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2011-09-29Merged revisions 338552 via svnmerge from Jason Parker
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r338552 | qwell | 2011-09-29 15:54:55 -0500 (Thu, 29 Sep 2011) | 9 lines 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/trunk@338553 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-29Blocked revisions 338493 via svnmergeLeif Madsen
................ r338493 | lmadsen | 2011-09-29 13:32:28 -0500 (Thu, 29 Sep 2011) | 14 lines 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/trunk@338494 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-29Merged revisions 338417 via svnmerge from Gregory Nietsky
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r338417 | irroot | 2011-09-29 14:16:42 +0200 (Thu, 29 Sep 2011) | 19 lines 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/trunk@338435 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-29Add CLI command "cdr show pgsql status" based on "cdr mysql status"Olle Johansson
Review: https://reviewboard.asterisk.org/r/923/ Thanks all for the code reviews and feedback. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@338415 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-29Just formatting.Olle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@338377 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-28Merged revisions 338323 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r338323 | rmudgett | 2011-09-28 17:36:57 -0500 (Wed, 28 Sep 2011) | 12 lines 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/trunk@338324 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-28Merged revisions 338253 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r338253 | rmudgett | 2011-09-28 16:22:05 -0500 (Wed, 28 Sep 2011) | 14 lines 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/trunk@338284 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-28Merged revisions 338228 via svnmerge from Jason Parker
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r338228 | qwell | 2011-09-28 15:54:35 -0500 (Wed, 28 Sep 2011) | 9 lines 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/trunk@338229 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-28Merged revisions 338225 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r338225 | rmudgett | 2011-09-28 15:26:39 -0500 (Wed, 28 Sep 2011) | 12 lines 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/trunk@338226 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-28Update CHANGES to reflect autopausebusy not being in Asterisk 10Terry Wilson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@338188 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-28Add autopausebusy and autopauseunavail queue optionsTerry Wilson
Make it possible to autopause on a busy or unavailable response from a device. (closes issue ASTERISK-16112) Reported by: jlpedrosa Patches: autopausebusy.txt by twilson Review: https://reviewboard.asterisk.org/r/1399/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@338187 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-28Updated for checking OSP Toolkit version 4.0.0.TransNexus OSP Development
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@338139 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-28Updated for OSP Toolkit 4.0.0.TransNexus OSP Development
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@338136 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-27Merged revisions 338085 via svnmerge from Paul Belanger
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
2011-09-27Whitespace (red blobs) fixesOlle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@338042 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-26Merged revisions 337974 via svnmerge from Richard Mudgett
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
2011-09-23Merged revisions 337902 via svnmerge from Gregory Nietsky
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
2011-09-23Merged revisions 337840 via svnmerge from Gregory Nietsky
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
2011-09-23Merged revisions 337775 via svnmerge from Russell Bryant
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
2011-09-22Merged revisions 337721 via svnmerge from Richard Mudgett
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
2011-09-22Blocked revisions 337433 via svnmergeGregory Nietsky
........ 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
2011-09-22Blocked revisions 337640 via svnmergePaul Belanger
........ 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
2011-09-22Merged revisions 337595,337597 via svnmerge from Jonathan Rose
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
2011-09-22Merged revisions 337542 via svnmerge from Gregory Nietsky
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
2011-09-22Merged revisions 337487 via svnmerge from Gregory Nietsky
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
2011-09-22Merged revisions 337431 via svnmerge from Gregory Nietsky
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
2011-09-21More silly spacing changesTilghman Lesher
..... Merged revisions 337353 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ..... Merged revisions 337380 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@337385 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-21................Tilghman Lesher
........ Dumb little spacing fix. ........ Merged revisions 337344 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ Merged revisions 337345 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@337346 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-21................Tilghman Lesher
........ Escape commas in keys and values, when keys and values are enumerated by commas. Review: https://reviewboard.asterisk.org/r/1433 ........ Merged revisions 337325 from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ Merged revisions 337342 from https://origsvn.digium.com/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@337343 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-21Merged revisions 337263 via svnmerge from Gregory Nietsky
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
2011-09-21Merged revisions 337261 via svnmerge from Gregory Nietsky
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
2011-09-21Merged revisions 337219 via svnmerge from Olle Johansson
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
2011-09-21Merged revisions 337178 via svnmerge from Olle Johansson
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
2011-09-20Merged revisions 337120 via svnmerge from Matthew Jordan
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
2011-09-20Merged revisions 337119 via svnmerge from Richard Mudgett
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
2011-09-20Updated 10 merge property.Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@337122 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-20Restore branch-10 merge properties.Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@337121 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-20Merged revisions 337115 via svnmerge from Leif Madsen
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
2011-09-20Merged revisions 337062 via svnmerge from Kinsey Moore
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
2011-09-20Merged revisions 337008 via svnmerge from Richard Mudgett
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
2011-09-20Merged revisions 336978 via svnmerge from Richard Mudgett
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
2011-09-20Merged revisions 336936 via svnmerge from Gregory Nietsky
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
2011-09-20Merged revisions 336878 via svnmerge from Russell Bryant
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
2011-09-19Merged revisions 336792 via svnmerge from Terry Wilson
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
2011-09-19Merged revisions 336789 via svnmerge from Tilghman Lesher
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
2011-09-19Merged revisions 336734 via svnmerge from Tilghman Lesher
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
2011-09-19Merged revisions 336717 via svnmerge from Jonathan Rose
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
2011-09-19Merged revisions 336659 via svnmerge from Richard Mudgett
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
2011-09-19Update merge 10 branch merge propterty.Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@336661 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-19Restore 10 branch merge properties.Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@336660 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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