In app.c, ast_app_getdata is called to stream the prompts and receive DTMF input. If ast_app_getdata() receives an empty string caused by the user inputing the end of string character, in this case '#', it should break from the prompt loop and return to app_read, but instead it cycles through all the prompts. I've added a return value for this special case in ast_readstring() which uses an enum I've delcared in apps.h. This enum is now used as a return value for ast_app_getdata().
(closes issue #14279)
Reported by: Marquis
Patches:
fix_app_read.patch uploaded by Marquis (license 32)
read-ampersanmd.patch2 uploaded by dvossel (license 671)
Tested by: Marquis, dvossel
Review: http://reviewboard.digium.com/r/177/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@180032 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r180006 | mmichelson | 2009-03-03 16:48:18 -0600 (Tue, 03 Mar 2009) | 17 lines
Clarify some documentation of queues.conf.sample
It had always been possible to explicitly specify a "blank"
value for a sound file in queues.conf and have no sound played
back. The problem with this is that it would result in some ugly
CLI warnings from file.c.
This commit introduces a check when playing a file in app_queue
to see if the name of the file is zero-length and return early if
that is the case. Also, the ability to specify the blank sound
files in queues.conf is now mentioned more clearly in queues.conf.sample
(closes issue #14227)
Reported by: caspy
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@180007 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When app_meetme finds a realtime conference, it doesn't get the filename and fileformat correctly when 'r' is set. Now app_meetme first checks to see if fileformat and filename are declared in the db, if they're not it checks the .conf file, if its not declared there either it then uses defaults.
(closes issue #14545)
Reported by: dalbaech
Patches:
app_meetme-realtime5.patch uploaded by dvossel (license 671)
Realtime_Conference_Record_workaround.txt uploaded by dalbaech (license 705)
Tested by: dvossel, dalbaech
Review: http://reviewboard.digium.com/r/180/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@179972 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r179532 | russell | 2009-03-02 17:34:13 -0600 (Mon, 02 Mar 2009) | 40 lines
Move ast_waitfor() down to avoid the results of the API call becoming stale.
This call to ast_waitfor() was being done way too soon in this section of code.
Specifically, there was code in between the call to waitfor and the code that
uses the result that puts the channel in autoservice. By putting the channel
in autoservice, the previous results of ast_waitfor() become meaningless,
as the autoservice thread will do it's own ast_waitfor() and ast_read()
on the channel.
So, when we came back out of autoservice and eventually hit the block of code
that calls ast_read() on the channel, there may not actually be any input on
the channel available. Even though the previous call to ast_waitfor() in
app_meetme said there was input, the autoservice thread has since serviced
the channel for some period of time.
This bug manifested itself while dvossel was doing some testing of MeetMe in
Asterisk trunk. He was using the timerfd timing module. When the code hit
ast_read() erroneously, it determined that it must have been called because of
input on the timer fd, as chan->fdno was set to AST_TIMING_FD, since that was
the cause of the last legitimate call to ast_read() done by autoservice.
In this test, an IAX2 channel was calling into the MeetMe conference. It was
_much_ more likely to be seen with an IAX2 channel because of the way audio
is handled. Every audio frame that comes in results in a call to
ast_queue_frame(), which then uses ast_timer_enable_continuous() to notify
the channel thread that a frame is waiting to be handled. So, the chances
of ast_waitfor() indicating that a channel needs servicing due to a timer
event on an IAX2 event is very high.
Finally, it is interesting to note that if a different timing interface was
being used, this bug would probably not be noticed. When ast_read() is called
and erroneously thinks that there is a timer event to handle, it calls the
ast_timer_ack() function. The pthread and dahdi timing modules handle the
ack() function being called when there is no event by simply ignoring it.
In the case of the timerfd module, it results in a read() on the timer fd
that will block forever, as there is no data to read. This caused Asterisk
to lock up very quickly.
Thanks to dvossel and mmichelson for the fun debugging session. :-)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@179533 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Prior to spandsp-0.0.6pre4 the t30_stats_t structure used a pages_transferred
integer to indicate the number of pages transferred (so far) during the fax
session. The spandsp-0.0.6pre4 release removed the pages_transferred integer
and replaced it with two different integers - pages_tx and pages_rx. This
revision uses the new integers for spandsp-0.0.6pre4 while maintaining backwards
compatibility for previous spandsp releases.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@177699 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Enabling this option by default proved to be a bad idea, as the talker detection
is not very reliable. So, make it optional again, and off by default.
(issue #13801)
Reported by: justdave
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@177101 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch includes a number of changes to the indications API. The primary
motivation for this work was to improve stability. The object management
in this API was significantly flawed, and a number of trivial situations could
cause crashes.
The changes included are:
1) Remove the module res_indications. This included the critical functionality
that actually loaded the indications configuration. I have seen many people
have Asterisk problems because they accidentally did not have an
indications.conf present and loaded. Now, this code is in the core,
and Asterisk will fail to start without indications configuration.
There was one part of res_indications, the dialplan applications, which did
belong in a module, and have been moved to a new module, app_playtones.
2) Object management has been significantly changed. Tone zones are now
managed using astobj2, and it is no longer possible to crash Asterisk by
issuing a reload that destroys tone zones while they are in use.
3) The API documentation has been filled out.
4) The API has been updated to follow our naming conventions.
5) Various bits of code throughout the tree have been updated to account
for the API update.
6) Configuration parsing has been mostly re-written.
7) "Code cleanup"
The code is from svn/asterisk/team/russell/indications/.
Review: http://reviewboard.digium.com/r/149/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176627 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The problem here is that the hint processing code was subscribed to the wrong
event type. So, it started processing state for a hint too soon, before the
device state cache had been updated.
Also, fix a similar bug in app_queue, as it was also subscribed to the wrong
event type.
(closes issue #14461)
Reported by: alecdavis
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176557 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r176249 | mmichelson | 2009-02-16 15:34:27 -0600 (Mon, 16 Feb 2009) | 14 lines
Open the DAHDI pseudo device and set it to be nonblocking atomically
Apparently on FreeBSD, attempting to set the O_NONBLOCKING flag separately
from opening the file was causing an "inappropriate ioctl for device" error.
While I cannot fathom why this would be happening, I certainly am not opposed
to making the code a bit more compact/efficient if it also fixes a bug.
(closes issue #14482)
Reported by: ys
Patches:
meetme.patch uploaded by ys (license 281)
Tested by: ys
........
r176252 | mmichelson | 2009-02-16 15:39:21 -0600 (Mon, 16 Feb 2009) | 3 lines
Remove unused variable and make dev-mode compilation happy
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176253 65c4cc65-6c06-0410-ace0-fbb531ad65f3
From a user point-of-view, this adds new CLI commands and Manager Actions to
better facilitate the reloading of queues and the resetting of their statistics.
The new CLI commands are the "queue reload" and "queue reset stats" commands.
The new manager actions are the QueueReload and QueueReset commands.
Review: http://reviewboard.digium.com/r/115
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175663 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r175590 | mmichelson | 2009-02-13 13:47:48 -0600 (Fri, 13 Feb 2009) | 16 lines
Fix a potential crash situation when using IMAP voicemail
If calling into VoiceMailMain when using IMAP storage, it was
possible to crash Asterisk by hanging up the phone when prompted
for a voicemail mailbox. This patch fixes the issue.
While it may appear that this patch is superficial, it allows code
execution to continue to the failure case just below the IMAP_STORAGE
code block where this patch has been applied
(closes issue #14473)
Reported by: dwpaul
Patches:
voicemail_imap_crash_no_mailbox.patch uploaded by dwpaul (license 689)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175591 65c4cc65-6c06-0410-ace0-fbb531ad65f3
If someone has configured the queue to play an position or holdtime
announcement, then it is odd and potentially unexpected to hear a
"Thank you for your patience" sound when no position or holdtime
was actually announced.
This fixes the announcement so that the "thanks" sound is only played
in the case that a position or holdtime was actually announced.
There is a way that the "thank you" sound can be played without a
position or holdtime, and that is to set announce-frequency to a value
but keep announce-position and announce-holdtime both turned off.
(closes issue #14227)
Reported by: caspy
Patches:
14227_v3.patch uploaded by putnopvut (license 60)
Tested by: caspy
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174948 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The 'd' option would not work for channel types which use RTP to transport
DTMF digits. The only way to allow for this to work was to answer the channel
if we saw that this option was enabled.
I realized that this may cause issues with CDRs, specifically with giving false
dispositions and answer times. I therefore modified ast_answer to take another
parameter which would tell if the CDR should be marked answered.
I also extended this to the Answer application so that the channel may be answered
but not CDRified if desired.
I also modified app_dictate and app_waitforsilence to only answer the channel if it
is not already up, to help not allow for faulty CDR answer times.
All of these changes are going into Asterisk trunk. For 1.6.0 and 1.6.1, however, all
the changes except for the change to the Answer application will go in since we do
not introduce new features into stable branches
(closes issue #14164)
Reported by: DennisD
Patches:
14164.patch uploaded by putnopvut (license 60)
Tested by: putnopvut
Review: http://reviewboard.digium.com/r/145
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174945 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When using the 'g' or 'e' options, the stack allocations that
were used could cause a stack overflow if a spyer stayed on the
line long enough without actually successfully spying on anyone.
The problem has been corrected by using static buffers and copying
the contents of the appropriate strings into them instead of using
functions like alloca or ast_strdupa
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174805 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The ignore_hangup, run_dead, and noanswer flags were never initilized to zero causing hangups to never be issued. If the external script expects to be notified of a hangup and never receives one, it runs indefinitely.
(closes issue #14251)
Reported by: chris-mac
Tested by: dvossel
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174325 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r173696 | jpeeler | 2009-02-05 14:47:51 -0600 (Thu, 05 Feb 2009) | 12 lines
Add new configuration option to make shared IMAP mailboxes function as expected.
The new option is "imapvmshareid" which is an ID to tag multiple mailboxes
using the same IMAP storage location to function as one mailbox. This allows
all messages to be retrieved for any user in the group. The patch alters the
'X-Asterisk-VM-Extension' header that is responsible for matching voicemails
for a given user.
(closes issue #13673)
Reported by: howardwilkinson
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173697 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r173692 | mmichelson | 2009-02-05 14:29:09 -0600 (Thu, 05 Feb 2009) | 12 lines
Fix situations where queue members could be autopaused unexpectedly
Specifically, this patch prevents us from autopausing members when
we receive a busy or congestion frame from them.
(closes issue #14376)
Reported by: fiddur
Patches:
14376.patch uploaded by putnopvut (license 60)
Tested by: fiddur
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173693 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r173559 | mmichelson | 2009-02-05 11:34:33 -0600 (Thu, 05 Feb 2009) | 25 lines
Fix a problem where a channel pointer becomes invalid due to masquerading or hanging up.
app_mixmonitor runs its own thread to monitor the channel's activity and write the mixed
audio to a file. Since this thread runs independently of the channel, it is possible that
the mixmonitor thread's channel pointer will point to freed memory when the channel either
is masqueraded or hangs up (technically, both cases are hangups, but we need to handle the
cases slightly differently).
The solution for this is to employ a datastore, which has the nice benefit of allowing us
to hook into channel masquerades and hangups and update our pointer as necessary. If this
looks familiar, this same technique is employed in app_chanspy. app_chanspy is a bit more
involved since it does a lot more operations on the channel that is being spied upon.
app_mixmonitor does have an extra touch that app_chanspy doesn't have, though. Since there
is a thread race between the channel's thread and the mixmonitor thread on a hangup, we em-
ploy a condition-and-boolean combination to ensure that the channel thread finishes with
our structure before the mixmonitor thread attempts to free it. No crashes!
(closes issue #14374)
Reported by: aragon
Patches:
14374.patch uploaded by putnopvut (license 60)
Tested by: aragon, putnopvut
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173589 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2. Use curl to download sound files, as curl is installed natively on OS X,
whereas wget and fetch are not.
(closes issue #14332)
Reported by: oej
Tested by: Corydon76
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173130 65c4cc65-6c06-0410-ace0-fbb531ad65f3
to be used within the dial app, before a call is bridged.
Many thanks to sobomax for submitting this patch.
Quoting from bug 11582:
"So the goal of the patch was to use the user configured feature code during the
call setup phase. The original ast_feature_interpret() function is not well suited
for this purpose as it uses much call bridge specific data and doesn't separate a
detection of feature from a feature handler call. So a new function ast_feature_detect()
has been extracted off the ast_feature_interpret() function but keeping the original
logic intact except some insignificant changes to locking.
"Having created the ast_feature_detect() function the possibility to use feature detection
in almost any place of the asterisk code. So a call to this function has been added to
wait_for_answer() function of app_dial.so module. This code doesn't call the feature handler
however and uses old call leg disconnect logic to make the changes as small and simple as
possible to prevent unexpected problems. A disconnect feature currently is the only one
supported during call setup as other features as call parking and call transfer don't make much
sense during call setup. However if need in some of the features would arise it is much easier to
implement as the infrastructure changes are already in place with this patch."
I have cleaned up the patch somewhat, and verified that the existing functionality is not
harmed, and that the new functionality works. Terry has committed his stuff, and there were
no conflicts (see 14274).
(closes issue #11583)
Reported by: sobomax
Patches:
patch-apps__app_dial.c uploaded by sobomax (license 359)
patch-include__asterisk__features.h uploaded by sobomax (license 359)
patch-res__res_features.c uploaded by sobomax (license 359)
enable-features-during-call-setup.diff uploaded by sobomax (license 359)
11583.newdiff uploaded by murf (license 17)
enable-features-during-call-setup-1.diff uploaded by sobomax (license 359)
11583.latest-patch uploaded by murf (license 17)
Tested by: sobomax, murf
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172890 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r172517 | twilson | 2009-01-30 11:47:41 -0600 (Fri, 30 Jan 2009) | 37 lines
Fix feature inheritance with builtin features
When using builtin features like parking and transfers, the AST_FEATURE_* flags
would not be set correctly for all instances when either performing a builtin
attended transfer, or parking a call and getting the timeout callback. Also,
there was no way on a per-call basis to specify what features someone should
have on picking up a parked call (since that doesn't involve the Dial() command).
There was a global option for setting whether or not all users who pickup a
parked call should have AST_FEATURE_REDIRECT set, but nothing for DISCONNECT,
AUTOMON, or PARKCALL.
This patch:
1) adds the BRIDGE_FEATURES dialplan variable which can be set either in the
dialplan or with setvar in channels that support it. This variable can be set
to any combination of 't', 'k', 'w', and 'h' (case insensitive matching of the
equivalent dial options), to set what features should be activated on this
channel. The patch moves the setting of the features datastores into the
bridging code instead of app_dial to help facilitate this.
2) adds global options parkedcallparking, parkedcallhangup, and
parkedcallrecording to be similar to the parkedcalltransfers option for
globally setting features.
3) has builtin_atxfer call builtin_parkcall if being transfered to the parking
extension since tracking everything through multiple masquerades, etc. is
difficult and error-prone
4) attempts to fix all cases of return calls from parking and completed builtin
transfers not having the correct permissions
(closes issue #14274)
Reported by: aragon
Patches:
fix_feature_inheritence.diff.txt uploaded by otherwiseguy (license 396)
Tested by: aragon, otherwiseguy
Review http://reviewboard.digium.com/r/138/
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172580 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r172438 | tilghman | 2009-01-29 16:54:29 -0600 (Thu, 29 Jan 2009) | 9 lines
Lose the CAP_NET_ADMIN at every fork, instead of at startup. Otherwise, if
Asterisk runs as a non-root user and the administrator does a 'restart now',
Asterisk loses the ability to set QOS on packets.
(closes issue #14004)
Reported by: nemo
Patches:
20090105__bug14004.diff.txt uploaded by Corydon76 (license 14)
Tested by: Corydon76
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172441 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Also, implement a private cause code (as suggested by Tilghman). This works with
chan_sip, but doesn't propagate through chan_local.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172318 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r172030 | murf | 2009-01-28 11:51:16 -0700 (Wed, 28 Jan 2009) | 46 lines
This patch fixes h-exten running misbehavior in manager-redirected
situations.
What it does:
1. A new Flag value is defined in include/asterisk/channel.h,
AST_FLAG_BRIDGE_HANGUP_DONT, which used as a messenge to the
bridge hangup exten code not to run the h-exten there (nor
publish the bridge cdr there). It will done at the pbx-loop
level instead.
2. In the manager Redirect code, I set this flag on the channel
if the channel has a non-null pbx pointer. I did the same for the
second (chan2) channel, which gets run if name2 is set...
and the first succeeds.
3. I restored the ending of the cdr for the pbx loop h-exten
running code. Don't know why it was removed in the first place.
4. The first attempt at the fix for this bug was to place code
directly in the async_goto routine, which was called from a
large number of places, and could affect a large number of
cases, so I tested that fix against a fair number of transfer
scenarios, both with and without the patch. In the process,
I saw that putting the fix in async_goto seemed not to affect
any of the blind or attended scenarios, but still, I was
was highly concerned that some other scenarios I had not tested
might be negatively impacted, so I refined the patch to
its current scope, and jmls tested both. In the process, tho,
I saw that blind xfers in one situation, when the one-touch
blind-xfer feature is used by the peer, we got strange
h-exten behavior. So, I inserted code to swap CDRs and
to set the HANGUP_DONT field, to get uniform behavior.
5. I added code to the bridge to obey the HANGUP_DONT flag,
skipping both publishing the bridge CDR, and running
the h-exten; they will be done at the pbx-loop (higher)
level instead.
6. I removed all the debug logs from the patch before committing.
7. I moved the AUTOLOOP set/reset in the h-exten code in res_features
so it's only done if the h-exten is going to be run. A very
minor performance improvement, but technically correct.
(closes issue #14241)
Reported by: jmls
Patches:
14241_redirect_no_bridgeCDR_or_h_exten_via_transfer uploaded by murf (license 17)
Tested by: murf, jmls
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172063 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This is to avoid having the list and counter of missed calls being touched by queue calls. Add the C option to queue() and nothing
will be logged on phones that support the Reason: header on SIP cancel, like the SNOM phones.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@171924 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The data passed to the end_bridge_callback was assumed to be data which was
still stack'd. The problem was that with some call features, attended transfers
in particular, a new bridge thread is started once the feature completes, meaning
that when the end_bridge_callback is called, the end_bridge_callback_data was
invalid.
To fix this problem, there are two measures taken
1. Instead of pointing to stacked data, we now used heap-allocated data for
passing to the end_bridge_callback in app_queue
2. Since bridges can end multiple times on a single logical call, we wait until
the final bridge is broken to actually set any queue variables. This is accomplished
through reference-counting and the use of an end_bridge_callback_data_fixup function
in app_queue.c
(closes issue #14260)
Reported by: ccesario
Patches:
14260.patch uploaded by putnopvut (license 60)
Tested by: ccesario
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@171618 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r170979 | seanbright | 2009-01-25 08:33:20 -0500 (Sun, 25 Jan 2009) | 9 lines
Resolve a logic error that was causing Page() to crash when more than one
channel was specified.
(closes issue #14308)
Reported by: bluefox
Patches:
20090124__bug14308.diff.txt uploaded by seanbright (license 71)
Tested by: kc0bvu
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@170980 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r170147 | file | 2009-01-22 12:50:54 -0400 (Thu, 22 Jan 2009) | 4 lines
If we are unable to request a DAHDI pseudo channel and we are using the user introduction without review option make sure it gets unset so other code does not blindly assume a DAHDI pseudo channel exists.
(closes issue #14282)
Reported by: cheesegrits
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@170148 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The fix being applied is a bit different for trunk and the 1.6.X branches.
For trunk, we only wish to strip off the characters beyond the second slash
if the channel is a Local channel (i.e. we are removing the /n from the device
name). Other channel technologies with multiple slashes (e.g. DAHDI) need the
information after the second slash in order to get the proper device state
information.
In addition to this fix, the 1.6.X branches are receiving a much more important
fix as well. The problem in 1.6.X is that the member's device name was being directly
changed instead of having a copy changed. This meant that we would strip off the
second slash and trailing characters and then leave the member's device name like
that permanently thereafter.
(closes issue #14014)
Reported by: kebl0155
Patches:
14014_number2.patch uploaded by putnopvut (license 60)
Tested by: kebl0155
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@169611 65c4cc65-6c06-0410-ace0-fbb531ad65f3
conferences. We were using the 'user number' field to compare against the
maximum allowed users, which works assuming users with lower user numbers
didn't leave the conference.
(closes issue #14117)
Reported by: sergedevorop
Patches:
20090114__bug14117-2.diff.txt uploaded by seanbright (license 71)
Tested by: sergedevorop
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168705 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r168628 | mmichelson | 2009-01-14 18:11:01 -0600 (Wed, 14 Jan 2009) | 16 lines
Fix some crashes from bad datastore handling in app_queue.c
* The queue_transfer_fixup function was searching for and removing
the datastore from the incorrect channel, so this was fixed.
* Most datastore operations regarding the queue_transfer datastore
were being done without the channel locked, so proper channel locking
was added, too.
(closes issue #14086)
Reported by: ZX81
Patches:
14086v2.patch uploaded by putnopvut (license 60)
Tested by: ZX81, festr
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168629 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r168593 | twilson | 2009-01-13 19:27:18 -0600 (Tue, 13 Jan 2009) | 20 lines
Don't overflow when paging more than 128 extensions
The number of available slots for calls in app_page was hardcoded to 128.
Proper bounds checking was not in place to enforce this limit, so if more than
128 extensions were passed to the Page() app, Asterisk would crash. This patch
instead dynamically allocates memory for the ast_dial structures and removes
the (non-functional) arbitrary limit.
This issue would have special importance to anyone who is dynamically creating
the argument passed to the Page application and allowing more than 128
extensions to be added by an outside user via some external interface.
The patch posted by a_villacis was slightly modified for some coding guidelines
and other cleanups. Thanks, a_villacis!
(closes issue #14217)
Reported by: a_villacis
Patches:
20080912-asterisk-app_page-fix-buffer-overflow.patch uploaded by a (license 660)
Tested by: otherwiseguy
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168594 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The "No one is answering..." verbose message contained 3 numbers that were not
explained in any way to whoever was viewing the message. It is more helpful now
since the message explains what the numbers mean. Also, the message has been
downgraded to "DEBUG" level.
(closes issue #14172)
Reported by: caio1982
Patches:
queue_answering_debug.diff uploaded by caio1982 (license 22)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168579 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds the functionality to app_queue of calculating
the average amount of time that channels are bridged for a
queue. The algorithm used to calculate the average is the same
exponential average currently used to calculate the average holdtime.
See the CHANGES file to see the methods you may use to view this
information.
(closes issue #13960)
Reported by: coolmig
Patches:
app_queue.c.diff.trunk-r158840 uploaded by coolmig (license 621)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@167792 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When placing a call to a queue which ran a gosub on the member's
channel, Asterisk would crash every time, stemming from the fact
that the member's channel was being hung up unexpectedly when the
Gosub completed. The necessary change was pretty much copied and
pasted from app_dial's similar changes made last week.
I also took the opportunity to change a LOG_DEBUG message in
app_dial to use ast_debug. I am guessing this was due to a direct
merge from 1.4 that was not corrected to use trunk's preferred
syntax.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166861 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
In order to merge this 1.4 patch into trunk,
I had to resolve some conflicts and wait for
Russell to make some changes to res_agi.
I re-ran all the tests; 39 calls in all, and
made fairly careful notes and comparisons: I
don't want this to blow up some aspect of
asterisk; I completely removed the KEEPALIVE
from the pbx.h decls. The first 3 scenarios
involving feature park; feature xfer to 700;
hookflash park to Park() app call all behave
the same, don't appear to leave hung channels,
and no crashes.
........
r166093 | murf | 2008-12-19 15:30:32 -0700 (Fri, 19 Dec 2008) | 131 lines
This merges the masqpark branch into 1.4
These changes eliminate the need for (and use of)
the KEEPALIVE return code in res_features.c;
There are other places that use this result code
for similar purposes at a higher level, these appear
to be left alone in 1.4, but attacked in trunk.
The reason these changes are being made in 1.4, is
that parking ends a channel's life, in some situations,
and the code in the bridge (and some other places),
was not checking the result code properly, and dereferencing
the channel pointer, which could lead to memory corruption
and crashes.
Calling the masq_park function eliminates this danger
in higher levels.
A series of previous commits have replaced some parking calls
with masq_park, but this patch puts them ALL to rest,
(except one, purposely left alone because a masquerade
is done anyway), and gets rid of the code that tests
the KEEPALIVE result, and the NOHANGUP_PEER result codes.
While bug 13820 inspired this work, this patch does
not solve all the problems mentioned there.
I have tested this patch (again) to make sure I have
not introduced regressions.
Crashes that occurred when a parked party hung up
while the parking party was listening to the numbers
of the parking stall being assigned, is eliminated.
These are the cases where parking code may be activated:
1. Feature one touch (eg. *3)
2. Feature blind xfer to parking lot (eg ##700)
3. Run Park() app from dialplan (eg sip xfer to 700)
(eg. dahdi hookflash xfer to 700)
4. Run Park via manager.
The interesting testing cases for parking are:
I. A calls B, A parks B
a. B hangs up while A is getting the numbers announced.
b. B hangs up after A gets the announcement, but
before the parking time expires
c. B waits, time expires, A is redialed,
A answers, B and A are connected, after
which, B hangs up.
d. C picks up B while still in parking lot.
II. A calls B, B parks A
a. A hangs up while B is getting the numbers announced.
b. A hangs up after B gets the announcement, but
before the parking time expires
c. A waits, time expires, B is redialed,
B answers, A and B are connected, after
which, A hangs up.
d. C picks up A while still in parking lot.
Testing this throroughly involves acting all the permutations
of I and II, in situations 1,2,3, and 4.
Since I added a few more changes (ALL references to KEEPALIVE in the bridge
code eliimated (I missed one earlier), I retested
most of the above cases, and no crashes.
H-extension weirdness.
Current h-extension execution is not completely
correct for several of the cases.
For the case where A calls B, and A parks B, the
'h' exten is run on A's channel as soon as the park
is accomplished. This is expected behavior.
But when A calls B, and B parks A, this will be
current behavior:
After B parks A, B is hung up by the system, and
the 'h' (hangup) exten gets run, but the channel
mentioned will be a derivative of A's...
Thus, if A is DAHDI/1, and B is DAHDI/2,
the h-extension will be run on channel
Parked/DAHDI/1-1<ZOMBIE>, and the
start/answer/end info will be those
relating to Channel A.
And, in the case where A is reconnected to
B after the park time expires, when both parties
hang up after the joyful reunion, no h-exten
will be run at all.
In the case where C picks up A from the
parking lot, when either A or C hang up,
the h-exten will be run for the C channel.
CDR's are a separate issue, and not addressed
here.
As to WHY this strange behavior occurs,
the answer lies in the procedure followed
to accomplish handing over the channel
to the parking manager thread. This procedure
is called masquerading. In the process,
a duplicate copy of the channel is created,
and most of the active data is given to the
new copy. The original channel gets its name
changed to XXX<ZOMBIE> and keeps the PBX
information for the sake of the original
thread (preserving its role as a call
originator, if it had this role to begin
with), while the new channel is without
this info and becomes a call target (a
"peer").
In this case, the parking lot manager
thread is handed the new (masqueraded)
channel. It will not run an h-exten
on the channel if it hangs up while
in the parking lot. The h exten will
be run on the original channel instead,
in the original thread, after the bridge
completes.
See bug 13820 for our intentions as
to how to clean up the h exten behavior.
Review: http://reviewboard.digium.com/r/29/
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166665 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r165889 | russell | 2008-12-19 09:03:02 -0600 (Fri, 19 Dec 2008) | 9 lines
Ensure that the chanspy datastore is fully initialized.
This patch resolved some random crash issues observed by a user on a BSD system
(closes issue #14111)
Reported by: ys
Patches:
app_chanspy.c.diff uploaded by ys (license 281)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@165890 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This is part of an effort to completely remove AST_PBX_KEEPALIVE and other
similar return codes from the source. While this usage was perfectly safe,
there are others that are problematic. Since we know ahead of time that
we do not want to PBX to destroy the channel, the PBX API has been changed
so that information can be provided as an argument, instead, thus removing
the need for the KEEPALIVE return value.
Further changes to get rid of KEEPALIVE and related code is being done by
murf. There is a patch up for that on review 29.
Review: http://reviewboard.digium.com/r/98/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@165723 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Jared Smith suggested that we add a way to be able to set variables
and functions on the outbound channel. I think that it's a great idea, so I
have added it as a todo so that it gets done at some point.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@165469 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r165255 | mmichelson | 2008-12-17 14:51:38 -0600 (Wed, 17 Dec 2008) | 7 lines
Fix some memory leaks found while looking at how realtime
configs are handled.
Also cleaned up some coding guidelines violations in app_realtime.c,
mostly related to spacing
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@165318 65c4cc65-6c06-0410-ace0-fbb531ad65f3
it was not being used before. Also, I've moved the urgent
folder check in messagecount() up a bit so that the flow is
a bit better.
This was something I noticed while taking a look at issue
#13973, although I don't think this is the underlying cause
of the issue.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@165142 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r164876 | russell | 2008-12-16 15:10:44 -0600 (Tue, 16 Dec 2008) | 6 lines
Do not dereference the channel if AST_PBX_KEEPALIVE has been returned.
This is a bug I noticed while looking at the code for app_macro. This return code
means that another thread has assumed ownership of the channel and it can no longer
be touched. (I hate this return code with a passion, by the way.)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164877 65c4cc65-6c06-0410-ace0-fbb531ad65f3
the 'i' options for app_dial and app_queue, in that they will ignore
any attempts by phones to forward the call.
(closes issue #13977)
Reported by: putnopvut
Patches:
page_ignore_forwards.patch uploaded by putnopvut (license 60)
Tested by: putnopvut, acunningham
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164428 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Fix reference counting used in the __queues_show function
* Add code to be sure that the "queue show" command does not
print information for a realtime queue which has been deleted
from the backend
* Add a missing unref to the realtime queue loading function for
the case where a queue is in the module's container but has been
deleted from the realtime backend
(closes issue #14033)
Reported by: cristiandimache
Patches:
14033.patch uploaded by putnopvut (license 60)
Tested by: cristiandimache
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164268 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r163080 | mmichelson | 2008-12-11 10:24:43 -0600 (Thu, 11 Dec 2008) | 14 lines
Fix a potential crash due to unsafe datastore handling.
This patch also contains a conversion from using long to time_t
for representing times for a queue, as well as some whitespace
fixes.
(closes issue #14060)
Reported by: nivek
Patches:
datastore_fixup.patch.corrected uploaded by nivek (license 636)
with slight modification from me
Tested by: nivek
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@163081 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r162286 | russell | 2008-12-09 14:57:35 -0600 (Tue, 09 Dec 2008) | 9 lines
Fix an issue where callers on an incoming call on an SLA trunk would not hear ringback.
We need to make sure that we don't start writing audio to the trunk channel until we're
actually ready to answer it. Otherwise, the channel driver will treat it as inband
progress, even though all they are getting is silence.
(closes issue #12471)
Reported by: mthomasslo
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@162291 65c4cc65-6c06-0410-ace0-fbb531ad65f3
add 1 to the priority when checking if the extension exists. Otherwise,
gosubs will fail.
This was discovered when investigating an asterisk-users mailing list post
made by Gary Hawkins.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@161493 65c4cc65-6c06-0410-ace0-fbb531ad65f3
emailbody and pagerbody to CR-LF so that the IMAP server doesn't spit out an
error. This was informally reported on #asterisk-dev a few weeks ago. Reviewed
by Mark M. on IRC.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@161349 65c4cc65-6c06-0410-ace0-fbb531ad65f3
for app_dial and app_queue to run a gosub when the call is answered.
* Check for the existence of the gosub target in gosub_exec. If it is nonexistent,
then this will cause errors when we attempt to actually run the gosub, including
a definite memory leak and potential crashes. Return an error in this situation
* Check the return value of pbx_exec in app_dial and app_queue before attempting
to actually run the gosub routine. If there was an error, we should not attempt
to run the gosub.
* Change a '|' to a ',' in app_queue.
* Add some extra curly braces where they had been missing previously.
(closes issue #13548)
Reported by: fiddur
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@160626 65c4cc65-6c06-0410-ace0-fbb531ad65f3
handling in app_queue was just completely wrong, mostly
because the channel operations being performed were being
done on the incorrect channel.
With this set of changes, a gosub will correctly run on
the answering queue member's channel. There are still crash
issues which occur if there are dialplan syntax errors, so
I cannot yet close the referenced issue.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@160555 65c4cc65-6c06-0410-ace0-fbb531ad65f3
a call fails
1) Hang up the original destination if the local channel cannot
be requested.
2) Hang up the local channel (in addition to the original destination)
if ast_call fails when calling the newly created local channel.
This prevents channels from sticking around forever in the
case of a botched call forward (e.g. to an extension which does not
exist).
(closes issue #13764)
Reported by: davidw
Patches:
13764_v2.patch uploaded by putnopvut (license 60)
Tested by: putnopvut, davidw
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@159554 65c4cc65-6c06-0410-ace0-fbb531ad65f3
for the "pls_hold_prompt" option. This does not affect any released
version of Asterisk, so there is no need to update the CHANGES
file for this.
(closes issue #13893)
Reported by: eliel
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@159250 65c4cc65-6c06-0410-ace0-fbb531ad65f3
it would be best to maintain API compatibility. Instead, this commit introduces
ao2_callback_data() which is functionally identical to ao2_callback() except
that it allows you to pass arbitrary data to the callback.
Reviewed by Mark Michelson via ReviewBoard:
http://reviewboard.digium.com/r/64
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@158959 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r158053 | mmichelson | 2008-11-20 11:33:06 -0600 (Thu, 20 Nov 2008) | 12 lines
Make sure to set the hangup cause on the calling channel in the case
that ast_call() fails. For incoming SIP channels, this was causing
us to send a 603 instead of a 486 when the call-limit was reached on
the destination channel.
(closes issue #13867)
Reported by: still_nsk
Patches:
13867.diff uploaded by putnopvut (license 60)
Tested by: blitzrage
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@158066 65c4cc65-6c06-0410-ace0-fbb531ad65f3
also, move UPGRADE.txt to UPGRADE-1.6.txt and make the new UPGRADE.txt contain information about upgrading between Asterisk 1.6 releases
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@157706 65c4cc65-6c06-0410-ace0-fbb531ad65f3
is in use so that the message is deleted from both
local and IMAP storage.
(closes issue #13642)
Reported by: jaroth
Patches:
deleteyes.patch uploaded by jaroth (license 50)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@157562 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r157365 | jpeeler | 2008-11-18 13:13:33 -0600 (Tue, 18 Nov 2008) | 6 lines
(closes issue #13899)
Reported by: akkornel
This fix is the result of a bug fix in ast_app_separate_args r124395. If an argument does not exist it should always be set to a null string rather than a null pointer.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@157366 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r157305 | mmichelson | 2008-11-18 12:25:55 -0600 (Tue, 18 Nov 2008) | 12 lines
Fix a crash in the end_bridge_callback of app_dial and
app_followme which would occur at the end of an attended
transfer. The error occurred because we initially stored
a pointer to an ast_channel which then was hung up due
to a masquerade.
This commit adds a "fixup" callback to the bridge_config
structure to allow for end_bridge_callback_data to be
changed in the case that a new channel pointer is needed
for the end_bridge_callback.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@157306 65c4cc65-6c06-0410-ace0-fbb531ad65f3
a function: in these cases, we have to use the heap, or garbage will result.
(closes issue #13898)
Reported by: alecdavis
Patches:
20081114__bug13898__2.diff.txt uploaded by Corydon76 (license 14)
Tested by: alecdavis
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@157253 65c4cc65-6c06-0410-ace0-fbb531ad65f3
hashing used by app_queue.c to be case-insensitive.
This is accomplished by adding a new case-insensitive
hashing function.
This was necessary to prevent bad refcount errors
(and potential crashes) which would occur due to the
fact that queues were initially read from the config
file in a case-sensitive manner. Then, when a user
issued a CLI command or manager action, we allowed
for case-insensitive input and used that input to
directly try to find the queue in the hash table. The result
was either that we could not find a queue that was input or
worse, we would end up hashing to a completely bogus value
based on the input.
This commit resolves the problem presented in
issue #13703. However, that issue was reported against
1.6.0. Since this fix introduces a behavior change, I am
electing to not place this same fix in to the 1.6.0 or 1.6.1
branches, and instead will opt for a change which does not
change behavior.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@156883 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r156816 | mmichelson | 2008-11-14 09:18:59 -0600 (Fri, 14 Nov 2008) | 10 lines
If the prompt to reenter a voicemail password timed out, it
resulted in the password not being saved, even if the input matched
what you gave when first prompted to enter a new password. This is
because the return value of ast_readstring was checked, but not checked
properly.
This bug was discovered by Jared Smith during an Asterisk training course.
Thanks for reporting it!
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@156817 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r156755 | tilghman | 2008-11-13 18:41:37 -0600 (Thu, 13 Nov 2008) | 6 lines
ast_waitfordigit() requires that the channel be up, for no good logical
reason. This prevents While/EndWhile from working within the "h"
extension.
Reported by: jgalarneau (for ABE C.2)
Fixed by: me
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@156756 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- MeetMe()
- MeetMeCount()
- MeetMeChannelAdmin()
- MeetMeAdmin()
- SLAStation()
- SLATrunk()
- Add an attribute to optionlist 'hasparams' with the same functionality as the one
we have in <parameter> and <argument> (the DTD was updated)
- Fix a leak when getting an attribute while parsing an <optionlist>.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@156575 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r156178 | jpeeler | 2008-11-12 11:53:44 -0600 (Wed, 12 Nov 2008) | 8 lines
(closes issue #13173)
Reported by: pep
This change adds an announce_thread responsible for playing announcements to an existing conference. This allows all announcing to be immediately stopped if necessary but more importantly allows other threads that need to play something to not block. There are multiple benefits to this, but the actual bug is for solving the scenario for a channel to be unusable after hang up for the entire duration of the parting announcement. The parting announcement can be extremely long depending on what the user recorded upon joining the conference.
Reviewed by Russell on Review Board:
http://reviewboard.digium.com/r/25/
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@156228 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r156167 | mmichelson | 2008-11-12 11:38:33 -0600 (Wed, 12 Nov 2008) | 7 lines
When doing some tests, I was having a crash at the end of every call
if an attended transfer occurred during the call. I traced the cause to
the CDR on one of the channels being NULL. murf suggested a check in
the end bridge callback to be sure the CDR is non-NULL before proceeding,
so that's what I'm adding.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@156169 65c4cc65-6c06-0410-ace0-fbb531ad65f3
ast_channel_search_locked need to be made. Specifically, the caller needs to be able to
pass arbitrary data which in turn is passed to the callback. This patch addresses all
of the nested functions currently in asterisk trunk.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@155590 65c4cc65-6c06-0410-ace0-fbb531ad65f3
ao2_callback and ao2_find). Currently, passing OBJ_POINTER to either
of these mandates that the passed 'arg' is a hashable object, making
searching for an ao2 object based on outside criteria difficult.
Reviewed by Russell and Mark M. via ReviewBoard:
http://reviewboard.digium.com/r/36/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@155401 65c4cc65-6c06-0410-ace0-fbb531ad65f3
along the way, change tags used in configure script, menuselect-deps and code for various dependencies to be consistently named
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@154151 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit introduces the first phase of an effort to manage documentation of the
interfaces in Asterisk in an XML format. Currently, a new format is available for
applications and dialplan functions. A good number of conversions to the new format
are also included.
For more information, see the following message to asterisk-dev:
http://lists.digium.com/pipermail/asterisk-dev/2008-October/034968.html
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@153365 65c4cc65-6c06-0410-ace0-fbb531ad65f3
had no effect
* Updated dialing API documentation to indicate that timeouts
are specified in milliseconds
* Added a new timeout argument to the Page application. If time
expires, any endpoints which have not answered will be hung up.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@153223 65c4cc65-6c06-0410-ace0-fbb531ad65f3
entry, then app_directory would crash when attempting to
read that entry from the file. We now check for the NULL
or empty string properly so that there will be no crash.
(closes issue #13804)
Reported by: bluecrow76
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@152646 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r152538 | murf | 2008-10-28 23:19:04 -0600 (Tue, 28 Oct 2008) | 14 lines
A little documentation cross-ref between features and
dial and queue... I wasted some time (stupidly) trying
to get the one-touch parking stuff working, because it
didn't occur to me that I had to also have the corresponding
options in the dial command! Duh! (In all this time, I never
set this up before!)
So, to keep some poor fool from suffering the same fate,
I made the features.conf.sample file mention the corresponding
opts in dial/queue; and the docs for dial/app specifically
mention the corresponding decls in the feature.conf file.
I hope this doesn't spoil some vast, eternal plan...
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@152605 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r152535 | murf | 2008-10-28 22:36:32 -0600 (Tue, 28 Oct 2008) | 46 lines
The magic trick to avoid this crash is not to
try to find the channel by name in the list,
which is slow and resource consuming, but rather
to pay attention to the result codes from the
ast_bridge_call, to which I added the
AST_PBX_NO_HANGUP_PEER_PARKED value, which
now are returned when a channel is parked.
Why? because CDR's aren't generated via parking,
so nothing is needed, but if a transfer occurred,
there are critical things I need.
If you get AST_PBX_KEEPALIVE,
then don't touch the channel pointer.
If you get AST_PBX_NO_HANGUP_PEER, or
AST_PBX_NO_HANGUP_PEER_PARKED, then don't
touch the peer pointer.
Updated the several places where the results
from a bridge were not being properly obeyed,
and fixed some code I had introduced so that
the results of the bridge were not overridden
(in trunk).
All the places that previously tested for
AST_PBX_NO_HANGUP_PEER now have to check for
both AST_PBX_NO_HANGUP_PEER and AST_PBX_NO_HANGUP_PEER_PARKED.
I tested this against the 4 common parking
scenarios:
1. A calls B; B answers; A parks B; B hangs up while A is getting the parking
slot announcement, immediately after being put on hold.
2. A calls B; B answers; A parks B; B hangs up after A has been hung up, but
before the park times out.
3. A calls B; B answers; B parks A; A hangs up while B is getting the parking slot announcement, immediately after being put on hold.
4. A calls B; B answers; B parks A; A hangs up after B has been hung up, but before the park times out.
No crash.
I also ran the scenarios above against valgrind, and accesses looked good.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@152536 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r152368 | tilghman | 2008-10-28 12:04:56 -0500 (Tue, 28 Oct 2008) | 8 lines
Reset all DIAL variables back to blank, in case Dial is called multiple times
per call (which could otherwise lead to inconsistent status reports).
(closes issue #13216)
Reported by: ruddy
Patches:
20081014__bug13216.diff.txt uploaded by Corydon76 (license 14)
Tested by: ruddy
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@152369 65c4cc65-6c06-0410-ace0-fbb531ad65f3
administrator to make the decision of what permissions will actually be given,
through the use of the process umask.
(Closes issue# 13751)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@151371 65c4cc65-6c06-0410-ace0-fbb531ad65f3
1) rename 'struct server_args' to 'struct ast_tcptls_session_args', to follow coding guidelines
2) make ast_make_file_from_fd() static and rename it to something that indicates what it really is for (again coding guidelines)
3) rename address variables inside 'struct ast_tcptls_session_args' to be more descriptive (dare i say it... coding guidelines)
4) change ast_tcptls_client_start() to use the new 'remote_address' field of the session args for the destination of the connection, and use the 'local_address' field to bind() the socket to the proper source address, if one is supplied
5) in chan_sip, ensure that we pass in the PP address we are bound to when creating outbound (client) connections, so that our connections will appear from the correct address
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@151101 65c4cc65-6c06-0410-ace0-fbb531ad65f3
(closes issue #11040)
Reported by: DEA
Patches:
rt-meetme-flag-fixes-v2.txt uploaded by DEA (license 3)
with additional fixes by me
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@150384 65c4cc65-6c06-0410-ace0-fbb531ad65f3
with regards to braces used on if statements.
(closes issue #13696)
Reported by: alecdavis
Patches:
app_skel.bug13696B.115850.diff.txt uploaded by alecdavis (license 585)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@150255 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r149200 | mmichelson | 2008-10-14 17:40:42 -0500 (Tue, 14 Oct 2008) | 12 lines
Update the queue with the correct number of calls and
whether the call was completed within the service level
when a transfer takes place. This way, we do not "break"
the leastrecent and fewestcalls strategies by not logging
a call until after the transferred call has ended.
(closes issue #13395)
Reported by: Marquis
Patches:
app_queue.c.transfer.patch uploaded by Marquis (license 32)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@149201 65c4cc65-6c06-0410-ace0-fbb531ad65f3
(closes issue #13688)
Reported by: irroot
Patches:
app_fax-span6.patch uploaded by irroot (license 52) with minor modifications by me
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@148868 65c4cc65-6c06-0410-ace0-fbb531ad65f3
'update2', which permits updates which match across multiple columns, instead
of requiring all tables to have a single unique identifier. All of the other
API calls with the exception of 'update' already had the ability to match on
multiple fields, so it was a missing and very desireable feature that an API
call implementing an update should have this, too.
This does not change any outward performance of Asterisk, but it should make
life easier for application developers who use the RealTime framework.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@148570 65c4cc65-6c06-0410-ace0-fbb531ad65f3
app_voicemail. Instead, include it where it is needed. This turned out to be a
relatively minor issue because other headers include logger.h as well.
Need to test -addons before merging this back to 1.6.0.
(closes issue #13605)
Reported by: tomo1657
Patches:
13605_seanbright.diff uploaded by seanbright (license 71)
Tested by: mmichelson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@148200 65c4cc65-6c06-0410-ace0-fbb531ad65f3
read the Urgent flag value from the IMAP headers.
(closes issue #13652)
Reported by: jaroth
Patches:
imapheaders.patch uploaded by jaroth (license 50)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@148144 65c4cc65-6c06-0410-ace0-fbb531ad65f3
adding the imapsecret alias for imappassword. Will rethink this one and
give it another shot on a rainy day TBD.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@147265 65c4cc65-6c06-0410-ace0-fbb531ad65f3
so include 'imapsecret' as an alias to 'imappassword' (and print a little notice
nudging users toward the right option name).
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@147191 65c4cc65-6c06-0410-ace0-fbb531ad65f3
our DAHDI operator mode stuff. Technically, it would work fine, as 'DAH'
is currently unique amongst our channel technologies, but as Jared points
out:
<@jsmith> Sure... as long as the technology starts whith DAH.... but
it could be DAHDOO!
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@147050 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This allows for the ODBC parts to work on OpenBSD as well.
99.99% of the work is done by seanbright (bow, bow) and I actually
did nothing but test and yell at him that it still didn't work :)
Thanks for helping out !
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@146925 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r146799 | tilghman | 2008-10-06 15:52:04 -0500 (Mon, 06 Oct 2008) | 8 lines
Dialplan functions should not actually return 0, unless they have modified the
workspace. To signal an error (and no change to the workspace), -1 should be
returned instead.
(closes issue #13340)
Reported by: kryptolus
Patches:
20080827__bug13340__2.diff.txt uploaded by Corydon76 (license 14)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@146802 65c4cc65-6c06-0410-ace0-fbb531ad65f3
and "leavewhenempty" options are configured in queues.conf.
Instead of using vague terms like "yes," "no," "loose," and
"strict," we now accept a comma-separated list of values
to determine when to consider a member available.
Extended details can be found in the queues.conf.sample
file. Note also that the above four referenced values are
still accepted for backwards-compatibility, but are mapped
internally to the new method of representing the option.
AST-105
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@146640 65c4cc65-6c06-0410-ace0-fbb531ad65f3
problem within strptime(3), which we are correcting here with ast_strptime().
(closes issue #11040)
Reported by: DEA
Patches:
20080910__bug11040.diff.txt uploaded by Corydon76 (license 14)
Tested by: DEA
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@145649 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This work is done by lmadsen, junky and mvanbaak
during AstriDevCon.
This is the second audit the CLI got, and
this time lmadsen made sure he had _ALL_ modules
loaded that have CLI commands in them.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@145121 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Reported by: nickpeirson
The user attached a patch, but the license is not yet
recorded. I took the liberty of finding and replacing
ALL index() calls with strchr() calls, and that
involves more than just main/pbx.c;
chan_oss, app_playback, func_cut also had calls
to index(), and I changed them out. 1.4 had no
references to index() at all.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@144569 65c4cc65-6c06-0410-ace0-fbb531ad65f3
when a file is invalid from when a file is missing. This is most important when
we have two configuration files. Consider the following example:
Old system:
sip.conf users.conf Old result New result
======== ========== ========== ==========
Missing Missing SIP doesn't load SIP doesn't load
Missing OK SIP doesn't load SIP doesn't load
Missing Invalid SIP doesn't load SIP doesn't load
OK Missing SIP loads SIP loads
OK OK SIP loads SIP loads
OK Invalid SIP loads incompletely SIP doesn't load
Invalid Missing SIP doesn't load SIP doesn't load
Invalid OK SIP doesn't load SIP doesn't load
Invalid Invalid SIP doesn't load SIP doesn't load
So in the case when users.conf doesn't load because there's a typo that
disrupts the syntax, we may only partially load users, instead of failing with
an error, which may cause some calls not to get processed. Worse yet, the old
system would do this with no indication that anything was even wrong.
(closes issue #10690)
Reported by: dtyoo
Patches:
20080716__bug10690.diff.txt uploaded by Corydon76 (license 14)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@142992 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r142675 | murf | 2008-09-11 22:29:34 -0600 (Thu, 11 Sep 2008) | 29 lines
Tested by: sergee, murf, chris-mac, andrew, KNK
This is a "second attempt" to restore the previous "endbeforeh" behavior
in 1.4 and up. In order to capture information concerning all the
legs of transfers in all their infinite combinations, I was forced
to this particular solution by a chain of logical necessities, the
first being that I was not allowed to rewrite the CDR mechanism from
the ground up!
This change basically leaves the original machinery alone, which allows
IVR and local channel type situations to generate CDR's as normal, but
a channel flag can be set to suppress the normal running of the h exten.
That flag would be set by the code that runs the h exten from the
ast_bridge_call routine, to prevent the h exten from being run twice.
Also, a flag in the ast_bridge_config struct passed into ast_bridge_call
can be used to suppress the running of the h exten in that routine. This
would happen, for instance, if you use the 'g' option in the Dial app.
Running this routine 'early' allows not only the CDR() func to be used
in the h extension for reading CDR variables, but also allows them to
be modified before the CDR is posted to the backends.
While I dearly hope that this patch overcomes all problems, and
introduces no new problems, reality suggests that surely someone
will have problems. In this case, please re-open 13251 (or 13289),
and we'll see if we can't fix any remaining issues.
** trunk note: some code to suppress the h exten being run
from app_queue was added; for the 'continue' option available
only in trunk/1.6.x.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@142676 65c4cc65-6c06-0410-ace0-fbb531ad65f3
issue 12979. The difference between this and the
1.6.0 and 1.6.1 versions is that this is a much more
invasive change. With this, we completely get rid
of the interfaces list, along with all its helper
functions.
Let me take a moment to say that this change personally
excites me since it may mean huge steps forward regarding
proper lock order in app_queue without having to strew
seemingly unnecessary locks all over the place. It also
results in a huge reduction in lines of code and complexity.
Way to go Brett!
(closes issue #12979)
Reported by: sigxcpu
Patches:
20080710_issue12979_queue_custom_state_interface_trunk_2.diff uploaded by bbryant (license 36)
Tested by: sigxcpu, putnopvut
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@142146 65c4cc65-6c06-0410-ace0-fbb531ad65f3
messages weren't showing up when I had set the debug level
high for just app_queue.c. It's because we were only checking
the global option_debug variable instead of using the awesome
macro which checks both the global and file-specific value
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@141998 65c4cc65-6c06-0410-ace0-fbb531ad65f3
status is if the queue has a joinempty or a leavewhenempty
setting which could cause the caller to not join the queue
or exit the queue. Prior to this patch, we could potentially
traverse the entire queue's member list for no reason since
even if the members are currently not available in some way
we're going to let the caller join the queue anyway.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@141906 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r140751 | mmichelson | 2008-09-02 18:47:49 -0500 (Tue, 02 Sep 2008) | 6 lines
After adding the context checking to app_voicemail
for IMAP storage, I left out a crucial place to
copy the context to the vm_state structure. This
is the correction.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@140752 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r140488 | mmichelson | 2008-08-29 12:34:17 -0500 (Fri, 29 Aug 2008) | 22 lines
After working on the ao2_containers branch, I noticed
something a bit strange. In all cases where we provide
a callback function to ao2_container_alloc, the callback
function would only return 0 or CMP_MATCH. After inspecting
the ao2_callback() code carefully, I found that if you're
only looking for one specific item, then you should return
CMP_MATCH | CMP_STOP. Otherwise, astobj2 will continue
traversing the current bucket until the end searching for
more matches.
In cases like chan_iax2 where in 1.4, all the peers are
shoved into a single bucket, this makes for potentially
terrible performance since the entire bucket will be
traversed even if the peer is one of the first ones come
across in the bucket.
All the changes I have made were for cases where the
callback function defined was passed to ao2_container_alloc
so that calls to ao2_find could find a unique instance
of whatever object was being stored in the container.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@140489 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r140421 | mmichelson | 2008-08-29 11:01:07 -0500 (Fri, 29 Aug 2008) | 12 lines
Add context checking when retrieving a vm_state.
This was causing a problem for people who had identically
named mailboxes in separate voicemail contexts.
This commit affects IMAP storage only.
(closes issue #13194)
Reported by: moliveras
Patches:
13194.patch uploaded by putnopvut (license 60)
Tested by: putnopvut, moliveras
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@140422 65c4cc65-6c06-0410-ace0-fbb531ad65f3
event if eventwhencalled is set to "vars" in
queues.conf. Yay for consistency.
(closes issue #13369)
Reported by: srt
Patches:
13369_agentringnoanswer_variables.diff uploaded by srt (license 378)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@139832 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r139347 | murf | 2008-08-21 17:03:50 -0600 (Thu, 21 Aug 2008) | 47 lines
(closes issue #13251)
Reported by: sergee
Tested by: murf
THis is a bold move for a static release fix, but I wouldn't have
made it if I didn't feel confident (at least a *bit* confident)
that it wouldn't mess everyone up.
The reasoning goes something like this:
1. We simply cannot do anything with CDR's at the current point
(in pbx.c, after the __ast_pbx_run loop). It's way too late to
have any affect on the CDRs. The CDR is already posted and gone,
and the remnants have been cleared.
2. I was very much afraid that moving the running of the 'h'
extension down into the bridge code (where it would be now
practical to do it), would result in a lot more calls to the
'h' exten, so I implemented it as another exten under another
name, but found, to my pleasant surprise, that there was a
1:1 correspondence to the running of the 'h' exten in the
pbx_run loop, and the new spot at the end of the bridge.
So, I ifdef'd out the current 'h' loop, and moved it into
the bridge code. The only difference I can see is the stuff
about the AST_PBX_KEEPALIVE, and hopefully, if this
is still an important decision point, I can replicate it
if there are complaints. To be perfectly honest,
the KEEPALIVE situation is not totally clear to me,
and how it relates to a post-bridge situation is less
clear. I suspect the users will point out everything
in total clarity if this steps on anyone's toes!
3. I temporarily swap the bridge_cdr into the channel
before running the 'h' exten, which makes it possible
for users to edit the cdr before it goes out the door.
And, of course, with the endbeforehexten config var set,
the users can also get at the billsec/duration vals.
After the h exten finishes, the cdr is swapped back
and processing continues as normal.
Please, all who deal with CDR's, please test this version
of Asterisk, and file bug reports as appropriate!
........
I also made a little fix to the app_dial's 'e' option,
that is related to my updates.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@139627 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r139213 | russell | 2008-08-20 17:14:35 -0500 (Wed, 20 Aug 2008) | 11 lines
Fix a crash in the ChanSpy application. The issue here is that if you call
ChanSpy and specify a spy group, and sit in the application long enough looping
through the channel list, you will eventually run out of stack space and the
application with exit with a seg fault. The backtrace was always inside of
a harmless snprintf() call, so it was tricky to track down. However, it turned
out that the call to snprintf() was just the biggest stack consumer in this
code path, so it would always be the first one to hit the boundary.
(closes issue #13338)
Reported by: ruddy
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@139215 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r138886 | mmichelson | 2008-08-19 13:50:53 -0500 (Tue, 19 Aug 2008) | 23 lines
Add a lock and unlock prior to the destruction of the chanspy_ds
lock to ensure that no other threads still have it locked. While
this should not happen under normal circumstances, it appears that
if the spyer and spyee hang up at nearly the same time, the following
may occur.
1. ast_channel_free is called on the spyee's channel.
2. The chanspy datastore is removed from the spyee's channel in
ast_channel_free.
3. In the spyer's thread, the spyer attempts to remove and destroy the datastore
from the spyee channel, but the datastore has already been removed in step 2,
so the spyer continues in the code.
4. The spyee's thread continues and calls the datastore's destroy callback,
chanspy_ds_destroy. This involves locking the chanspy_ds.
5. Now the spyer attempts to destroy the chanspy_ds lock. The problem is that in step 4,
the spyee has locked this lock, meaning that the spyer is attempting to destroy a lock
which is currently locked by another thread.
The backtrace provided in issue #12969 supports the idea that this is possible
(and has even occurred). This commit does not close the issue, but should help
in preventing one type of crash associated with the use of app_chanspy.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@138887 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r138685 | mmichelson | 2008-08-18 15:01:14 -0500 (Mon, 18 Aug 2008) | 10 lines
Change the inequalities used in app_queue with regards
to timeouts from being strict to non-strict for more
accuracy.
(closes issue #13239)
Reported by: atis
Patches:
app_queue_timeouts_v2.patch uploaded by atis (license 242)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@138687 65c4cc65-6c06-0410-ace0-fbb531ad65f3
trunk.
For an explanation of what "imap_consistency" is,
please see svn revision 134223 to the 1.4 branch.
Coincidentally, this also fixes a recent bug report
regarding the inability to save messages to the new
folder when using IMAP storage since they will would
be flagged as "seen" and not be recognized as new
messages.
(closes issue #13234)
Reported by: jaroth
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@136715 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r135799 | murf | 2008-08-05 17:13:20 -0600 (Tue, 05 Aug 2008) | 34 lines
(closes issue #12982)
Reported by: bcnit
Tested by: murf
I discovered that also, in the previous bug fixes and changes,
the cdr.conf 'unanswered' option is not being obeyed, so
I fixed this.
And, yes, there are two 'answer' times involved in this
scenario, and I would agree with you, that the first
answer time is the time that should appear in the CDR.
(the second 'answer' time is the time that the bridge
was begun).
I made the necessary adjustments, recording the first
answer time into the peer cdr, and then using that to
override the bridge cdr's value.
To get the 'unanswered' CDRs to appear, I purposely
output them, using the dial cmd to mark them as
DIALED (with a new flag), and outputting them if
they bear that flag, and you are in the right mode.
I also corrected one small mention of the Zap device
to equally consider the dahdi device.
I heavily tested 10-sec-wait macros in dial, and
without the macro call; I tested hangups while the
macro was running vs. letting the macro complete
and the bridge form. Looks OK. Removed all the
instrumentation and debug.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@135821 65c4cc65-6c06-0410-ace0-fbb531ad65f3
such as "Work" and "Family" would be subfolders of the
INBOX. This is an invalid assumption to make, but it could
be desirable to set up folders in this manner, so a new
option for voicemail.conf, "imapparentfolder" has been
added to allow for this.
(closes issue #13142)
Reported by: jaroth
Patches:
parentfolder.patch uploaded by jaroth (license 50)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@135067 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r134758 | mmichelson | 2008-07-31 10:56:18 -0500 (Thu, 31 Jul 2008) | 16 lines
Add more timeout checks into app_queue, specifically
targeting areas where an unknown and potentially
long time has just elapsed. Also added a check
to try_calling() to return early if the timeout
has elapsed instead of potentially setting a negative
timeout for the call (thus making it have *no* timeout
at all).
(closes issue #13186)
Reported by: miquel_cabrespina
Patches:
13186.diff uploaded by putnopvut (license 60)
Tested by: miquel_cabrespina
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@134759 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r133169 | mmichelson | 2008-07-23 14:39:47 -0500 (Wed, 23 Jul 2008) | 12 lines
As suggested by seanbright, the PSEUDO_CHAN_LEN in
app_chanspy should be set at load time, not at compile
time, since dahdi_chan_name is determined at load time.
Also changed the next_unique_id_to_use to have the
static qualifier.
Also added the dahdi_chan_name_len variable so that
strlen(dahdi_chan_name) isn't necessary. Thanks to
seanbright for the suggestion.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@133171 65c4cc65-6c06-0410-ace0-fbb531ad65f3
own repository, and must be installed like any other library for Asterisk to
use. The two modules that require it are codec_resample and app_jack.
To install libresample:
$ svn co http://svn.digium.com/svn/libresample/trunk libresample
$ cd libresample
$ ./configure
$ make
$ sudo make install
This code is currently in our own repository because the build system did not
include the appropriate targets for building a dynamic library or for installing
the library.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@132390 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r131369 | mmichelson | 2008-07-16 15:23:02 -0500 (Wed, 16 Jul 2008) | 14 lines
Move the init_queue call back to where it used to be (changed
Sept 12 last year). It was moved then to prevent a memory leak.
Since then, the same memory leak recurred and was fixed in a
better way.
Now it has been found that the placement of this init_queue
call can cause problems if a realtime queue has values changed
to an empty string. The problem is that the default value
for that queue parameter would not be set.
(closes issue #13084)
Reported by: elbriga
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@131375 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r131299 | mmichelson | 2008-07-16 13:57:34 -0500 (Wed, 16 Jul 2008) | 13 lines
Make absolutely certain that the transfer datastore
is removed from the calling channel once the caller
is finished in the queue. This could have weird con-
sequences when dialing local queue members when multiple
transfers occur on a single call.
Also fixed a memory leak that would occur when an
attended transfer occurred from a queue member.
(closes issue #13047)
Reported by: festr
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@131300 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r130959 | tilghman | 2008-07-15 12:19:13 -0500 (Tue, 15 Jul 2008) | 8 lines
astman_send_error does not need a newline appended -- the API takes care of
that for us.
(closes issue #13068)
Reported by: gknispel_proformatique
Patches:
asterisk_1_4_astman_send.patch uploaded by gknispel (license 261)
asterisk_trunk_astman_send.patch uploaded by gknispel (license 261)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@131044 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r130792 | mmichelson | 2008-07-14 12:50:21 -0500 (Mon, 14 Jul 2008) | 8 lines
Add a check to the CAN_EARLY_BRIDGE macro in app_dial to
be sure there are no audiohooks present on the channels
involved. This fixed a one-way audio situation I had in
my test setup. I couldn't find any open issues that suggested
one-way audio with regards to mixmonitor (or other audiohook)
usage, though.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@130794 65c4cc65-6c06-0410-ace0-fbb531ad65f3
fn2 was used in three functions. In every case, it was initialized
in the function it was used in. This meant there was no need
to have it in a malloc'd structure just taking up space. Furthermore
two of the functions it was used in were completely unnecessary since
fn2 was set to exactly the same value as the vm_state's fn string.
fn2 was a char array sized at PATH_MAX. On my system, PATH_MAX is
4096. This equates to a 4K memory savings per vm_state allocated.
Since there is a vm_state malloc'd for every voicemail user on
the system, this could potentially add up nicely if there are lots
of users. In addition, a vm_state is allocated on the stack each
time a caller calls the VoiceMailMain application, meaning that
there is a significant stack savings with this patch too.
Of course, a single vm_state struct still takes up approximately
20K on my system (when using IMAP storage. Without IMAP storage,
there would be about another 300 bytes fewer usage), even with
this removal. Further optimizations are probably possible,
but most likely not as easy as this one.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@129734 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r129149 | tilghman | 2008-07-08 15:27:47 -0500 (Tue, 08 Jul 2008) | 8 lines
Cause SIP to return a 480 instead of a 404 when a sip peer exists, but is not
registered.
(closes issue #12885)
Reported by: ibc
Patches:
20080701__bug12885__2.diff.txt uploaded by Corydon76 (license 14)
Tested by: ibc
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@129152 65c4cc65-6c06-0410-ace0-fbb531ad65f3
t38_terminal_release, and make sure that the phase E handler gets called
with proper status.
(closes issue #13020)
Reported by: dimas
Patches:
v1-appfax.patch uploaded by dimas (license 88)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@129006 65c4cc65-6c06-0410-ace0-fbb531ad65f3
app_voicemail would attempt to play a file called vm-foo instead of playing
vm-INBOX to play the "new" sound file. This commit fixes that issue.
This may fix one of the problems reported in issue #12987
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@128731 65c4cc65-6c06-0410-ace0-fbb531ad65f3
MixMonitor to mix audio. However, it was pointed out to me that because
of this, the command set for the MONITOR_EXEC variable is ignored as well.
This means that people can't do their own custom mixing commands at the end
of recordings in order to make, for instance, stereo recordings of calls.
With this patch, app_queue will set the "joinfiles" variable for the channel's
monitor if MONITOR_EXEC is not zero-length. This means that for normal audio
mixing, MixMonitor is still the preferred choice, but we allow custom
mixing to be done with the two Monitor streams if desired.
(closes issue #12923)
Reported by: snyfer
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@125649 65c4cc65-6c06-0410-ace0-fbb531ad65f3
to how language is handled between two channels whose native
language is different. Prior to this patch, app_dial would have
the callee inherit the caller's language, and app_queue would not.
After this patch, app_dial no longer has the language inheritance
capability. This seems to make the most sense since it seems more
natural for a person to hear files played back in his/her native
language instead of the language of the person on the far end of
the call. See the CHANGES file for hints on how to keep the
previous behavior of app_dial if desired.
(closes issue #12489)
Reported by: bcnit
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@125647 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r125585 | mmichelson | 2008-06-26 17:52:39 -0500 (Thu, 26 Jun 2008) | 11 lines
Add the interface of a queue member to the output of the "queue show" command
so that it can easily be associated with a queue member's name. This helps
so that the appropriate queue member can be removed or paused since the
interface is required, not the member's name.
(closes issue #12783)
Reported by: davevg
Patches:
app_queue.diff uploaded by davevg (license 209) with small mod from me
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@125586 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r125476 | mmichelson | 2008-06-26 15:56:01 -0500 (Thu, 26 Jun 2008) | 11 lines
Prior to this patch, the "queue show" command used cached
information for realtime queues instead of giving up-to-date
info. Now realtime is queried for the latest and greatest in
queue info.
(closes issue #12858)
Reported by: bcnit
Patches:
queue_show.patch uploaded by putnopvut (license 60)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@125477 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r125132 | kpfleming | 2008-06-25 17:21:30 -0500 (Wed, 25 Jun 2008) | 10 lines
allow tonezone to live in a different place than DAHDI/Zaptel, since dahdi-tools and dahdi-linux are now separate packages and can be installed in different places
don't include tonezone.h in dahdi_compat.h, because only a couple of modules need it
get app_rpt building again after the DAHDI changes
(closes issue #12911)
Reported by: tzafrir
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@125138 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r124910 | tilghman | 2008-06-24 16:08:52 -0500 (Tue, 24 Jun 2008) | 8 lines
Occasionally control characters find their way into CallerID. These need to
be stripped prior to placing CallerID in the headers of an email.
(closes issue #12759)
Reported by: RobH
Patches:
20080602__bug12759__2.diff.txt uploaded by Corydon76 (license 14)
Tested by: RobH
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@124912 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Joshua Colp
David Vossel
Mark Michelson
Brian Degenhardt
Russell Bryant
Steve Murphy
Tilghman Lesher
Dwayne M. Hubbard
Jeff Peeler
Terry Wilson
Olle Johansson
Sean Bright
Kevin P. Fleming
BJ Weschke
Eliel C. Sardanons
Matthew Nicholson
Jason Parker
Michiel van Baak
Brett Bryant
Luigi Rizzo