asterisk

Commit Graph

Author	SHA1	Message	Date
Paul Belanger	cf4d5e585a	Revert previous commit New feature should be added into trunk, unfortunately it is too late for the Asterisk 10 branch. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@337640 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	afe021b267	Forgot to svn add new files to r337595 Part of Generating security events for chan_sip (issue ASTERISK-18264) Reported by: Michael L. Young Patches: security_events_chan_sip_v4.patch (License #5026) by Michael L. Young Reviewboard: https://reviewboard.asterisk.org/r/1362/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@337597 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	857e4fdb14	Generate Security events in chan_sip using new Security Events Framework Security Events Framework was added in 1.8 and support was added for AMI to generate events at that time. This patch adds support for chan_sip to generate security events. (closes issue ASTERISK-18264) Reported by: Michael L. Young Patches: security_events_chan_sip_v4.patch (license #5026) by Michael L. Young Review: https://reviewboard.asterisk.org/r/1362/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@337595 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Gregory Nietsky	34ae017509	Merged revisions 337541 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r337541 \| irroot \| 2011-09-22 13:39:49 +0200 (Thu, 22 Sep 2011) \| 8 lines Add warned to ast_srtp to prevent errors on each frame from libsrtp The first 9 frames are not reported as some devices dont use srtp from first frame these are suppresed. the warning is then output only once every 100 frames. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@337542 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Gregory Nietsky	c28595382c	Merged revisions 337486 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r337486 \| irroot \| 2011-09-22 11:22:26 +0200 (Thu, 22 Sep 2011) \| 10 lines If IP address is used in chan_h323 host parameter of peer configuration. module tries to resolve IP address to IP address and fails. Simple fix to set family of socket this is a hangover from ipv6 changes. (closes issue ASTERISK-18237) (issue ASTERISK-17278) (issue ASTERISK-17500) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@337487 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Gregory Nietsky	ee9db5269c	Revert commit r337261 This commit is for trunk not version 10 ----- Adds a timeout argument to app_originate the default is 30s this will be used if the timout supplied is invalid or no timeout is supplied. ----- git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@337433 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Gregory Nietsky	4272dcbb1a	Merged revisions 337430 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r337430 \| irroot \| 2011-09-22 08:18:33 +0200 (Thu, 22 Sep 2011) \| 19 lines Its possible to loose audio on ast_write when the channel is not transcoded correctly. in the case of DAHDI the channel is hungup. This patch tries to "fix" the problem and make the channel compatiable and warn the user of this problem. Please note there is a underlying problem with codec negotion this does not fix the problem it does try to rectify it and prevent loss of service. Review: https://reviewboard.asterisk.org/r/1442/ (closes issue ASTERISK-17541) (closes issue ASTERISK-18063) (issue ASTERISK-14384) (issue ASTERISK-17502) (issue ASTERISK-18325) (issue ASTERISK-18422) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@337431 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Tilghman Lesher	94d511c5fa	More silly spacing changes	14 years ago
Tilghman Lesher	e71a6090f2	........	14 years ago
Tilghman Lesher	e964770729	........	14 years ago
Gregory Nietsky	614479dd24	Whitespace fixup from SRTP patch git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@337263 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Gregory Nietsky	edecb53861	Adds a timeout argument to app_originate the default is 30s this will be used if the timout supplied is invalid or no timeout is supplied. Contributed by: jacco (thank you for the work) Review: https://reviewboard.asterisk.org/r/1310/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@337261 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Olle Johansson	725dbbab22	Make ast_pbx_run() not default to s@default if extension is not found Review: https://reviewboard.asterisk.org/r/1446/ This is a bug - or architecture mistake - that has been in Asterisk for a very long time. It was exposed by the AMI originate action and possibly some other applications. Most channel drivers checks if an extension exists BEFORE starting a pbx on an inbound call, so most calls will not depend on this issue. Thanks everyone involved in the review and on IRC and the mailing list for a quick review and all the feedback. (closes issue ASTERISK-18578) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@337219 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Olle Johansson	42c967eea9	Change strictrtp option to default to yes in the RTP module Suggested by Kapejod on Facebook Review: https://reviewboard.asterisk.org/r/1448/ (closes issue ASTERISK-18587) Thanks for quick feedback to kpfleming and Tilghman --Denna och nedanstående rader kommer inte med i loggmeddelandet-- M CHANGES M configs/rtp.conf.sample M res/res_rtp_asterisk.c git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@337178 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	944cdaa94d	Merged revisions 337118 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r337118 \| mjordan \| 2011-09-20 17:38:54 -0500 (Tue, 20 Sep 2011) \| 21 lines Fix for incorrect voicemail duration in external notifications This patch fixes an issue where the voicemail duration was being reported with a duration significantly less than the actual sound file duration. Voicemails that contained mostly silence were reporting the duration of only the sound in the file, as opposed to the duration of the file with the silence. This patch fixes this by having two durations reported in the __ast_play_and_record family of functions - the sound_duration and the actual duration of the file. The sound_duration, which is optional, now reports the duration of the sound in the file, while the actual full duration of the file is reported in the duration parameter. This allows the voicemail applications to use the sound_duration for minimum duration checking, while reporting the full duration to external parties if the voicemail is kept. (issue ASTERISK-2234) (closes issue ASTERISK-16981) Reported by: Mary Ciuciu, Byron Clark, Brad House, Karsten Wemheuer, KevinH Tested by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1443 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@337120 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	c813912d14	Fix crash with STRREPLACE function. The ast_func_read() function calls the .read2 callback with the len parameter set to zero indicating no size restrictions on the supplied ast_str buffer. The value was used to dimension a local starts[] array with the array subsequently used. * Reworked the strreplace() function to perform the string replacement in a straight forward manner. Eliminated the need for the starts[] array. (closes issue ASTERISK-18545) Reported by: Federico Alves Patches: jira_asterisk_18545_v10.patch (license #5621) patch uploaded by rmudgett Tested by: rmudgett, Federico Alves git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@337119 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Leif Madsen	1eb8bca0f9	Merged revisions 337115 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r337115 \| lmadsen \| 2011-09-20 17:18:25 -0500 (Tue, 20 Sep 2011) \| 7 lines Update RedHat Init script to work with Heartbeat. The current RedHat init script was not LSB compatible. This change will make it LSB compatible so that it can work correctly with Heartbeat. (Closes issue ASTERISK-18253) Reported by: c0rnoTa ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@337116 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	09af5fa552	Merged revisions 337061 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r337061 \| kmoore \| 2011-09-20 16:04:11 -0500 (Tue, 20 Sep 2011) \| 11 lines Make CANMATCH with the new pattern match engine behave more like the old one When checking an extension for E_CANMATCH using the new extension matching algorithm, an exact match was not returned as a possible match resulting in the queue failing to allow a caller to exit on DTMF. This removes the requirement that an extension be longer than acquired digits for an E_CANMATCH operation to succeed. (closes issue ASTERISK-18044) Review: https://reviewboard.asterisk.org/r/1367/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@337062 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	8d832ccfca	Merged revisions 337007 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r337007 \| rmudgett \| 2011-09-20 14:10:30 -0500 (Tue, 20 Sep 2011) \| 15 lines Check if a channel was created before using the pointer in sig_ss7_new_ast_channel(). Fixes the crash in ASTERISK-17955 gdb-11918.txt backtrace. * Added some missing libss7 access lock protection. * Prevent cancelling the ss7_linkset() thread at inoportune times just like the pri_dchannel() thread. (issue ASTERISK-17955) Reported by: Ian M Sherman Patches: jira_asterisk_17955_v1.8.patch (license #5621) patch uploaded by rmudgett (attached to related ASTERISK-17966) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@337008 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	4fdae215bd	Merged revisions 336977 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r336977 \| rmudgett \| 2011-09-20 13:12:17 -0500 (Tue, 20 Sep 2011) \| 21 lines Fix deadlock from not releasing SS7 linkset lock. sig_ss7_hangup() failed to release the SS7 linkset lock if the call had the alreadyhungup flag set. * Made unlock the SS7 linkset lock in sig_ss7_hangup() if the alreadyhungup flag is set. * Made ss7_start_call() not hold any locks while creating the channel for an incoming call to prevent deadlock. * Made ss7_grab() a void function, since it could never fail, to simplify calling code. * Made obtain the channel lock to do softhangup in some places. Patches: jira_ast_668_v1.8.patch (license #5621) patch uploaded by rmudgett JIRA AST-668 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@336978 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Gregory Nietsky	a31b5ce87e	Allow Setting Auth Tag Bit length Based on invite or config option	14 years ago
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

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