asterisk

Commit Graph

Author	SHA1	Message	Date
TransNexus OSP Development	89205c35e8	Updated for checking OSP Toolkit version 4.0.0. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@338139 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
TransNexus OSP Development	a4c37776f4	Updated for OSP Toolkit 4.0.0. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@338136 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Paul Belanger	c19baf655e	Merged revisions 338085 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r338085 \| pabelanger \| 2011-09-27 16:13:14 -0400 (Tue, 27 Sep 2011) \| 9 lines Merged revisions 338084 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r338084 \| pabelanger \| 2011-09-27 16:10:13 -0400 (Tue, 27 Sep 2011) \| 2 lines Upgrade app_macro to core ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@338086 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Olle Johansson	6e0f7be7c9	Whitespace (red blobs) fixes git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@338042 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	55b70ae625	Merged revisions 337974 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r337974 \| rmudgett \| 2011-09-26 14:35:23 -0500 (Mon, 26 Sep 2011) \| 37 lines Merged revisions 337973 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r337973 \| rmudgett \| 2011-09-26 14:30:39 -0500 (Mon, 26 Sep 2011) \| 30 lines Fix deadlock when using dummy channels. Dummy channels created by ast_dummy_channel_alloc() should be destoyed by ast_channel_unref(). Using ast_channel_release() needlessly grabs the channel container lock and can cause a deadlock as a result. * Analyzed use of ast_dummy_channel_alloc() and made use ast_channel_unref() when done with the dummy channel. (Primary reason for the reported deadlock.) * Made app_dial.c:dial_exec_full() not call ast_call() holding any channel locks. Chan_local could not perform deadlock avoidance correctly. (Potential deadlock exposed by this issue. Secondary reason for the reported deadlock since the held lock was part of the deadlock chain.) * Fixed some uses of ast_dummy_channel_alloc() not checking the returned channel pointer for failure. * Fixed some potential chan=NULL pointer usage in func_odbc.c. Protected by testing the bogus_chan value. * Fixed needlessly clearing a 1024 char auto array when setting the first char to zero is enough in manager.c:action_getvar(). (closes issue ASTERISK-18613) Reported by: Thomas Arimont Patches: jira_asterisk_18613_v1.8.patch (license #5621) patch uploaded by rmudgett Tested by: Thomas Arimont ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@337975 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Gregory Nietsky	6a0fa4e321	Merged revisions 337902 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r337902 \| irroot \| 2011-09-23 21:18:14 +0200 (Fri, 23 Sep 2011) \| 10 lines Merged revisions 337898 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r337898 \| irroot \| 2011-09-23 21:14:30 +0200 (Fri, 23 Sep 2011) \| 4 lines Spelling fix ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@337910 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Gregory Nietsky	b4d8f26ecd	Merged revisions 337840 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r337840 \| irroot \| 2011-09-23 10:39:22 +0200 (Fri, 23 Sep 2011) \| 17 lines Merged revisions 337839 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r337839 \| irroot \| 2011-09-23 10:34:03 +0200 (Fri, 23 Sep 2011) \| 11 lines Make sure a CDR is on the stack for call in the Queue. Only let update_cdr act on the last CDR in the stack. In some circumstances [Attended transfer to queue] a CDR record is not inserted for this call where it should. (closes issue ASTERISK-18567) Review: https://reviewboard.asterisk.org/r/1266 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@337855 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	e734bccdcd	Merged revisions 337775 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r337775 \| russell \| 2011-09-22 19:45:35 -0500 (Thu, 22 Sep 2011) \| 18 lines Merged revisions 337774 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r337774 \| russell \| 2011-09-22 19:44:19 -0500 (Thu, 22 Sep 2011) \| 11 lines Comment out entries in sample res_pktccops.conf. With these options enabled, they can cause Asterisk to freak out by SYN flooding a network and eating the CPU. Obviously it would be good to fix the code so that this can't happen, but we can at least change the default configuration so it doesn't happen. This was reported downstream to the Fedora issue tracker: https://bugzilla.redhat.com/show_bug.cgi?id=658431 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@337776 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	e39f6bba33	Merged revisions 337721 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r337721 \| rmudgett \| 2011-09-22 16:37:41 -0500 (Thu, 22 Sep 2011) \| 25 lines Merged revisions 337720 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r337720 \| rmudgett \| 2011-09-22 16:29:46 -0500 (Thu, 22 Sep 2011) \| 18 lines Made ISDN not add numbering plan prefix strings to empty numbers. When the Caller-ID is restricted, the expected behavior is for the Caller-ID to be blank. In chan_dahdi, the national prefix is placed onto the Caller-ID number even if it is restricted (empty) causing the Caller-ID to be the national prefix rather than blank. This behavior was lost when sig_pri was extracted from chan_dahdi. * Made not add prefix strings to empty connected line, calling, and ANI number strings. (closes issue ASTERISK-18577) Reported by: Kris Shaw Patches: jira_asterisk_18577_v1.8.patch (license #5621) patch uploaded by rmudgett Tested by: Kris Shaw ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@337722 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Gregory Nietsky	cbde0c2431	Blocked revisions 337433 via svnmerge ........ r337433 \| irroot \| 2011-09-22 08:42:42 +0200 (Thu, 22 Sep 2011) \| 12 lines 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/trunk@337681 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Paul Belanger	d947e72055	Blocked revisions 337640 via svnmerge ........ r337640 \| pabelanger \| 2011-09-22 14:43:35 -0400 (Thu, 22 Sep 2011) \| 5 lines 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/trunk@337641 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	5982bdcb7c	Merged revisions 337595,337597 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r337595 \| jrose \| 2011-09-22 10:35:50 -0500 (Thu, 22 Sep 2011) \| 12 lines 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/ ........ r337597 \| jrose \| 2011-09-22 10:47:05 -0500 (Thu, 22 Sep 2011) \| 10 lines 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/trunk@337600 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Gregory Nietsky	8a74aa9ef9	Merged revisions 337542 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r337542 \| irroot \| 2011-09-22 13:44:22 +0200 (Thu, 22 Sep 2011) \| 14 lines 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/trunk@337543 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Gregory Nietsky	308ec93d64	Merged revisions 337487 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r337487 \| irroot \| 2011-09-22 11:26:26 +0200 (Thu, 22 Sep 2011) \| 16 lines 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/trunk@337488 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Gregory Nietsky	3935595e43	Merged revisions 337431 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r337431 \| irroot \| 2011-09-22 08:29:09 +0200 (Thu, 22 Sep 2011) \| 25 lines 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/trunk@337432 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Tilghman Lesher	90a7ed9901	More silly spacing changes	14 years ago
Tilghman Lesher	4730309675	................	14 years ago
Tilghman Lesher	4310b5ad59	................	14 years ago
Gregory Nietsky	2bb0d456eb	Merged revisions 337263 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r337263 \| irroot \| 2011-09-21 13:15:48 +0200 (Wed, 21 Sep 2011) \| 1 line Whitespace fixup from SRTP patch ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@337283 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Gregory Nietsky	8f10934c18	Merged revisions 337261 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r337261 \| irroot \| 2011-09-21 12:42:06 +0200 (Wed, 21 Sep 2011) \| 10 lines 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/trunk@337262 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Olle Johansson	7b08b2cf53	Merged revisions 337219 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r337219 \| oej \| 2011-09-21 11:32:50 +0200 (Ons, 21 Sep 2011) \| 13 lines 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/trunk@337220 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Olle Johansson	2ae7ae00c8	Merged revisions 337178 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r337178 \| oej \| 2011-09-21 10:51:41 +0200 (Ons, 21 Sep 2011) \| 14 lines 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/trunk@337179 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	e218748ac1	Merged revisions 337120 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r337120 \| mjordan \| 2011-09-20 17:49:36 -0500 (Tue, 20 Sep 2011) \| 28 lines 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/trunk@337124 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	1313c12847	Merged revisions 337119 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r337119 \| rmudgett \| 2011-09-20 17:47:45 -0500 (Tue, 20 Sep 2011) \| 16 lines 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/trunk@337123 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	38a7c68851	Updated 10 merge property. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@337122 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	bbafe3bd2c	Restore branch-10 merge properties. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@337121 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Leif Madsen	6b715d8f5c	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/trunk@337117 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	486b6042f3	Merged revisions 337062 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r337062 \| kmoore \| 2011-09-20 16:05:01 -0500 (Tue, 20 Sep 2011) \| 18 lines 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/trunk@337063 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	7fe331fd59	Merged revisions 337008 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r337008 \| rmudgett \| 2011-09-20 14:12:24 -0500 (Tue, 20 Sep 2011) \| 22 lines 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/trunk@337009 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	b3768f04c3	Merged revisions 336978 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r336978 \| rmudgett \| 2011-09-20 13:14:40 -0500 (Tue, 20 Sep 2011) \| 28 lines 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/trunk@336988 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Gregory Nietsky	8493c46308	Merged revisions 336936 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r336936 \| irroot \| 2011-09-20 18:51:59 +0200 (Tue, 20 Sep 2011) \| 14 lines Allow Setting Auth Tag Bit length Based on invite or config option Update the SIP SRTP API to allow use of 32 or 80 bit taglen. Curently only 80 bit is supported. The outgoing invite will use the taglen of the incoming invite preventing one-way audio. (Closes issue ASTERISK-17895) Review: https://reviewboard.asterisk.org/r/1173/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@336937 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	14d3f891e0	Merged revisions 336878 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r336878 \| russell \| 2011-09-19 20:03:55 -0500 (Mon, 19 Sep 2011) \| 43 lines 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/trunk@336879 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	098efb6641	Merged revisions 336792 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r336792 \| twilson \| 2011-09-19 17:13:34 -0500 (Mon, 19 Sep 2011) \| 9 lines 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/trunk@336837 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Tilghman Lesher	8c06ce6cc9	Merged revisions 336789 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r336789 \| tilghman \| 2011-09-19 16:41:16 -0500 (Mon, 19 Sep 2011) \| 2 lines Ensure substring will not be found in the previous match. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@336790 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Tilghman Lesher	5e7121b44f	Merged revisions 336734 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r336734 \| tilghman \| 2011-09-19 15:29:40 -0500 (Mon, 19 Sep 2011) \| 18 lines 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/trunk@336735 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	364eb56835	Merged revisions 336717 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r336717 \| jrose \| 2011-09-19 15:16:23 -0500 (Mon, 19 Sep 2011) \| 14 lines 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/trunk@336732 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	5c71a502a7	Merged revisions 336659 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r336659 \| rmudgett \| 2011-09-19 13:51:19 -0500 (Mon, 19 Sep 2011) \| 38 lines 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/trunk@336662 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	f2fe72628e	Update merge 10 branch merge propterty. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@336661 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	d414984d3f	Restore 10 branch merge properties. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@336660 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jason Parker	052ed863f3	Remove weird mergeinfo props that make merges annoying sometimes. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@336600 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Leif Madsen	b1b315fcb2	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/trunk@336574 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	0f9330b58c	Merged revisions 336570 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r336570 \| rmudgett \| 2011-09-19 10:32:00 -0500 (Mon, 19 Sep 2011) \| 11 lines 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/trunk@336571 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Olle Johansson	1ec4cb8ea0	Merged revisions 336502 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r336502 \| oej \| 2011-09-19 15:38:53 +0200 (Mån, 19 Sep 2011) \| 12 lines 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/trunk@336505 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Gregory Nietsky	d9306c4087	Merged revisions 336500 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r336500 \| irroot \| 2011-09-19 15:31:50 +0200 (Mon, 19 Sep 2011) \| 19 lines 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/trunk@336503 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Olle Johansson	cab155e437	Merged revisions 336441 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r336441 \| oej \| 2011-09-19 14:15:06 +0200 (Mån, 19 Sep 2011) \| 9 lines 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/trunk@336453 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Olle Johansson	5b4b76d3aa	Merged revisions 336381 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r336381 \| oej \| 2011-09-19 12:05:00 +0200 (Mån, 19 Sep 2011) \| 16 lines 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/trunk@336382 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	46a21ca6d9	Merged revisions 336316 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r336316 \| twilson \| 2011-09-16 17:11:39 -0500 (Fri, 16 Sep 2011) \| 9 lines 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/trunk@336317 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	9223069c6e	Merged revisions 336313 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r336313 \| twilson \| 2011-09-16 17:07:00 -0500 (Fri, 16 Sep 2011) \| 12 lines 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/trunk@336315 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	beae2df26e	Merged revisions 336307 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r336307 \| jrose \| 2011-09-16 16:09:20 -0500 (Fri, 16 Sep 2011) \| 20 lines 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/trunk@336311 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Sean Bright	9112b5c75d	Merged revisions 336235 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r336235 \| seanbright \| 2011-09-16 15:10:39 -0400 (Fri, 16 Sep 2011) \| 9 lines 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/trunk@336236 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago

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