21679 Commits (5f1a062fa6f012bbed54e2e4797fdd19a6048c18)

 

All Branches   

Search

Author	SHA1	Message	Date
Russell Bryant	5f1a062fa6	Merged revisions 336877 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r336877 | russell | 2011-09-19 19:56:20 -0500 (Mon, 19 Sep 2011) | 36 lines Fix crashes in ast_rtcp_write(). This patch addresses crashes related to RTCP handling. The backtraces just show a crash in ast_rtcp_write() where it appears that the RTP instance is no longer valid. There is a race condition with scheduled RTCP transmissions and the destruction of the RTP instance. This patch utilizes the fact that ast_rtp_instance is a reference counted object and ensures that it will not get destroyed while a reference is still around due to scheduled RTCP transmissions. RTCP transmissions are scheduled and executed from the chan_sip scheduler context. This scheduler context is processed in the SIP monitor thread. The destruction of an RTP instance occurs when the associated sip_pvt gets destroyed (which happens when the sip_pvt reference count reaches 0). However, the SIP monitor thread is not the only thread that can cause a sip_pvt to get destroyed. The sip_hangup function, executed from a channel thread, also decrements the reference count on a sip_pvt and could cause it to get destroyed. While this is being changed anyway, the patch also removes calling ast_sched_del() from within the RTCP scheduler callback. It's not helpful. Simply returning 0 prevents the callback from being rescheduled. (closes issue ASTERISK-18570) Related issues that look like they are the same problem: (issue ASTERISK-17560) (issue ASTERISK-15406) (issue ASTERISK-15257) (issue ASTERISK-13334) (issue ASTERISK-9977) (issue ASTERISK-9716) Review: https://reviewboard.asterisk.org/r/1444/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@336878 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	17124a2510	Merged revisions 336791 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r336791 | twilson | 2011-09-19 17:07:58 -0500 (Mon, 19 Sep 2011) | 2 lines Don't interfere with T.38 reinvites This is an update to the fix for ASTERISK-18340 and ASTERISK-17725 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@336792 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Tilghman Lesher	49882ed416	Ensure substring will not be found in the previous match. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@336789 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Tilghman Lesher	e1c3a38653	Merged revisions 336733 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r336733 | tilghman | 2011-09-19 15:27:03 -0500 (Mon, 19 Sep 2011) | 11 lines Various changes to allow 1.8 to compile on Mac OS X Lion (10.7) * Makefile workaround for 10.6 extended to work on 10.7 and later. * Now uses the 'weak' symbol for Lion systems, which no longer support 'weak_import' Closes ASTERISK-17612. Closes ASTERISK-18213. Tested by: tilghman, oej. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@336734 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	8285989c1c	Merged revisions 336716 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r336716 | jrose | 2011-09-19 15:07:36 -0500 (Mon, 19 Sep 2011) | 7 lines Document applications that play audio and do not answer unanswered calls. This patch is part of an effort to document early media and its usage. If you are interested in contributing to this documentation effort, there are probably other applications worth documenting as well as an Asterisk wiki article at https://wiki.asterisk.org/wiki/display/AST/Early+Media+and+the+Progress+Application ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@336717 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	6fd0d3805d	Merged revisions 336658 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r336658 | rmudgett | 2011-09-19 13:46:40 -0500 (Mon, 19 Sep 2011) | 31 lines Made Dial d and H options no longer immediately auto-answer the calling leg. The Dial d and H options break DTMF attended transfer atxferdropcall option. 1) Party A calls party B. 2) Party B does a DTMF attended transfer to Party C. If the dialplan uses the Dial d or H options to call Party C then the Dial application answers the call immediately before initiating the call leg to Party C. The premature answer causes the transfer code to not invoke the atxferdropcall=no behavior for a blonde transfer since Party C has "answered". The transfer code thinks that Party B has "consulted" with Party C when Party B hangs up and completes the transfer to Party A. Party A now hears ringback until Party C actually answers. ASTERISK-13294 Dial d option. ASTERISK-11067 Dial H option to disconnect before answer. The referenced issues made Dial answer with the d and H options because many SIP and ISDN phones cannot send DTMF before the call is connected. * Made require the dialplan to control when or if the call needs to be answered to use the Dial application d and H options. (The call is no longer surprise answered when using the Dial d or H options.) Review: https://reviewboard.asterisk.org/r/1381/ JIRA AST-623 JIRA AST-666 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@336659 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Leif Madsen	bf2776bef7	Merged revisions 336572 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r336572 | lmadsen | 2011-09-19 10:41:16 -0500 (Mon, 19 Sep 2011) | 7 lines Update get_ilbc_source.sh script to work again. Recently iLBC support in Asterisk has changed after the acquisition of GIPS by Google. More information about how this may affect you is available in a blog post at: http://blogs.asterisk.org/2011/09/19/ilbc-support-in-asterisk-after-googles-acquisition-of-gips/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@336573 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	b774bd8aae	Merged revisions 336569 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r336569 | rmudgett | 2011-09-19 10:25:34 -0500 (Mon, 19 Sep 2011) | 4 lines Rework sig_pri_hangup() to be simpler and clearer. JIRA AST-675 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@336570 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Olle Johansson	a4fdbcccfd	Revert accidental change ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@336504 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Olle Johansson	dd12759127	Merged revisions 336501 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r336501 | oej | 2011-09-19 15:33:50 +0200 (Mån, 19 Sep 2011) | 5 lines Add diversion header to a 302 redirect response if we have diversion data (closes issue ASTERISK-18143) patch by oej ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@336502 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Gregory Nietsky	d88b2a8f96	Merged revisions 336499 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r336499 | irroot | 2011-09-19 15:27:52 +0200 (Mon, 19 Sep 2011) | 13 lines A long time ago in a galaxy far far away a IPv6 update was made, chan_h323 was not updated causeing all to flee to chan_ooh323. the brave Jedi [asterisk developers] pondered this miscarrige of justice and restored order to the force for the sake of closing out 2 old issues. (closes issue ASTERISK-17278) (closes issue ASTERISK-17500) Reported by: dread, sybasesql Tested by: irroot Reviewed by: IRC (russellb, kpfleming) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@336500 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Olle Johansson	07b63b7705	Merged revisions 336440 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r336440 | oej | 2011-09-19 14:06:48 +0200 (Mån, 19 Sep 2011) | 2 lines Make sure manager_debug option is reset at reload ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@336441 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Olle Johansson	0cfc3c6e9e	Merged revisions 336378 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r336378 | oej | 2011-09-19 11:40:44 +0200 (Mån, 19 Sep 2011) | 9 lines Add missing unlock at MWI message sending time (closes issue ASTERISK-18573) Patches: sip_mwi_lock.patch (license #5041) by Gregory Hinton Nietsky Thanks to irrot for the reminder, to Gregory for the patch! ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@336381 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	d1dffa1ebd	Merged revisions 336314 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r336314 | twilson | 2011-09-16 17:10:56 -0500 (Fri, 16 Sep 2011) | 2 lines Whitespace fix ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@336316 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	aa01874fbd	Merged revisions 336312 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r336312 | twilson | 2011-09-16 17:04:25 -0500 (Fri, 16 Sep 2011) | 5 lines Add missing frame types to func_frame_trace Also casts control frames to the proper enum so that the compile will catch new additions. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@336313 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	1701266596	Merged revisions 336294 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r336294 | jrose | 2011-09-16 14:53:40 -0500 (Fri, 16 Sep 2011) | 13 lines Fix bad RTP media bridges in directmedia calls on peers separated by multiple Asterisk nodes. In a situation involving devices on separate Asterisk trunks, the remote RTP bridge would break when starting a call with directmedia. This patch queues a new type of control frame so that our RTP bridge loop can properly detect when these situations occur and check to see if peers need to be updated in order to send their media to the proper location. (Closes issue ASTERISK-18340) Reported by: Thomas Arimont (Closes issue ASTERISK-17725) Reported by: kwk Tested by: twilson, jrose ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@336307 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Sean Bright	04224c2036	Merged revisions 336234 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r336234 | seanbright | 2011-09-16 15:06:27 -0400 (Fri, 16 Sep 2011) | 2 lines Make a note that inotify won't work with an NFS mounted spooler directory. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@336235 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Gregory Nietsky	6b77498a29	Merged revisions 336166 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r336166 | irroot | 2011-09-16 12:09:17 +0200 (Fri, 16 Sep 2011) | 16 lines The round robin routing routine in chan_misdn.c is broken. it rotates between ports but never checks the channels in the ports. i have extensivly tested it and verified it works on 1 upto 4 ports. before the patch only 1 out of each port was used now all are used as expected. (closes issue ASTERISK-18413) Reported by: irroot Tested by: irroot Reviewed by: irroot Review: https://reviewboard.asterisk.org/r/1410/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@336167 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Gregory Nietsky	a687f85aa5	Merged revisions 336093 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r336093 | irroot | 2011-09-15 17:46:21 +0200 (Thu, 15 Sep 2011) | 20 lines Locking order in app_queue.c causes deadlocks. a channel lock must never be held with the queues container lock held. the deadlock occured on masquerade. the queues container lock is a relic of the past the old queue module lock. with ao2 there is no need to hold this lock when dealing with members this patch removes unneeded locks. (closes issue ASTERISK-18101) (closes issue ASTERISK-18487) Reported by: Paul Rolfe, Jason Legault Tested by: irroot, Jason Legault, Paul Rolfe Reviewed by: Matthew Nicholson Review: https://reviewboard.asterisk.org/r/1402/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@336094 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
David Vossel	2d4def5c7f	Removes some no-op code found in format_cap.c. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@336091 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Olle Johansson	9795ec3ef0	Meetme: Introducing a new option "k" to kill a conference if there's only a single member left. ... When using Meetme as a modular call bridge from third party applications, it's handy to make it behave like a normal call bridge. When the second to last person exists, the last person will be kicked out of the conference when this option is enabled. (closes issue ASTERISK-18234) Review: https://reviewboard.asterisk.org/r/1376/ Patch by oej, sponsored by ClearIT, Solna, Sweden git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@336042 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Gregory Nietsky	25de12ee73	Merged revisions 335978 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r335978 | irroot | 2011-09-15 10:15:22 +0200 (Thu, 15 Sep 2011) | 11 lines lock the channel before calling ast_bridged_channel() to prevent a seg fault. AMI agents list called on shutdown causes a segfault, introducing proper locking will prevent this. (closes issue ASTERISK-18092) Reported by: agustina Patches: chan_agent.patch (License #5041) patch uploaded by irroot ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@335991 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	ff58a3f1b9	Merged revisions 335911 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r335911 | rmudgett | 2011-09-14 13:21:35 -0500 (Wed, 14 Sep 2011) | 13 lines Remove unnecessary libpri dependency checks in the configure script. Using the --with-pri option with the configure script generated an error about not having PRI_L2_PERSISTENCE if you did not have the absolute latest libpri SVN checkout installed. The AST_EXT_LIB_SETUP_DEPENDENT macro in the configure.ac script seems to be for libraries that are dependent upon other libraries and not necessarily for optional/added features within a library. (closes issue ASTERISK-18535) Reported by: Michael Keuter ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@335912 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	d37ea04b2f	Merged revisions 335851 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r335851 | rmudgett | 2011-09-14 10:53:25 -0500 (Wed, 14 Sep 2011) | 11 lines Fixed cut-n-paste regression using the wrong variable. Fixes the missing DAHDI channels when using the newer chan_dahdi.conf sections for channel configuration. (closes issue ASTERISK-18496) Reported by: Sean Darcy Patches: jira_asterisk_18496_v1.8.patch (license #5621) patch uploaded by rmudgett Tested by: Sean Darcy, rmudgett ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@335852 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	7d245dbc30	Merged revisions 335790 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r335790 | mnicholson | 2011-09-14 08:28:16 -0500 (Wed, 14 Sep 2011) | 4 lines The tech and data members of fast_originate_helper are not string fields. ASTERISK-17709 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@335791 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	d297a943e2	Merged revisions 335720 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r335720 | rmudgett | 2011-09-13 17:10:15 -0500 (Tue, 13 Sep 2011) | 1 line Remove obsolete todo comment about PICKUPRESULT. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@335721 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Tzafrir Cohen	1e54d13c73	do parse defaultlanguage from asterisk.conf ... Do parse the option "defaultlanguage" from the [options] section of asterisk.conf, as in the sample config file. Otherwise the build-time default language (normally "en") is always the default one. Review: https://reviewboard.asterisk.org/r/1342/ Signed-off-by: Tzafrir Cohen (License #5035) <tzafrir.cohen@xorcom.com> Original-Commit: http://svn.digium.com/svn/asterisk/branches/1.8@335716 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@335717 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Paul Belanger	ad5e4787ed	Blocked revisions 335714 via svnmerge ... ........ r335714 | pabelanger | 2011-09-13 17:30:18 -0400 (Tue, 13 Sep 2011) | 2 lines Meetme should have 'core' support level (closes issue ASTERISK-18542) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@335715 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Tilghman Lesher	9a9b6b293b	Merged revisions 335655 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r335655 | tilghman | 2011-09-13 13:52:38 -0500 (Tue, 13 Sep 2011) | 4 lines Move mandatory checks closer to the beginning of the file. If these are going to fail, they should fail as quickly as possible. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@335656 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	4acfe8e5c9	Merged revisions 335618 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r335618 | mnicholson | 2011-09-13 13:20:52 -0500 (Tue, 13 Sep 2011) | 5 lines Don't limit the size of appdata for manager originate actions. ASTERISK-17709 Patch by: tilghman (with modifications) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@335653 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	d287e6116a	Merged revisions 335497 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r335497 | russell | 2011-09-13 02:11:36 -0500 (Tue, 13 Sep 2011) | 15 lines Fix a crash in res_ais. This patch resolves a crash observed in a load testing environment that involved the use of the res_ais module. I observed some crashes where the event delivery callback would get called, but the length parameter incidcating how much data there was to read was 0. The code assumed (with good reason I would think) that if this callback got called, there was an event available to read. However, if the rare case that there's nothing there, catch it and return instead of blowing up. More specifically, the change always ensure that the size of the received event in the cluster is always big enough to be a real ast_event. Review: https://reviewboard.asterisk.org/r/1423/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@335510 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	e0a8394e8e	Merged revisions 335433 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r335433 | mnicholson | 2011-09-12 10:54:41 -0500 (Mon, 12 Sep 2011) | 6 lines Properly set caller_warning and callee_warning before we try to use them. ASTERISK-18199 Patch by: elguero Testing by: rtang ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@335434 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	07e04a7784	Blocked revisions 335431 via svnmerge ... This change is already in asterisk 10. ........ r335431 | mnicholson | 2011-09-12 10:49:24 -0500 (Mon, 12 Sep 2011) | 5 lines Prevent a race condition when the bridge technology changes. This change was ported from asterisk 10. ASTERISK-18155 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@335432 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	a00837f83e	Merged revisions 335341 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r335341 | kmoore | 2011-09-12 09:21:17 -0500 (Mon, 12 Sep 2011) | 10 lines Ensure frames are not written to dialed channel if ringback is requested When a single channel was dialed and there was media to be forwarded to the calling channel, the media was written without regard for ringback causing silence to be heard in some circumstances. This regression was introduced when the meaning of "single" changed to mean only the number of channels dialed. (closes issue ASTERISK-18083) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@335346 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Olle Johansson	7c7e697c3a	Merged revisions 335319 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r335319 | oej | 2011-09-12 15:25:30 +0200 (Mån, 12 Sep 2011) | 12 lines Lock the peer->mvipvt to avoid crashes with SIP history enabled After the launch of 1.6 event-based MWI we have two threads handling the peer->mwipvt, which cause issues with SIP history additions in combination with the max limit for number of history entries. Review: https://reviewboard.asterisk.org/r/1373/ (closes issue ASTERISK-18288) Thanks to irrot for peer review. Work with this bug funded by IPvision AS ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@335323 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	10f33d5e70	Merged revisions 335320 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r335320 | kmoore | 2011-09-12 08:25:42 -0500 (Mon, 12 Sep 2011) | 9 lines Prevent IAX2 from getting IPv6 addresses via DNS IAX2 does not support IPv6 and getting such addresses from DNS can cause error messages on the remote end involving bad IPv4 address casts in the presence of IPv6/IPv4 tunnels. This patch ensures that IAX2 will not encounter IPv6 addresses via DNS queries. (closes issue ASTERISK-18090) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@335321 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Stefan Schmidt	0f50837b20	Merged revisions 335259 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r335259 | schmidts | 2011-09-12 11:09:19 +0000 (Mon, 12 Sep 2011) | 6 lines build_peer doesnt unlink a peer object from peers_by_ip container which leads to a wrong refcounter value. adding an ao2_unlink from the peers_by_ip container fix it. Review: https://reviewboard.asterisk.org/r/1428/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@335260 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	4e57652651	Merged revisions 335064 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r335064 | mjordan | 2011-09-09 11:09:09 -0500 (Fri, 09 Sep 2011) | 23 lines Updated SIP 484 handling; added Incomplete control frame When a SIP phone uses the dial application and receives a 484 Address Incomplete response, if overlapped dialing is enabled for SIP, then the 484 Address Incomplete is forwarded back to the SIP phone and the HANGUPCAUSE channel variable is set to 28. Previously, the Incomplete application dialplan logic was automatically triggered; now, explicit dialplan usage of the application is required. Additionally, this patch adds a new AST_CONTOL_FRAME type called AST_CONTROL_INCOMPLETE. If a channel driver receives this control frame, it is an indication that the dialplan expects more digits back from the device. If the device supports overlap dialing it should attempt to notify the device that the dialplan is waiting for more digits; otherwise, it can handle the frame in a manner appropriate to the channel driver. (closes issue ASTERISK-17288) Reported by: Mikael Carlsson Tested by: Matthew Jordan Review: https://reviewboard.asterisk.org/r/1416/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@335078 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Gregory Nietsky	d5544999b5	Move code for VALID_EXTEN from app_readexten to func_dialplan ...	14 years ago
Richard Mudgett	ce8cb2769e	Merged revisions 334953 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r334953 | rmudgett | 2011-09-08 17:27:40 -0500 (Thu, 08 Sep 2011) | 10 lines Fix crash with res_fax when MALLOC_DEBUG and "core stop gracefully" are used. Asterisk crashes if MALLOC_DEBUG is enabled when res_fax tries to unregister its logger level. * Make ast_logger_unregister_level() use ast_free() instead of free(). When MALLOC_DEBUG is enabled, ast_free() does not degenerate into a call to free(). Therefore, if you allocated memory with a form of ast_malloc you must free it with ast_free. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@334954 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Paul Belanger	d4198b3ecf	Merged revisions 334843 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r334843 | pabelanger | 2011-09-07 15:35:52 -0400 (Wed, 07 Sep 2011) | 4 lines Cleanup chan_iax2.c log messages Review: https://code.asterisk.org/code/cru/CR-AST-11 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@334844 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	7617bf7bfe	Merged revisions 334840 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r334840 | rmudgett | 2011-09-07 14:31:44 -0500 (Wed, 07 Sep 2011) | 10 lines Fix AMI action Park crash. * Made AMI action Park not say anything to the parker channel (AMI header Channel2) since the AMI action is a third party parking the call. (This is a change in behavior that cannot be preserved without a lot of effort.) * Made not play pbx-parkingfailed if the Park 's' option is used. JIRA AST-660 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@334841 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Stefan Schmidt	3badf91d58	Merged revisions 334682 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r334682 | schmidts | 2011-09-07 13:26:50 +0000 (Wed, 07 Sep 2011) | 3 lines Adding the Feature to sent a Reason Header in a SIP Cancel message by set the flag AST_FLAG_ANSWERED_ELSEWHERE before doing a masquerade in the pickup function. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@334747 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Stefan Schmidt	8211ef0f13	another clean up ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@334746 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Stefan Schmidt	1e13555525	Adding the Feature to sent a Reason Header in a SIP Cancel message by set the flag AST_FLAG_ANSWERED_ELSEWHERE before doing a masquerade in the pickup function. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@334682 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Alec L Davis	d1af4d4f15	Merged revisions 334620 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r334620 | alecdavis | 2011-09-07 20:12:49 +1200 (Wed, 07 Sep 2011) | 2 lines peroid typo ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@334621 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Alec L Davis	b8a0261ffb	Merged revisions 334616 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r334616 | alecdavis | 2011-09-07 19:33:39 +1200 (Wed, 07 Sep 2011) | 10 lines Prevent segfault if call arrives before Asterisk is fully booted. Prevent ast_pbx_start and ast_run_start from starting a new thread unless asterisk is fully booted. alecdavis (license 585) Tested by: alecdavis Review: https://reviewboard.asterisk.org/r/1407/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@334617 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Paul Belanger	67d2ce9e73	authdebug is now disabled by default ... To enable this functionaility again set authdebug = yes in iax.conf Review: https://reviewboard.asterisk.org/r/1414/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@334514 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Gregory Nietsky	5bf2fe764a	Merged revisions 334453 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r334453 | irroot | 2011-09-06 15:48:03 +0200 (Tue, 06 Sep 2011) | 13 lines Make SQL query in app_voicemail.c portable LIMIT is not portable. Regression from r312212 (closes issue ASTERISK-18255) Reported by: Leif Madsen Tested by: Leif Madsen Review: https://reviewboard.asterisk.org/r/1415/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@334455 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	2ce6c668ca	Merged revisions 334355 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r334355 | rmudgett | 2011-09-02 15:59:49 -0500 (Fri, 02 Sep 2011) | 19 lines MusicOnHold has extra unref which may lead to memory corruption and crash. The problem happens when a call is disconnected and you had started a MOH class that does not use the files mode. If you define REF_DEBUG and recreate the problem, it will announce itself with the following warning: Attempt to unref mohclass 0xb70722e0 (default) when only 1 ref remained, and class is still in a container! * Fixed moh_alloc() and moh_release() functions not handling the state->class reference consistently. (closes issue ASTERISK-18346) Reported by: Mark Murawski Patches: jira_asterisk_18346_v1.8.patch (license #5621) patch uploaded by rmudgett Tested by: rmudgett, Mark Murawski Review: https://reviewboard.asterisk.org/r/1404/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@334357 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago