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
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@334012 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@334009 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@334006 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@333836 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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>,
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@333507 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@332503 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@332264 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@332118 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@332021 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* 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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@331955 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* 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
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@331771 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@331714 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@331517 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* 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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@331248 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@330705 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@330581 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
..........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@330050 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@329012 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@328935 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@328823 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@328608 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@327211 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@326411 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@326291 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@326144 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* 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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@325935 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@325212 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@324914 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@324484 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@324481 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@324237 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@323371 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@322585 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@322322 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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]
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@321155 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@321100 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@320883 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@320180 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@319938 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@319654 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@319552 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
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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
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@319469 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@318783 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@318671 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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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/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@318549 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@318499 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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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)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@318337 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
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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
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@318231 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@317867 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@317865 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Matthew Nicholson
Richard Mudgett
Kinsey Moore
Terry Wilson
Tzafrir Cohen
Paul Belanger
Jonathan Rose
David Vossel
Jason Parker
Sean Bright
Mark Murawki
Leif Madsen
Russell Bryant
Tilghman Lesher
Kevin P. Fleming
Gregory Nietsky
Brett Bryant
Alec L Davis