple and, as far as I can see, works well for this situation.
With this set of changes, the logic used becomes: If the member
we are going to call is part of another queue, the other queue
has a higher weight than the queue we are calling from, and the
higher weight queue has at least as many callers as available
members, then do not try to contact the queue member. If the
higher weighted queue has fewer callers than available members,
then there is no reason to deny the call to this member since the
other queue can afford to spare a member. Since the fix involved
writing a generic function for determining the number of
available members in the queue, I also modified the is_our_turn
function to make use of the new num_available_members function to
determine if it is our turn to try calling a member. There is one
small behavior change. Before writing this patch, if you had
autofill disabled, then if you were the head caller in a queue,
you would automatically be told that it was your turn to try
calling a member. This did not take into account whether there
were actually any queue members available to take the call. Now
we actually make sure there is at least one member available to
take the call if autofill is disabled. (closes issue #13220)
Reported by: garychen Review:
http://reviewboard.digium.com/r/202/ ........
2009-03-30 14:37 +0000 [r184948] Joshua Colp <jcolp@digium.com>
* /, channels/chan_sip.c: Merged revisions 184947 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r184947 | file | 2009-03-30 11:35:47 -0300 (Mon, 30 Mar 2009) |
14 lines Improve our handling of T38 in the initial INVITE from a
device. We now answer with matching media streams to what is
requested. If an INVITE is received with both a T38 and RTP media
stream this means we answer with both. For any outgoing calls
created as a result of this inbound one no T38 is requested in
the initial INVITE. Instead if we start receiving udptl packets
we trigger a reinvite on the outbound side. (closes issue #12437)
Reported by: marsosa Tested by: pinga-fogo, okrief, file, afu
Review: http://reviewboard.digium.com/r/208/ ........
2009-03-30 13:55 +0000 [r184910] Russell Bryant <russell@digium.com>
* channels/h323/Makefile.in: Fix build error when chan_h323 is not
being built. (reported by cai1982 in #asterisk-dev)
2009-03-29 05:52 +0000 [r184838-184843] Russell Bryant <russell@digium.com>
* /, apps/app_followme.c: Merged revisions 184842 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r184842 | russell | 2009-03-29 00:51:55 -0500 (Sun, 29 Mar 2009)
| 5 lines Ensure targs variable is fully initialized. (closes
issue #14758) Reported by: tim_ringenbach ........
* channels/Makefile: Simplify chan_h323 build to not require a
second run of "make". (closes issue #14715) Reported by: jthurman
Patches: h323-makefile-1.6.2.0-beta1.patch uploaded by jthurman
(license 614) Tested by: tzafrir, russell
2009-03-27 20:08 +0000 [r184798-184801] Leif Madsen <lmadsen@digium.com>
* apps/app_ices.c: Fix a typo in app_ices. (closes issue #14765)
Reported by: timeshell Patches: app_ices.svn-1.6.0.diff uploaded
by timeshell (license 399)
* include/asterisk/doxyref.h: Update commit message guidelines in
re: to punctuation. The doxygen documentation has now been
updated to state explicitly that I want punctuation atthe end of
the first sentence in a commit message. :).
2009-03-27 19:10 +0000 [r184762] Kevin P. Fleming <kpfleming@digium.com>
* main/channel.c, bridges/bridge_softmix.c,
include/asterisk/timing.h, include/asterisk/channel.h,
channels/chan_iax2.c, main/timing.c: Improve timing interface to
remember which provider provided a timer The ability to
load/unload timing interfaces is nice, but it means that when a
timer is allocated, it may come from provider A, but later
provider B becomes the 'preferred' provider. If this happens, all
timer API calls on the timer that was provided by provider A will
actually be handed to provider B, which will say WTF and return
an error. This patch changes the timer API to include a pointer
to the provider of the timer handle so that future operations on
the timer will be forwarded to the proper provider. (closes issue
#14697) Reported by: moy Review:
http://reviewboard.digium.com/r/211/
2009-03-27 18:04 +0000 [r184693-184726] Russell Bryant <russell@digium.com>
* main/manager.c, apps/app_minivm.c: Use ast_random() instead of
rand() to ensure we use the best RNG available.
* include/asterisk/app.h, apps/app_dumpchan.c, main/app.c,
apps/app_queue.c, apps/app_voicemail.c, main/cli.c: Change
global_app_buf to ast_str_thread_global_buf.
2009-03-27 15:57 +0000 [r184639-184677] Joshua Colp <jcolp@digium.com>
* bridges/bridge_softmix.c: Fix a potential timer leak in
bridge_softmix. It is possible for a bridge to be created without
actually being used. In that scenario a timing file descriptor
would be opened and not closed. To fix this the timing file
descriptor is now closed in the destroy callback, not the thread
function.
* res/res_agi.c: Fix speech structure leak in the AGI speech
recognition integration. The AGI dialplan applications did not
destroy the speech structure automatically if it was not
destroyed by the running AGI script. They will now do this.
(issue LUMENVOX-15)
* bridges/bridge_softmix.c: Remove a cast that is not needed.
2009-03-27 14:00 +0000 [r184630] Russell Bryant <russell@digium.com>
* include/asterisk/utils.h, main/pbx.c, res/ais/evt.c,
main/event.c, pbx/pbx_dundi.c, main/asterisk.c: Change g_eid to
ast_eid_default.
2009-03-27 13:57 +0000 [r184566-184628] Joshua Colp <jcolp@digium.com>
* bridges/bridge_softmix.c: Fix a potential race condition when
creating a software based mixing bridge. It was possible for no
timer to become available between creating the bridge and
starting it. We now open a timer when creating it and keep it
open until the bridge is destroyed.
* /, channels/chan_sip.c: Merged revisions 184565 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r184565 | file | 2009-03-27 10:06:45 -0300 (Fri, 27 Mar 2009) | 9
lines Fix an issue where nat=yes would not always take effect for
the RTP session on outgoing calls. If calls were placed using an
IP address or hostname the global nat setting was copied over but
was not set on the RTP session itself. This caused the RTP stack
to not perform symmetric RTP actions. (closes issue #14546)
Reported by: acunningham ........
2009-03-27 02:20 +0000 [r184512-184531] Russell Bryant <russell@digium.com>
* include/asterisk/lock.h: Fix some issues with rwlock corruption
that caused deadlock like symptoms. When dvossel and I were doing
some load testing last week, we noticed that we could make
Asterisk trunk lock up instantly when we started generating a
bunch of calls. The backtraces of locked threads were bizarre,
and many were stuck on an _unlock_ of an rwlock. The changes are:
1) Fix a number of places where a backtrace would be loaded into
an invalid index of the backtrace array. It's an off by one
error, which ends up writing over the rwlock itself. 2) Ensure
that in the array of held locks, we NULL out an index once it is
not being used so that it's not confusing when analyzing its
contents. 3) Remove a bunch of logging referring to an rwlock
operating being done with "deep reentrancy". It is normal for
_many_ threads to hold a read lock on an rwlock.
* main/file.c: Don't act surprised if we get a -1 indication.
* main/heap.c, include/asterisk/heap.h: Pass more useful
information through to lock tracking when DEBUG_THREADS is on.
2009-03-26 22:18 +0000 [r184448] Kevin P. Fleming <kpfleming@digium.com>
* /, sounds/Makefile: Merged revisions 184447 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r184447 | kpfleming | 2009-03-26 17:17:32 -0500 (Thu, 26 Mar
2009) | 3 lines use new, improved 8kHz prompts ........
2009-03-26 21:09 +0000 [r184389] David Vossel <dvossel@digium.com>
* /, apps/app_test.c: Merged revisions 184388 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r184388 | dvossel | 2009-03-26 16:07:32 -0500 (Thu, 26 Mar 2009)
| 8 lines pri loop TestClient/TestServer fails: server SEND DTMF
8 app_test was failing when sending the last DTMF digit, 8,
because of the 100ms pause issued after DTMF is sent. During this
pause the other side would hang up causing the test to look like
it failed. Now the other side waits a second before hanging up.
(closes issue #12442) Reported by: tzafrir ........
2009-03-25 22:11 +0000 [r184339-184344] Russell Bryant <russell@digium.com>
* main/event.c: Remove unneeded AST_LIST_ENTRY() and comment on the
purpose of ast_event_ref.
* channels/chan_unistim.c, channels/chan_dahdi.c,
include/asterisk/devicestate.h, include/asterisk/event.h,
channels/chan_sip.c, apps/app_minivm.c, res/ais/evt.c,
main/devicestate.c, main/event.c, include/asterisk/_private.h,
include/asterisk/strings.h, channels/chan_iax2.c,
main/asterisk.c, channels/chan_mgcp.c, apps/app_voicemail.c:
Improve performance of the ast_event cache functionality. This
code comes from svn/asterisk/team/russell/event_performance/.
Here is a summary of the changes that have been made, in order of
both invasiveness and performance impact, from smallest to
largest. 1) Asterisk 1.6.1 introduces some additional logic to be
able to handle distributed device state. This functionality comes
at a cost. One relatively minor change in this patch is that the
extra processing required for distributed device state is now
completely bypassed if it's not needed. 2) One of the things that
I noticed when profiling this code was that a _lot_ of time was
spent doing string comparisons. I changed the way strings are
represented in an event to include a hash value at the front. So,
before doing a string comparison, we do an integer comparison on
the hash. 3) Finally, the code that handles the event cache has
been re-written. I tried to do this in a such a way that it had
minimal impact on the API. I did have to change one API call,
though - ast_event_queue_and_cache(). However, the way it works
now is nicer, IMO. Each type of event that can be cached (MWI,
device state) has its own hash table and rules for hashing and
comparing objects. This by far made the biggest impact on
performance. For additional details regarding this code and how
it was tested, please see the review request. (closes issue
#14738) Reported by: russell Review:
http://reviewboard.digium.com/r/205/
2009-03-25 19:22 +0000 [r184280] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: Fix issue with a T38 reinvite being sent
even if not configured to do so. If we receive a T38 request
negotiate control frame we should only attempt to do so if the
option is enabled on the dialog.
2009-03-25 14:38 +0000 [r184220] Eliel C. Sardanons <eliels@gmail.com>
* /, main/asterisk.c: Merged revisions 184188 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r184188 | eliel | 2009-03-25 10:12:54 -0400 (Wed, 25 Mar 2009) |
13 lines Avoid destroying the CLI line when moving the cursor
backward and trying to autocomplete. When moving the cursor
backward and pressing TAB to autocomplete, a NULL is put in the
line and we are loosing what we have already wrote after the
actual cursor position. (closes issue #14373) Reported by: eliel
Patches: asterisk.c.patch uploaded by eliel (license 64) Tested
by: lmadsen ........
2009-03-25 14:33 +0000 [r184147-184219] Russell Bryant <russell@digium.com>
* main/timing.c: Include poll-compat.h
* main/timing.c: Change poll() to ast_poll().
* utils/Makefile, include/asterisk/compat.h: Fix build issues on
Mac OSX. (closes issue #14714) Reported by: ygor
2009-03-24 22:40 +0000 [r184079] Mark Michelson <mmichelson@digium.com>
* /, apps/app_senddtmf.c: Merged revisions 184078 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r184078 | mmichelson | 2009-03-24 17:34:45 -0500 (Tue, 24 Mar
2009) | 9 lines Change NULL pointer check to be ast_strlen_zero.
The 'digit' variable is guaranteed to be non-NULL, so the if
statement could never evaluate true. Changing to ast_strlen_zero
makes the logic correct. This was found while reviewing
ast_channel_ao2 code review. ........
2009-03-24 22:00 +0000 [r184037-184043] Russell Bryant <russell@digium.com>
* main/channel.c: Put siren7 and siren14 in ast_best_codec() just
so they're in there somewhere.
* channels/chan_iax2.c: Exclude slin16, siren7, and siren14 from
bandwidth=low and =medium The default codec configuration for
chan_iax2 is bandwidth=low. I noticed slin16 being negotiated as
the codec in some test calls, but that no longer happens after
this change.
2009-03-24 20:01 +0000 [r183995] David Vossel <dvossel@digium.com>
* channels/chan_sip.c, configs/sip.conf.sample, CHANGES: SIP
preferred codec only feature Added an option to respond to a SIP
invite with only the single most preferred joint codec. This
limits the options of what codecs the other side can use. (closes
issue #12485) Reported by: bamby Review:
http://reviewboard.digium.com/r/206/
2009-03-24 15:26 +0000 [r183865-183914] Tilghman Lesher <tlesher@digium.com>
* /, configs/voicemail.conf.sample: Merged revisions 183913 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r183913 | tilghman | 2009-03-24 10:25:42 -0500 (Tue, 24 Mar 2009)
| 3 lines Additionally note that the operator option needs an 'o'
extension. (Related to issue #14731) ........
* main/http.c: Allow browsers to cache images and other static
content.
2009-03-23 22:35 +0000 [r183831] Richard Mudgett <rmudgett@digium.com>
* channels/chan_misdn.c, channels/misdn/Makefile,
channels/misdn/chan_misdn_config.h, channels/misdn/ie.c,
channels/misdn/isdn_msg_parser.c, channels/misdn/portinfo.c,
channels/misdn/isdn_lib.c, channels/misdn_config.c: Removed
trailing whitespace in chan_misdn files.
2009-03-23 18:58 +0000 [r183766] Mark Michelson <mmichelson@digium.com>
* /, res/res_monitor.c: Merged revisions 183700 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r183700 | mmichelson | 2009-03-23 12:59:28 -0500 (Mon, 23 Mar
2009) | 7 lines Fix a memory leak in res_monitor.c The only way
that this leak would occur is if Monitor were started using the
Manager interface and no File: header were given. Discovered
while reviewing the ast_channel_ao2 review request. ........
2009-03-23 18:06 +0000 [r183701] Leif Madsen <lmadsen@digium.com>
* channels/chan_dahdi.c: Fixes a documentation error introduced
during the CLI cleanup at AstriDevCon 2008. (closes issue #14655)
Reported by: ulogic Patches: chan_dahdi.patch uploaded by ulogic
(license 728) Tested by: lmadsen
2009-03-22 21:00 +0000 [r183652] Joshua Colp <jcolp@digium.com>
* main/bridging.c: Fix a minor logic flaw with the bridge generic
thread. We only want to move the channel pointers that are
actually present.
2009-03-20 17:00 +0000 [r183560] Russell Bryant <russell@digium.com>
* /, channels/chan_iax2.c: Merged revisions 183559 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r183559 | russell | 2009-03-20 11:53:25 -0500 (Fri, 20
Mar 2009) | 2 lines Fix a crash in IAX2 registration handling
found during load testing with dvossel. ........
2009-03-20 16:25 +0000 [r183553-183555] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Fix chan_sip so it builds.
* include/asterisk/rtp.h, main/rtp.c, main/asterisk.exports: Remove
symbols I just added to main/asterisk.exports and instead rename
the functions.
* main/asterisk.exports: Add some missing symbols to
main/asterisk.exports Hey! chan_sip.so loads now!
2009-03-20 12:12 +0000 [r183511] Eliel C. Sardanons <eliels@gmail.com>
* channels/chan_dahdi.c: Remove duplicate <description> inside the
xml documentation.
2009-03-19 20:30 +0000 [r183436] David Vossel <dvossel@digium.com>
* apps/app_dial.c, /, main/features.c, include/asterisk/features.h:
Merged revisions 183386 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r183386 | dvossel | 2009-03-19 14:40:07 -0500 (Thu, 19 Mar 2009)
| 6 lines Cleaning up a few things in detect disconnect patch
Initialized ast_call_feature in detect_disconnect to avoid
accessing uninitialized memory. Cleaned up /param tags in
features.h. No longer send dynamic features in
ast_feature_detect. issue #11583 ........
2009-03-19 19:22 +0000 [r183321-183345] Tilghman Lesher <tlesher@digium.com>
* /: Recorded merge of revisions 183342 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r183342 | tilghman | 2009-03-19 14:21:30 -0500 (Thu, 19 Mar 2009)
| 2 lines Reordering, to change prior to unlocking ........
* channels/chan_dahdi.c, /: Merged revisions 183319 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r183319 | tilghman | 2009-03-19 14:15:33 -0500 (Thu, 19
Mar 2009) | 8 lines Delay signalling progress until a PRI channel
really signals progress. (closes issue #13034) Reported by:
klaus3000 Patches: 20090316__bug13034.diff.txt uploaded by
tilghman (license 14) patch_trunk_183progress_klaus3000.txt
uploaded by klaus3000 (license 65) Tested by: klaus3000 ........
2009-03-19 18:34 +0000 [r183312] Jason Parker <jparker@digium.com>
* /, main/asterisk.exports: Merged revisions 183291 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r183291 | qwell | 2009-03-19 13:28:16 -0500 (Thu, 19 Mar
2009) | 1 line Export some more required symbols. ........
2009-03-19 18:10 +0000 [r183244] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c: Fix a memory leak associated with queues. For
every attempt that app_queue made to place an outbound call to a
queue member, we would allocate a queue_end_bridge structure.
When the bridge for the call had completed, we would free the
structure. Unfortunately not all call attempts actually end up
bridged to a member, so we need to be more selective of when to
allocate the structure. With this change, the allocation occurs
in an area where we can guarantee that the call will be bridged.
(closes issue #14680) Reported by: caspy Patches: 14680.patch
uploaded by mmichelson (license 60) Tested by: caspy
2009-03-19 18:00 +0000 [r183239-183242] Russell Bryant <russell@digium.com>
* /, configure, include/asterisk/autoconfig.h.in, configure.ac,
main/loader.c: Merged revisions 183241 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r183241 | russell | 2009-03-19 12:52:52 -0500 (Thu, 19 Mar 2009)
| 2 lines Remove the use of RTLD_NOLOAD, as it is not behaving
like expected. ........
* /, main/asterisk.exports: Merged revisions 183238 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r183238 | russell | 2009-03-19 12:41:39 -0500 (Thu, 19
Mar 2009) | 1 line Allow the AES API to work. ........
2009-03-19 17:00 +0000 [r183196] Tilghman Lesher <tlesher@digium.com>
* res/res_odbc.exports: 2 symbols defined when DEBUG_THREADS
2009-03-19 16:28 +0000 [r183172] David Vossel <dvossel@digium.com>
* apps/app_dial.c, /, main/features.c, include/asterisk/features.h:
Merged revisions 183126 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r183126 | dvossel | 2009-03-19 11:15:16 -0500 (Thu, 19 Mar 2009)
| 17 lines Allow disconnect feature before a call is bridged
feature.conf has a disconnect option. By default this option is
set to '*', but it could be anything. If a user wishes to
disconnect a call before the other side answers, only '*' will
work, regardless if the disconnect option is set to something
else. This is because features are unavailable until bridging
takes place. The default disconnect option, '*', was hardcoded in
app_dial, which doesn't make any sense from a user perspective
since they may expect it to be something different. This patch
allows features to be detected from outside of the bridge, but
not operated on. In this case, the disconnect feature can be
detected before briding and handled outside of features.c.
(closes issue #11583) Reported by: sobomax Patches:
patch-apps__app_dial.c uploaded by sobomax (license 359)
11583.latest-patch uploaded by murf (license 17)
detect_disconnect.diff uploaded by dvossel (license 671) Tested
by: sobomax, dvossel Review: http://reviewboard.digium.com/r/195/
........
2009-03-19 16:22 +0000 [r183124-183148] Russell Bryant <russell@digium.com>
* /, main/asterisk.exports: Merged revisions 183145 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r183145 | russell | 2009-03-19 11:21:56 -0500 (Thu, 19
Mar 2009) | 1 line Add missing semicolon in exports script.
........
* /, main/asterisk.exports: Merged revisions 183123 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r183123 | russell | 2009-03-19 11:13:18 -0500 (Thu, 19
Mar 2009) | 2 lines Allow the CallerID API to work again.
........
2009-03-19 16:07 +0000 [r183117] Mark Michelson <mmichelson@digium.com>
* /, channels/chan_sip.c: Merged revisions 183115 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r183115 | mmichelson | 2009-03-19 11:04:02 -0500 (Thu, 19 Mar
2009) | 14 lines Fix an issue where cancelled outgoing SIP calls
would erroneously report the device as "in use." A user was
having an issue where if an outgoing SIP call was canceled, the
SIP device would remain in use if we had not received any
response to the initial INVITE we sent out. The SIP device would
remain in use until the autocongestion timer was exhausted. I
tracked down the cause of this to be the section of code I am
removing here. I asked several people what the purpose of this
code was meant to be, but no one could give me any sort of answer
as to why this was here. The person who was having this issue has
been using this patch for several months and it has stopped the
problems they have had. AST-196 ........
2009-03-19 15:37 +0000 [r183057-183108] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: Improve our triggering of a T38 switchover
internally when triggered by a received reinvite. Previously we
reached across the channel bridge to get the other party's SIP
dialog structure in order to trigger an outgoing reinvite. This
is extremely dangerous to do and only works if bridged to another
SIP channel. This patch changes this to use the T38 control frame
method of requesting a switchover. This change also causes the
SIP channel driver to propogate back whether the switchover
worked or not instead of blindly accepting the incoming T38
reinvite. Review: http://reviewboard.digium.com/r/200/
* main/channel.c: Fix an issue where a T38 control frame would get
dropped. If two channels were bridged together using a generic
bridge the T38 control frame would get passed up instead of being
indicated on the other channel.
2009-03-18 21:28 +0000 [r183032] Kevin P. Fleming <kpfleming@digium.com>
* res/res_ael_share.exports (added): allow this module to export
everything for now
2009-03-18 21:18 +0000 [r183028] Jeff Peeler <jpeeler@digium.com>
* channels/h323/ast_h323.cxx: Add some code removed by mistake from
commit 182722 that works around a file descriptor leak in
versions of PWLib prior to 1.12.0.
2009-03-18 19:41 +0000 [r182960] Tilghman Lesher <tlesher@digium.com>
* main/asterisk.exports: Fixing a lost symbol in manager.c
2009-03-18 11:40 +0000 [r182848-182883] Kevin P. Fleming <kpfleming@digium.com>
* include/asterisk/callerid.h, channels/chan_dahdi.c, /,
main/callerid.c: Merged revisions 182882 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r182882 | kpfleming | 2009-03-18 06:31:41 -0500 (Wed, 18 Mar
2009) | 3 lines fix another symbol namespace issue (reported by
Andrew on asterisk-dev) ........
* res/res_phoneprov.c, res/res_config_ldap.c, res/res_curl.c,
res/res_config_sqlite.c, res/res_jabber.exports, res/res_odbc.c,
res/res_odbc.exports: a few more namespace updates...
res_ael_share still needs some work before this can be merged to
other release branches
2009-03-18 02:28 +0000 [r182847] Russell Bryant <russell@digium.com>
* apps/app_nbscat.c, /, main/Makefile,
include/asterisk/autoconfig.h.in, configure.ac, main/utils.c,
include/asterisk/io.h, include/asterisk/channel.h, main/poll.c,
main/io.c, main/channel.c, channels/chan_skinny.c, configure,
apps/app_mp3.c, res/res_agi.c, channels/chan_alsa.c,
include/asterisk/poll-compat.h, main/asterisk.c: Merged revisions
182810 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r182810 | russell | 2009-03-17 21:09:13 -0500 (Tue, 17 Mar 2009)
| 44 lines Fix cases where the internal poll() was not being used
when it needed to be. We have seen a number of problems caused by
poll() not working properly on Mac OSX. If you search around,
you'll find a number of references to using select() instead of
poll() to work around these issues. In Asterisk, we've had poll.c
which implements poll() using select() internally. However, we
were still getting reports of problems. vadim investigated a bit
and realized that at least on his system, even though we were
compiling in poll.o, the system poll() was still being used. So,
the primary purpose of this patch is to ensure that we're using
the internal poll() when we want it to be used. The changes are:
1) Remove logic for when internal poll should be used from the
Makefile. Instead, put it in the configure script. The logic in
the configure script is the same as it was in the Makefile.
Ideally, we would have a functionality test for the problem, but
that's not actually possible, since we would have to be able to
run an application on the _target_ system to test poll()
behavior. 2) Always include poll.o in the build, but it will be
empty if AST_POLL_COMPAT is not defined. 3) Change uses of poll()
throughout the source tree to ast_poll(). I feel that it is good
practice to give the API call a new name when we are changing its
behavior and not using the system version directly in all cases.
So, normally, ast_poll() is just redefined to poll(). On systems
where AST_POLL_COMPAT is defined, ast_poll() is redefined to
ast_internal_poll(). 4) Change poll() in main/poll.c to be
ast_internal_poll(). It's worth noting that any code that still
uses poll() directly will work fine (if they worked fine before).
So, for example, out of tree modules that are using poll() will
not stop working or anything. However, for modules to work
properly on Mac OSX, ast_poll() needs to be used. (closes issue
#13404) Reported by: agalbraith Tested by: russell, vadim
http://reviewboard.digium.com/r/198/ ........
2009-03-18 02:21 +0000 [r182826] Kevin P. Fleming <kpfleming@digium.com>
* res/res_config_pgsql.c, /, res/res_snmp.c, res/res_smdi.exports
(added), main/Makefile, include/asterisk/astobj2.h,
res/res_agi.exports (added), Makefile.rules, main/astobj2.c,
main/asterisk.exports (added), res/res_odbc.exports (added),
res/res_speech.exports (added), res/res_config_odbc.c,
res/res_features.exports (added), build_tools/strip_nonapi
(removed), res/res_adsi.exports (added), default.exports (added),
makeopts.in, res/res_jabber.exports (added),
res/res_monitor.exports (added): Merged revisions 182808 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r182808 | kpfleming | 2009-03-17 20:55:22 -0500 (Tue, 17 Mar
2009) | 5 lines Improve the build system to *properly* remove
unnecessary symbols from the runtime global namespace. Along the
way, change the prefixes on some internal-only API calls to use a
common prefix. With these changes, for a module to export symbols
into the global namespace, it must have *both* the
AST_MODFLAG_GLOBAL_SYMBOLS flag and a linker script that allows
the linker to leave the symbols exposed in the module's .so file
(see res_odbc.exports for an example). ........
2009-03-17 21:28 +0000 [r182762] Russell Bryant <russell@digium.com>
* funcs/func_channel.c, CHANGES: Add support for the "name" option
in the CHANNEL() function. Review:
http://reviewboard.digium.com/r/199/
2009-03-17 20:47 +0000 [r182722] Jeff Peeler <jpeeler@digium.com>
* channels/h323/compat_h323.cxx, channels/h323/ast_h323.cxx,
configure, autoconf/ast_check_openh323.m4,
channels/h323/compat_h323.h, channels/chan_h323.c,
channels/h323/ast_h323.h, channels/h323/chan_h323.h: Allow H.323
Plus library to be used in addition to the OpenH323 library
Chan_h323 can now be compiled against both the previously
supported versions of OpenH323 as well as the current H.323 Plus
(version 1.20.2). The configure script has been modified to look
in the default install location of h323 to hopefully help avoid
using the environment variables OPENH323DIR and PWLIBDIR. Also,
the CLI command "h323 show version" has been added which
indicates which version of h323 is in use. (closes issue #11261)
Reported by: vhatz Patches: asterisk-1.6.0.6-h323plus.patch
uploaded by jthurman (license 614)
2009-03-17 18:06 +0000 [r182596-182607] David Vossel <dvossel@digium.com>
* CHANGES: Fixing CHANGES in rev 182596. Progress DTMF was added
into app_dial's D() option. In CHANGES it should have been
updated under 1.6.3 rather than 1.6.2.
* apps/app_dial.c, CHANGES: Option to send DTMF when receiving
PROGRESS status The D() option in app_dial is only able to send
DTMF after the call has been answered. A progress option has been
added to D() to allow DTMF to be sent upon receiving PROGRESS.
This allows DTMF to be sent before the call is answered. (closes
issue #12123) Reported by: VoipForces Patches:
app_dial.c_patch_trunk_valid uploaded by VoipForces (license 419)
dtmf_progress.patch uploaded by dvossel (license 671) Tested by:
VoipForces, dvossel
2009-03-17 15:22 +0000 [r182553] Russell Bryant <russell@digium.com>
* main/channel.c: Tweak the handling of the frame list inside of
ast_answer(). This does not change any behavior, but moves the
frames from the local frame list back to the channel read queue
using an O(n) algorithm instead of O(n^2).
2009-03-17 14:59 +0000 [r182525-182530] Kevin P. Fleming <kpfleming@digium.com>
* main/channel.c: correct logic flaw in ast_answer() changes in
r182525
* main/channel.c, main/features.c, include/asterisk/channel.h:
Improve behavior of ast_answer() to not lose incoming frames
ast_answer(), when supplied a delay before returning to the
caller, use ast_safe_sleep() to implement the delay.
Unfortunately during this time any incoming frames are discarded,
which is problematic for T.38 re-INVITES and other sorts of
channel operations. When a delay is not passed to ast_answer(),
it still delays for up to 500 milliseconds, waiting for media to
arrive. Again, though, it discards any control frames, or
non-voice media frames. This patch rectifies this situation, by
storing all incoming frames during the delay period on a list,
and then requeuing them onto the channel before returning to the
caller. http://reviewboard.digium.com/r/196/
2009-03-17 14:24 +0000 [r182521] Sean Bright <sean@malleable.com>
* autoconf/ast_ext_lib.m4: Don't include a space before the
optional extra text that may follow a help string.
2009-03-17 05:51 +0000 [r182450] Tilghman Lesher <tlesher@digium.com>
* /, main/db.c: Merged revisions 182449 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r182449 | tilghman | 2009-03-17 00:50:52 -0500 (Tue, 17 Mar 2009)
| 7 lines Fix race in astdb The underlying db1 implementation
does not fully isolate the pages retrieved from astdb, so the
lock protecting accesses needs to be extended until the copy from
the shared memory structure is done. (closes issue #14682)
Reported by: makoto ........
2009-03-17 01:54 +0000 [r182408] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c: OPENR2 uses an incorrect string value if
the extension delimiter is not present. * Fixed OPENR2 using an
incorrect string value if the extension delimiter is not present
in the Dial() function. This was fixed for SS7 and PRI in trunk
-r172400. * Made OPENR2 stripmsd behavior the same as the SS7,
PRI, and others. * Removed trailing whitespace that appeared with
OPENR2.
2009-03-16 20:53 +0000 [r182362] Russell Bryant <russell@digium.com>
* UPGRADE.txt, CHANGES: Update UPGRADE.txt and CHANGES for 1.6.3
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