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2011-01-19Clarify a source comment about configuration template categories.Sean Bright
(closes issue #18578) Reported by: astmiv Patches: asterisk.main.config.2.patch uploaded by astmiv (license 1189) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@302836 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-19Merged revisions 302789 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r302789 | russell | 2011-01-19 17:06:46 -0600 (Wed, 19 Jan 2011) | 11 lines Merged revisions 302788 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r302788 | russell | 2011-01-19 17:06:14 -0600 (Wed, 19 Jan 2011) | 4 lines Turn a noisy verbose message into a debug message. This can drown your console if you're using the AMI over HTTP. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@302790 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-19Merged revisions 302785 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r302785 | russell | 2011-01-19 16:35:15 -0600 (Wed, 19 Jan 2011) | 15 lines Resolve a memory leak with the manager interface is disabled. The intent of this check as it stands in previous versions of Asterisk was to check if there are any active sessions. If there were no sessions, then the function would return immediately and not bother with queueing up the manager event to be processed. Since the conversion of storing sessions in an astobj2 container, this check will always pass. I changed it to go back to checking what was intended. The side effect of this was that if the AMI is disabled, the manager event queue is populated anyway, but the code that runs to clear out the queue never runs. A producer with no consumer is a bad thing. Reported internally by kmorgan. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@302786 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-19Merged revisions 302713 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r302713 | rmudgett | 2011-01-19 15:29:22 -0600 (Wed, 19 Jan 2011) | 29 lines Merged revisions 302693 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r302693 | rmudgett | 2011-01-19 15:25:41 -0600 (Wed, 19 Jan 2011) | 22 lines Merged revisions 302671 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r302671 | rmudgett | 2011-01-19 15:21:56 -0600 (Wed, 19 Jan 2011) | 15 lines DTMF transfer plays the wrong sounds for wrong number or other call failure. * Set the default for features.conf.sample xferfailsound option to "beeperr" as documented instead of "pbx-invalid" and corrected the use of it in DTMF blind transfer (#1). * Improved DTMF blind transfer handling of wrong numbers. Most of the concerns in this issue were taken care of by the patch for issue 17999: Issues with DTMF triggered attended transfers. (closes issue #18379) Reported by: gincantalupo Tested by: rmudgett ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@302732 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-19Merged revisions 302634 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r302634 | tilghman | 2011-01-19 14:24:57 -0600 (Wed, 19 Jan 2011) | 22 lines Merged revisions 302599 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r302599 | tilghman | 2011-01-19 14:13:24 -0600 (Wed, 19 Jan 2011) | 15 lines Kill zombies. When we ast_safe_fork() with a non-zero argument, we're expected to reap our own zombies. On a zero argument, however, the zombies are only reaped when there aren't any non-zero forked children alive. At other times, we accumulate zombies. This code is forward ported from res_agi in 1.4, so that forked children are always reaped, thus preventing an accumulation of zombie processes. (closes issue #18515) Reported by: ernied Patches: 20101221__issue18515.diff.txt uploaded by tilghman (license 14) Tested by: ernied ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@302644 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-19Merged revisions 302555 via svnmerge from Sean Bright
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r302555 | seanbright | 2011-01-19 14:03:32 -0500 (Wed, 19 Jan 2011) | 14 lines Merged revisions 302554 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r302554 | seanbright | 2011-01-19 14:02:29 -0500 (Wed, 19 Jan 2011) | 7 lines Don't call strlen() when we only need to look at the next character or two. (closes issue #18042) Reported by: wdoekes Patches: astsvn-inefficient-ast-uri-decode.patch uploaded by wdoekes (license 717) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@302556 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-19Merged revisions 302552 via svnmerge from Sean Bright
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r302552 | seanbright | 2011-01-19 13:54:47 -0500 (Wed, 19 Jan 2011) | 14 lines Merged revisions 302551 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r302551 | seanbright | 2011-01-19 13:54:03 -0500 (Wed, 19 Jan 2011) | 7 lines Remove an extraneous \r\n at the end of a parking manager events. (closes issue #18363) Reported by: clegall_proformatique Patches: asterisk_1.8_295998_parking_manager_events_format.patch uploaded by clegall proformatique (license 1139) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@302553 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-19Merged revisions 302505 via svnmerge from Sean Bright
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r302505 | seanbright | 2011-01-19 12:58:11 -0500 (Wed, 19 Jan 2011) | 14 lines Merged revisions 302504 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r302504 | seanbright | 2011-01-19 12:56:32 -0500 (Wed, 19 Jan 2011) | 7 lines Make sure that h_length is set when we short-circuit out of ast_gethostbyname. (closes issue #16135) Reported by: thedavidfactor Patches: utils.patch uploaded by thedavidfactor (license 903) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@302507 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-18Merged revisions 302318 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r302318 | rmudgett | 2011-01-18 16:04:14 -0600 (Tue, 18 Jan 2011) | 1 line Use the expanded format type instead of plain int. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@302319 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-18Merged revisions 302266 via svnmerge from Jeff Peeler
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r302266 | jpeeler | 2011-01-18 14:19:57 -0600 (Tue, 18 Jan 2011) | 34 lines Merged revisions 302265 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r302265 | jpeeler | 2011-01-18 14:13:52 -0600 (Tue, 18 Jan 2011) | 27 lines Convert device state callbacks to ao2 objects to fix a deadlock in chan_sip. Lock scenario presented here: Thread 1 holds ast_rdlock_contexts &conlock holds handle_statechange hints holds handle_statechange hint waiting for cb_extensionstate Locked Here: chan_sip.c line 7428 (find_call) Thread 2 holds handle_request_do &netlock holds find_call sip_pvt_ptr waiting for ast_rdlock_contexts &conlock Locked Here: pbx.c line 9911 (ast_rdlock_contexts) Chan_sip has an established locking order of locking the sip_pvt and then getting the context lock. So the as stated by the summary, the operations in thread 2 have been modified to no longer require the context lock. (closes issue #18310) Reported by: one47 Patches: statecbs_ao2.mk2.patch uploaded by one47 (license 23), modified by me Review: https://reviewboard.asterisk.org/r/1072/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@302270 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-18Merged revisions 302267 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r302267 | russell | 2011-01-18 14:19:57 -0600 (Tue, 18 Jan 2011) | 5 lines Don't enable AO2_DEBUG by default if AST_DEVMODE is on. AO2_DEBUG is not important and is causing a false compiler warning to be generated on my Ubuntu Natty dev box. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@302268 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-18Merged revisions 302174 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r302174 | rmudgett | 2011-01-18 12:11:43 -0600 (Tue, 18 Jan 2011) | 102 lines Merged revisions 302173 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r302173 | rmudgett | 2011-01-18 12:07:15 -0600 (Tue, 18 Jan 2011) | 95 lines Merged revisions 302172 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r302172 | rmudgett | 2011-01-18 12:04:36 -0600 (Tue, 18 Jan 2011) | 88 lines Issues with DTMF triggered attended transfers. Issue #17999 1) A calls B. B answers. 2) B using DTMF dial *2 (code in features.conf for attended transfer). 3) A hears MOH. B dial number C 4) C ringing. A hears MOH. 5) B hangup. A still hears MOH. C ringing. 6) A hangup. C still ringing until "atxfernoanswertimeout" expires. For v1.4 C will ring forever until C answers the dead line. (Issue #17096) Problem: When A and B hangup, C is still ringing. Issue #18395 SIP call limit of B is 1 1. A call B, B answered 2. B *2(atxfer) call C 3. B hangup, C ringing 4. Timeout waiting for C to answer 5. Recall to B fails because B has reached its call limit. Because B reached its call limit, it cannot do anything until the transfer it started completes. Issue #17273 Same scenario as issue 18395 but party B is an FXS port. Party B cannot do anything until the transfer it started completes. If B goes back off hook before C answers, B hears ringback instead of the expected dialtone. ********** Note for the issue #17273 and #18395 fix: DTMF attended transfer works within the channel bridge. Unfortunately, when either party A or B in the channel bridge hangs up, that channel is not completely hung up until the transfer completes. This is a real problem depending upon the channel technology involved. For chan_dahdi, the channel is crippled until the hangup is complete. Either the channel is not useable (analog) or the protocol disconnect messages are held up (PRI/BRI/SS7) and the media is not released. For chan_sip, a call limit of one is going to block that endpoint from any further calls until the hangup is complete. For party A this is a minor problem. The party A channel will only be in this condition while party B is dialing and when party B and C are conferring. The conversation between party B and C is expected to be a short one. Party B is either asking a question of party C or announcing party A. Also party A does not have much incentive to hangup at this point. For party B this can be a major problem during a blonde transfer. (A blonde transfer is our term for an attended transfer that is converted into a blind transfer. :)) Party B could be the operator. When party B hangs up, he assumes that he is out of the original call entirely. The party B channel will be in this condition while party C is ringing, while attempting to recall party B, and while waiting between call attempts. WARNING: The ATXFER_NULL_TECH conditional is a hack to fix the problem. It will replace the party B channel technology with a NULL channel driver to complete hanging up the party B channel technology. The consequences of this code is that the 'h' extension will not be able to access any channel technology specific information like SIP statistics for the call. ATXFER_NULL_TECH is not defined by default. ********** (closes issue #17999) Reported by: iskatel Tested by: rmudgett JIRA SWP-2246 (closes issue #17096) Reported by: gelo Tested by: rmudgett JIRA SWP-1192 (closes issue #18395) Reported by: shihchuan Tested by: rmudgett (closes issue #17273) Reported by: grecco Tested by: rmudgett Review: https://reviewboard.asterisk.org/r/1047/ ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@302178 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-13Add dialplan variables for asterisk.conf directoriesPaul Belanger
Review: https://reviewboard.asterisk.org/r/1075/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@301729 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-12Merged revisions 301595 via svnmerge from Matthew Nicholson
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r301595 | mnicholson | 2011-01-12 12:51:37 -0600 (Wed, 12 Jan 2011) | 22 lines Merged revisions 301594 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r301594 | mnicholson | 2011-01-12 12:50:31 -0600 (Wed, 12 Jan 2011) | 15 lines Removed a usleep(1) that shouldn't be necessary in session_do, and removed the ms_t member from the mansession_session structure. Merged revisions 301591 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r301591 | mnicholson | 2011-01-12 12:39:03 -0600 (Wed, 12 Jan 2011) | 5 lines Don't store the thread id for the manager session in the structure we pass to the thread for the manager session. ABE-2543 ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@301596 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-12Merged revisions 301504 via svnmerge from Jeff Peeler
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r301504 | jpeeler | 2011-01-12 12:12:08 -0600 (Wed, 12 Jan 2011) | 26 lines Merged revisions 301503 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r301503 | jpeeler | 2011-01-12 12:11:49 -0600 (Wed, 12 Jan 2011) | 19 lines Merged revisions 301502 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r301502 | jpeeler | 2011-01-12 12:10:42 -0600 (Wed, 12 Jan 2011) | 12 lines Fix CPU spike when pressing DTMF after agent login. The problem here is that DTMF was being continuously deferred and requeued since ast_safe_sleep is called in a loop. There are serveral other places in the code that sleeps and then loops in a similar fashion. Because of this fact I opted to not defer DTMF any more, which will not affect the original fix: https://reviewboard.asterisk.org/r/674 (closes issue #18130) Reported by: rgj ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@301505 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-12Merged revisions 301446 via svnmerge from David Vossel
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r301446 | dvossel | 2011-01-12 10:05:12 -0600 (Wed, 12 Jan 2011) | 2 lines Removal of unused variables so Asterisk will compile. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@301447 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-12Merged revisions 301402 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r301402 | tilghman | 2011-01-11 18:26:39 -0600 (Tue, 11 Jan 2011) | 7 lines Call execl() directly for a better solution for paths with spaces. (closes issue #18600) Reported by: ebroad Patches: 20110111__issue18600__2.diff.txt uploaded by tilghman (license 14) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@301403 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-11Merged revisions 301308 via svnmerge from Matthew Nicholson
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r301308 | mnicholson | 2011-01-11 12:51:40 -0600 (Tue, 11 Jan 2011) | 18 lines Merged revisions 301307 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r301307 | mnicholson | 2011-01-11 12:42:05 -0600 (Tue, 11 Jan 2011) | 11 lines Merged revisions 301305 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r301305 | mnicholson | 2011-01-11 12:34:40 -0600 (Tue, 11 Jan 2011) | 4 lines Prevent buffer overflows in ast_uri_encode() ABE-2705 ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@301309 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-10Merged revisions 301263 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r301263 | tilghman | 2011-01-10 16:39:31 -0600 (Mon, 10 Jan 2011) | 8 lines Little endian machines were not converted properly. (closes issue #18583) Reported by: jcovert Patches: 20110110__issue18583.diff.txt uploaded by tilghman (license 14) Tested by: jcovert ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@301264 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-04Merged revisions 300521 via svnmerge from Leif Madsen
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r300521 | lmadsen | 2011-01-04 15:53:27 -0600 (Tue, 04 Jan 2011) | 17 lines Merged revisions 300520 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r300520 | lmadsen | 2011-01-04 15:52:41 -0600 (Tue, 04 Jan 2011) | 9 lines Fix backwards and broken XML documentation. (closes issue #18547) Reported by: jcovert Patches: xmldoc.c.patch uploaded by jcovert (license 551) chan_iax2.c.doc.patch uploaded by jcovert (license 551) chan_sip.c.patch uploaded by jcovert (license 551) chan_agent.c.patch uploaded by jcovert (license 551) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@300522 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-03Merged revisions 300166 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r300166 | rmudgett | 2011-01-03 17:14:55 -0600 (Mon, 03 Jan 2011) | 11 lines Merged revisions 300165 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r300165 | rmudgett | 2011-01-03 17:02:13 -0600 (Mon, 03 Jan 2011) | 4 lines Use correct variable for atxfercallbackretries config option. * Misc formatting changes. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@300168 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-12-31Support an alternate configuration file for the 'logger reload' command.Tilghman Lesher
(closes issue #17668) Reported by: tilghman Patches: 20100718__logger_reload_altconf__2.diff.txt uploaded by tilghman (license 14) Review: (by lmadsen, russell within comments on issue tracker) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@300044 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-12-29Remove some trailing whitespace and steal revision 300000.Sean Bright
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@300000 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-12-29Merged revisions 299989 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r299989 | tilghman | 2010-12-29 16:02:59 -0600 (Wed, 29 Dec 2010) | 4 lines Quote arguments, just in case there's a space in a pathname. (Diagnosed by pabelanger on #asterisk-dev, fixed by me.) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@299990 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-12-20New astobj2 flag for issuing a callback without locking the container.David Vossel
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@299135 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-12-20Some scheduler API cleanup and improvements.Russell Bryant
Previously, I had added the ast_sched_thread stuff that was a generic scheduler thread implementation. However, if you used it, it required using different functions for modifying scheduler contents. This patch reworks how this is done and just allows you to optionally start a thread on the original scheduler context structure that has always been there. This makes it trivial to switch to the generic scheduler thread implementation without having to touch any of the other code that adds or removes scheduler entries. In passing, I made some naming tweaks to add ast_ prefixes where they were not there before. Review: https://reviewboard.asterisk.org/r/1007/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@299091 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-12-20Merged revisions 299088 via svnmerge from Leif Madsen
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r299088 | lmadsen | 2010-12-20 10:18:26 -0600 (Mon, 20 Dec 2010) | 13 lines Merged revisions 299087 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r299087 | lmadsen | 2010-12-20 10:18:03 -0600 (Mon, 20 Dec 2010) | 5 lines Note that Park() timeout is milliseconds. (closes issue #15758) Reported by: mmurdock Tested by: mmurdock, seanbright ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@299089 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-12-20Typos: recieved => receivedTzafrir Cohen
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@299005 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-12-18Merged revisions 298960 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r298960 | tilghman | 2010-12-17 17:52:04 -0600 (Fri, 17 Dec 2010) | 20 lines Merged revisions 298957 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r298957 | tilghman | 2010-12-17 17:30:55 -0600 (Fri, 17 Dec 2010) | 13 lines Merged revisions 298905 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r298905 | tilghman | 2010-12-17 15:40:56 -0600 (Fri, 17 Dec 2010) | 6 lines Let Asterisk find better backtrace information with libbfd. The menuselect option BETTER_BACKTRACES, if enabled, will use libbfd to search for better symbol information within both the Asterisk binary, as well as loaded modules, to assist when using inline backtraces to track down problems. Review: https://reviewboard.asterisk.org/r/1055/ ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@298961 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-12-12Add support for several platforms to obtain the real thread ID.Jeff Peeler
Already had the pthread ID which is not the same. The most obvious enhancement is in the "core show threads" output. As stated in the utils header, if the platform isn't supported -1 is reported (instead of the process ID previously). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@298137 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-12-10Merged revisions 298051 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r298051 | tilghman | 2010-12-10 10:26:46 -0600 (Fri, 10 Dec 2010) | 18 lines Merged revisions 298050 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r298050 | tilghman | 2010-12-10 10:24:13 -0600 (Fri, 10 Dec 2010) | 11 lines Portability issue on OpenSolaris. Also detect the required structure element, because OpenSolaris defines SIOCGIFHWADDR, but without support for IP sockets. (closes issue #18442) Reported by: ranjtech Patches: 20101209__issue18442.diff.txt uploaded by tilghman (license 14) Tested by: ranjtech ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@298052 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-12-09Merged revisions 297952 via svnmerge from Terry Wilson
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r297952 | twilson | 2010-12-09 14:48:44 -0600 (Thu, 09 Dec 2010) | 10 lines Don't crash after Set(CDR(userfield)=...) in ast_bridge_call Instead of setting peer->cdr = NULL, set it to not post. (closes issue #18415) Reported by: macbrody Patches: patch-18415 uploaded by jsolares (license 1167) Tested by: jsolares, twilson ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@297956 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-12-07Merged revisions 297825 via svnmerge from Jeff Peeler
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r297825 | jpeeler | 2010-12-07 16:59:30 -0600 (Tue, 07 Dec 2010) | 26 lines Merged revisions 297824 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r297824 | jpeeler | 2010-12-07 16:58:54 -0600 (Tue, 07 Dec 2010) | 19 lines Merged revisions 297823 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r297823 | jpeeler | 2010-12-07 16:57:48 -0600 (Tue, 07 Dec 2010) | 12 lines Revert code that changed SSRC for DTMF. Some previous behavior was attempted to be restored, but mistakingly I did not realize that the previous behavior was incorrect. This fixes DTMF not being detected since DTMF shouldn't cause the SSRC to change. (related to issue #17404) (closes issue #18189) (closes issue #18352) Reported by: marcbou Tested by: cmbaker82 ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@297826 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-12-02Merged revisions 297312 via svnmerge from Terry Wilson
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r297312 | twilson | 2010-12-02 12:13:49 -0600 (Thu, 02 Dec 2010) | 28 lines Merged revisions 297311 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r297311 | twilson | 2010-12-02 12:07:39 -0600 (Thu, 02 Dec 2010) | 21 lines Merged revisions 297310 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r297310 | twilson | 2010-12-02 12:00:27 -0600 (Thu, 02 Dec 2010) | 12 lines Initialize offset for adaptive jitter buffer When the adaptive jitter buffer is enabled in sip.conf, the first frame placed in the jitter buffer fails with something like: jb_warning_output: Resyncing the jb. last_delay 0, this delay -215886466, threshold 1000, new offset 215886466 This happens because the offset is not initialized before calling jb_put(). This patch modifies jb_put_first_adaptive() to set the offset to the frame's timestamp. Review: https://reviewboard.asterisk.org/r/1041/ ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@297356 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-30move devices from hints into an ao2_containerStefan Schmidt
by splitting up devices from hints into an own ao2_container the callback to get these devices for statechange handling is faster. with this changes the length of a device used in a hint isnt longer restricted to 80 characters. Tests showed that calling handle_statechange is 40 times faster if no hints are used and 25 times faster if there are any hints. (closes issue #17928) Reported by: mdu113 Tested by: schmidts Review: https://reviewboard.asterisk.org/r/1003/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@296752 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-29Merged revisions 296534 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r296534 | tilghman | 2010-11-29 01:28:44 -0600 (Mon, 29 Nov 2010) | 20 lines Merged revisions 296533 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r296533 | tilghman | 2010-11-29 01:27:09 -0600 (Mon, 29 Nov 2010) | 13 lines I love standards. There are so many to choose from. Except when there isn't one. Linux and *BSD disagree on the elements within the ucred structure. Detect which one is in use on the system. (closes issue #18384) Reported by: bjm Patches: cred-diffs uploaded by bjm (license 473) 20101127__issue18384__1.6.2.diff.txt uploaded by tilghman (license 14) 20101127__issue18384__1.8.diff.txt uploaded by tilghman (license 14) Tested by: tilghman, bjm ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@296535 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-26Merged revisions 296391 via svnmerge from Olle Johansson
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r296391 | oej | 2010-11-26 22:37:21 +0100 (Fre, 26 Nov 2010) | 24 lines Merged revisions 296351 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r296351 | oej | 2010-11-26 13:23:03 +0100 (Fre, 26 Nov 2010) | 17 lines Merged revisions 296309 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r296309 | oej | 2010-11-26 10:53:31 +0100 (Fre, 26 Nov 2010) | 11 lines Fix bugs in saying numbers using the Swedish language syntax (closes issue #18355) Reported by: oej Patch by: oej Much help from Peter Lindahl. Testing by the ClearIT team during a coffee break. Review: https://reviewboard.asterisk.org/r/1033/ ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@296393 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-24Merged revisions 296230 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r296230 | russell | 2010-11-24 17:29:44 -0600 (Wed, 24 Nov 2010) | 20 lines Merged revisions 296221 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r296221 | russell | 2010-11-24 17:28:19 -0600 (Wed, 24 Nov 2010) | 13 lines Merged revisions 296213 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r296213 | russell | 2010-11-24 17:26:43 -0600 (Wed, 24 Nov 2010) | 6 lines Make Asterisk less crashy. Since we might not put a new translation path on the channel, go ahead and set it to NULL right after destroying the old one to ensure we don't try to free an invalid translation path later on. ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@296235 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-24Merged revisions 296084 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r296084 | russell | 2010-11-24 14:23:46 -0600 (Wed, 24 Nov 2010) | 26 lines Merged revisions 296083 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r296083 | russell | 2010-11-24 14:23:11 -0600 (Wed, 24 Nov 2010) | 19 lines Merged revisions 296082 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r296082 | russell | 2010-11-24 14:22:32 -0600 (Wed, 24 Nov 2010) | 12 lines Fix false reporting of an error by set_format(). In the case that the native format was able to be changed to match the new requested format, the code proceeded to attempt to build a translation path, anyway. The result would be NULL, since no translation path is necessary and resulted in this function thinking an error has occurred. This case is now specifically caught and no attempt to build a translation path is attempted. Thanks to our automated tests and bamboo.asterisk.org for catching this problem and making a whole lot of noise when things started failing. :-) ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@296085 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-24Merged revisions 296002 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r296002 | russell | 2010-11-24 11:13:08 -0600 (Wed, 24 Nov 2010) | 52 lines Merged revisions 296001 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r296001 | russell | 2010-11-24 11:03:16 -0600 (Wed, 24 Nov 2010) | 45 lines Merged revisions 296000 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r296000 | russell | 2010-11-24 10:48:39 -0600 (Wed, 24 Nov 2010) | 38 lines Handle failures building translation paths more effectively. The problem scenario occurred on a heavily loaded system that was using the codec_dahdi module and exceeded the hardware transcoding capacity. The failure mode at that point was not good. The report came in to us as an Asterisk lock-up. The "core show locks" shows a ton of threads locked up (but no obvious deadlock). Upon deeper investigation, when the system is in this state, the CPU was maxed out. The CPU was being consumed by the Asterisk logger spewing messages on every audio frame for calls set up after transcoder capacity was reached. The purpose of this patch is to make Asterisk handle failures to create a translation path in a more graceful manner. If we can't translate, then the call just needs to be dropped, as it's not going to work. These are the changes: 1) In set_format() of channel.c (which is called by set_read_format() and set_write_format()), it was ignoring if ast_translator_build_path() failed and returned NULL. It now pays attention to that case and returns a result reflecting failure. With this change in place, the bridging code will immediately detect a failure and end the bridge instead of proceeding to try to bridge frames that can't be translated and making channel drivers freak out by sending them frames in a format they weren't expecting. 2) In ast_indicate_data() of channel.c, failure of ast_playtones_start() was ignored. It is now reflected in the return value of the function. This didn't turn out to have any affect on the bug, but seemed like a good change to leave in. 3) In app_dial(), when only sending a call to a single endpoint, it will attempt to do some bridging of its own of early audio. It uses make_compatible() when it's going to do this. However, it ignored failure from make compatible. So, even with the fix from #1, if there was early audio going through app_dial, there would still be a period of invalid frames passing through. After detecting failure here, Dial() exits. ABE-2658 ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@296034 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-23Merged revisions 295949 via svnmerge from Olle Johansson
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r295949 | oej | 2010-11-23 11:30:05 +0100 (Tis, 23 Nov 2010) | 21 lines Merged revisions 295907 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r295907 | oej | 2010-11-23 10:36:38 +0100 (Tis, 23 Nov 2010) | 14 lines Merged revisions 295906 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r295906 | oej | 2010-11-23 10:28:14 +0100 (Tis, 23 Nov 2010) | 8 lines Fix support of saynumber(1,n) in the Swedish language (closes issue #18353) Reported by: oej Review: https://reviewboard.asterisk.org/r/1031/ ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@295950 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-22Merged revisions 295866 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r295866 | rmudgett | 2010-11-22 13:36:10 -0600 (Mon, 22 Nov 2010) | 60 lines Merged revisions 295843 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r295843 | rmudgett | 2010-11-22 13:28:23 -0600 (Mon, 22 Nov 2010) | 53 lines Merged revisions 295790 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r295790 | rmudgett | 2010-11-22 12:46:26 -0600 (Mon, 22 Nov 2010) | 46 lines The channel redirect function (CLI or AMI) hangs up the call instead of redirecting the call. To recreate the problem: 1) Party A calls Party B 2) Invoke CLI "channel redirect" command to redirect channel call leg associated with A. 3) All associated channels are hung up. Note that if the CLI command were done on the channel call leg associated with B it works. This regression was a result of the fix for issue #16946 (https://reviewboard.asterisk.org/r/740/). The regression affects all features that use an async goto to execute the dialplan because of an external event: Channel redirect, AMI redirect, SIP REFER, and FAX detection. The struct ast_channel._softhangup code is a mess. The variable is used for several purposes that do not necessarily result in the call being hung up. I have added doxygen comments to describe how the various _softhangup bits are used. I have corrected all the places where the variable was tested in a non-bit oriented manner. The primary fix is the new AST_CONTROL_END_OF_Q frame. It acts as a weak hangup request so the soft hangup requests that do not normally result in a hangup do not hangup. JIRA SWP-2470 JIRA SWP-2489 (closes issue #18171) Reported by: SantaFox (closes issue #18185) Reported by: kwemheuer (closes issue #18211) Reported by: zahir_koradia (closes issue #18230) Reported by: vmarrone (closes issue #18299) Reported by: mbrevda (closes issue #18322) Reported by: nerbos Review: https://reviewboard.asterisk.org/r/1013/ ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@295867 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-20Merged revisions 295711 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r295711 | russell | 2010-11-19 18:50:00 -0600 (Fri, 19 Nov 2010) | 36 lines Merged revisions 295710 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r295710 | russell | 2010-11-19 18:45:51 -0600 (Fri, 19 Nov 2010) | 29 lines Fix cache of device state changes for multiple servers. This patch addresses a regression where device states across multiple servers were not being processing completely correctly. The code works to determine the overall state by looking at the last known state of a device on each server. However, there was a regression due to some invasive rewrites of how the cache works that led to the cache only storing the last device state change for a device, regardless of which server it was on. The code is set up to cache device state change events by ensuring that each event in the cache has a unique device name + entity ID (server ID). The code that was responsible for comparing raw information elements (which EID is) always returned a match due to a memcmp() with a length of 0. There isn't much code to fix the actual bug. This patch also introduces a new CLI command that was very useful for debugging this problem. The command allows you to dump the contents of the event cache. (closes issue #18284) Reported by: klaus3000 Patches: issue18284.rev1.txt uploaded by russell (license 2) Tested by: russell, klaus3000 (closes issue #18280) Reported by: klaus3000 Review: https://reviewboard.asterisk.org/r/1012/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@295712 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-16Merged revisions 295282 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r295282 | rmudgett | 2010-11-16 17:02:36 -0600 (Tue, 16 Nov 2010) | 16 lines Merged revisions 295281 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r295281 | rmudgett | 2010-11-16 16:57:07 -0600 (Tue, 16 Nov 2010) | 9 lines Merged revisions 295280 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r295280 | rmudgett | 2010-11-16 16:52:06 -0600 (Tue, 16 Nov 2010) | 1 line Dead code elimination in channel.c:ast_channel_bridge() variable who. ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@295283 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-16Remove a trailing space.Russell Bryant
(testing something with bamboo ...) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@295125 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-10Merged revisions 294501 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r294501 | russell | 2010-11-10 06:46:27 -0600 (Wed, 10 Nov 2010) | 14 lines Merged revisions 294500 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r294500 | russell | 2010-11-10 06:41:41 -0600 (Wed, 10 Nov 2010) | 7 lines Improve a debug message to be more readable and consistent. (closes issue #18282) Reported by: klaus3000 Patches: ast_devstate2str-patch.txt uploaded by klaus3000 (license 65) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294502 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-09Merged revisions 294466 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294466 | rmudgett | 2010-11-09 16:46:45 -0600 (Tue, 09 Nov 2010) | 1 line Allow ast_do_masquerade() failure to be reported again. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294467 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-09Merged revisions 294349 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294349 | rmudgett | 2010-11-09 10:55:32 -0600 (Tue, 09 Nov 2010) | 17 lines Analog lines do not transfer CONNECTED LINE or execute the interception macros. Add connected line update for sig_analog transfers and simplify the corresponding sig_pri and chan_misdn transfer code. Note that if you create a three-way call in sig_analog before transferring the call, the distinction of the caller/callee interception macros make little sense. The interception macro writer needs to be prepared for either caller/callee macro to be executed. The current implementation swaps which caller/callee interception macro is executed after a three-way call is created. Review: https://reviewboard.asterisk.org/r/996/ JIRA ABE-2589 JIRA SWP-2372 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294351 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-08Merged revisions 294278 via svnmerge from Jeff Peeler
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r294278 | jpeeler | 2010-11-08 15:59:45 -0600 (Mon, 08 Nov 2010) | 23 lines Merged revisions 294277 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r294277 | jpeeler | 2010-11-08 15:58:13 -0600 (Mon, 08 Nov 2010) | 16 lines Fix playback failure when using IAX with the timerfd module. To fix this issue the alert pipe will now be used when the timerfd module is in use. There appeared to be a race that was not solved by adding locking in the timerfd module, but needed to be there anyway. The race was between the timer being put in non-continuous mode in ast_read on the channel thread and the IAX frame scheduler queuing a frame which would enable continuous mode before the non-continuous mode event was read. This race for now is simply avoided. (closes issue #18110) Reported by: tpanton Tested by: tpanton I put tested by tpanton because it was tested on his hardware. Thanks for the remote access to debug this issue! ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294279 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-03Merged revisions 293803 via svnmerge from Terry Wilson
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r293803 | twilson | 2010-11-03 11:05:14 -0700 (Wed, 03 Nov 2010) | 25 lines Avoid valgrind warnings for ast_rtp_instance_get_xxx_address The documentation for ast_rtp_instance_get_(local/remote)_address stated that they returned 0 for success and -1 on failure. Instead, they returned 0 if the address structure passed in was already equivalent to the address instance local/remote address or 1 otherwise. 90% of the calls to these functions completely ignored the return address and passed in an uninitialized struct, which would make valgrind complain even though the operation was technically safe. This patch fixes the documentation and converts the get_xxx_address functions to void since all they really do is copy the address and cannot fail. Additionally two new functions (ast_rtp_instance_get_and_cmp_(local/remote)_address) are created for the 3 times where the return value was actually checked. The get_and_cmp_local_address function is currently unused, but exists for the sake of symmetry. The only functional change as a result of this change is that we will not do an ast_sockaddr_cmp() on (mostly uninitialized) addresses before doing the ast_sockaddr_copy() in the get_*_address functions. So, even though it is an API change, it shouldn't have a noticeable change in behavior. Review: https://reviewboard.asterisk.org/r/995/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@293809 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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