asterisk

Commit Graph

Author	SHA1	Message	Date
Gregory Nietsky	283f63c594	Remove T38 Gateway capability when detaching framehook. SET(FAXOPT(gateway)=no) does not remove the capability when detaching the framehook. small patch to fix this problem. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@338904 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	af0161689f	Merged revisions 338800 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r338800 \| rmudgett \| 2011-09-30 17:05:10 -0500 (Fri, 30 Sep 2011) \| 12 lines Fix segfault in analog_ss_thread() not checking ast_read() for NULL. NOTE: The problem was reported against v1.6.2. It is unlikely to ever happen on v1.8 and above since chan_dahdi.c:analog_ss_thread() is unlikely to be used. The version in sig_analog.c has largely replaced it. (closes issue ASTERISK-18648) Reported by: Stephan Bosch Patches: jira_asterisk_18648_v1.8.patch (license #5621) patch uploaded by rmudgett Tested by: Stephan Bosch ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@338801 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	681d28f83d	Merged revisions 338718 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r338718 \| jrose \| 2011-09-30 13:54:30 -0500 (Fri, 30 Sep 2011) \| 1 line Adds documentation for QueueMemberStatus event generation ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@338719 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	67749ae7bb	Fix formatting of AMI header for SIP show peer. ASTERISK-17486 exposed the problem for AMI parsers. (closes issue ASTERISK-18649) Reported by: Jacek Konieczny Patches: asterisk-sipshowpeer_response_end.patch (license #6298) patch uploaded by Jacek Konieczny ........ Merged revisions 338663 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@338664 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Paul Belanger	c3cbe3478c	Merged revisions 338555 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r338555 \| pabelanger \| 2011-09-29 17:12:21 -0400 (Thu, 29 Sep 2011) \| 2 lines Test modules should depend on the TEST_FRAMEWORK flag ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@338556 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jason Parker	6eb8315095	Merged revisions 338551 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r338551 \| qwell \| 2011-09-29 15:54:13 -0500 (Thu, 29 Sep 2011) \| 1 line Test modules have a support level of core. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@338552 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Leif Madsen	dd73844f72	Merged revisions 338492 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r338492 \| lmadsen \| 2011-09-29 13:31:33 -0500 (Thu, 29 Sep 2011) \| 6 lines Update documentation for SIP_HEADER. The SIP_HEADER function only works on the the initial SIP INVITE. The documentation was updated in trunk, but not in 1.8 or 10, so I'm making them match. (Closes issue ASTERISK-18640) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@338493 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Gregory Nietsky	540b7a38a9	Merged revisions 338416 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r338416 \| irroot \| 2011-09-29 14:13:05 +0200 (Thu, 29 Sep 2011) \| 12 lines The rtptimeout setting is ignored on a per peer basis. Not only is the rtptimeout ignored in some cases but rtpkeepalive and rtpholdtimeout is affected. this commit also removes rtptimeout/rtpholdtimeout on text rtp. (closes issue ASTERISK-18559) Review: https://reviewboard.asterisk.org/r/1452 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@338417 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	5d5dc6650d	Merged revisions 338322 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r338322 \| rmudgett \| 2011-09-28 17:35:52 -0500 (Wed, 28 Sep 2011) \| 5 lines Make duplicate call ptr warning message more helpful. * Adds the value of the call ptr to the duplicate call ptr message to help trace why there is a duplicate call ptr. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@338323 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	58ddd59548	Merged revisions 338235 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r338235 \| rmudgett \| 2011-09-28 16:17:45 -0500 (Wed, 28 Sep 2011) \| 7 lines Fix inconsistency in LOG_VERBOSE/AST_LOG_VERBOSE declaration. (closes issue ASTERISK-17973) Reported by: Luke H Patches: logger_h.patch (license #6278) patch uploaded by Luke H ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@338253 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jason Parker	351d8edd83	Merged revisions 338227 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r338227 \| qwell \| 2011-09-28 15:52:47 -0500 (Wed, 28 Sep 2011) \| 1 line Add support levels to non-module sections of menuselect (cflags, utils, etc). ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@338228 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	0e77f0fd5e	Merged revisions 338224 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r338224 \| rmudgett \| 2011-09-28 15:24:41 -0500 (Wed, 28 Sep 2011) \| 5 lines Fix chan_dahd compiling with gcc 4.6 when PRI and SS7 not present. (closes issue ASTERISK-18357) Reported by: Matthew Nicholson ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@338225 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Paul Belanger	80a5b370ff	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/branches/10@338085 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	0764556d4f	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/branches/10@337974 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Gregory Nietsky	e4b65a77b1	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/branches/10@337902 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Gregory Nietsky	876cf3f78e	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/branches/10@337840 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	c62350dd70	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/branches/10@337775 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	d95dcd14e3	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/branches/10@337721 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
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

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