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2011-01-20Merged revisions 302921 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r302921 | russell | 2011-01-20 10:12:15 -0600 (Thu, 20 Jan 2011) | 9 lines Merged revisions 302920 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r302920 | russell | 2011-01-20 10:11:58 -0600 (Thu, 20 Jan 2011) | 2 lines Resolve a compiler warning. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@302922 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-20Merged revisions 302918 via svnmerge from Leif Madsen
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r302918 | lmadsen | 2011-01-20 09:45:39 -0600 (Thu, 20 Jan 2011) | 16 lines Merged revisions 302917 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r302917 | lmadsen | 2011-01-20 09:42:05 -0600 (Thu, 20 Jan 2011) | 8 lines Option L() is milliseconds, not seconds. > Change the verbose output of option L() to say milliseconds and not seconds > as the value is in milliseconds. > > (closes issue #18264) > Reported by: jacco > Patches: > app_dial_patch.txt uploaded by lmadsen (license 10) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@302919 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-20On systems which have LLVM, use that compiler. Should result in a massive ↵Tilghman Lesher
speed increase. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@302879 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-19Merged revisions 302837 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r302837 | russell | 2011-01-19 17:56:48 -0600 (Wed, 19 Jan 2011) | 2 lines Only check container count if it exists. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@302838 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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 302834 via svnmerge from Sean Bright
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r302834 | seanbright | 2011-01-19 18:49:00 -0500 (Wed, 19 Jan 2011) | 14 lines Merged revisions 302833 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r302833 | seanbright | 2011-01-19 18:47:22 -0500 (Wed, 19 Jan 2011) | 7 lines Support greetingsfolder as documented in voicemail.conf.sample. (closes issue #17870) Reported by: edhorton Patches: __20100816-app_voicemail-greetingsfolder-support.txt uploaded by lmadsen (license 10) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@302835 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-19Merged revisions 302831 via svnmerge from Paul Belanger
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r302831 | pabelanger | 2011-01-19 18:29:45 -0500 (Wed, 19 Jan 2011) | 2 lines Add binutils-dev for BETTER_BACKTRACES ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@302832 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 302680 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r302680 | tilghman | 2011-01-19 15:23:31 -0600 (Wed, 19 Jan 2011) | 16 lines Merged revisions 302675 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r302675 | tilghman | 2011-01-19 15:22:45 -0600 (Wed, 19 Jan 2011) | 9 lines Merged revisions 302663 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r302663 | tilghman | 2011-01-19 15:20:28 -0600 (Wed, 19 Jan 2011) | 2 lines Add some API documentation ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@302686 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 302600 via svnmerge from Jason Parker
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r302600 | qwell | 2011-01-19 14:14:40 -0600 (Wed, 19 Jan 2011) | 1 line Fix typo pointed out on asterisk-users list. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@302601 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 302549 via svnmerge from Sean Bright
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r302549 | seanbright | 2011-01-19 13:43:11 -0500 (Wed, 19 Jan 2011) | 17 lines Merged revisions 302548 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r302548 | seanbright | 2011-01-19 13:37:09 -0500 (Wed, 19 Jan 2011) | 10 lines Properly handle partial reads from fgets() when handling AGIs. When fgets() failed with EAGAIN, we were continually decrementing the available space left in our buffer, resulting in botched command handling. (closes issue #16032) Reported by: notahat Patches: agi_buffer_patch2.diff uploaded by fnordian (license 110) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@302550 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
2011-01-19Merged revisions 302462 via svnmerge from Paul Belanger
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r302462 | pabelanger | 2011-01-19 12:09:35 -0500 (Wed, 19 Jan 2011) | 9 lines Merged revisions 302461 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r302461 | pabelanger | 2011-01-19 12:08:01 -0500 (Wed, 19 Jan 2011) | 2 lines Handle 'Resource temporarily unavailable' error more gracefully. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@302463 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-19Merged revisions 302417 via svnmerge from Sean Bright
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r302417 | seanbright | 2011-01-19 10:53:20 -0500 (Wed, 19 Jan 2011) | 16 lines Merged revisions 302416 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r302416 | seanbright | 2011-01-19 10:52:44 -0500 (Wed, 19 Jan 2011) | 9 lines Remove references to priorityjumping from the sample extensions.conf. Priority jumping was removed from pbx_config in r68970. (closes issue #18622) Reported by: kshumard Patches: extensions.conf.sample.patch uploaded by kshumard (license 92) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@302418 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-19Merged revisions 302414 via svnmerge from Sean Bright
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r302414 | seanbright | 2011-01-19 10:45:17 -0500 (Wed, 19 Jan 2011) | 7 lines Initialize an uninitialized variable. (closes issue #18640) Reported by: jcovert Patches: chan_sip.c.patch uploaded by jcovert (license 551) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@302415 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-19Merged revisions 302412 via svnmerge from Sean Bright
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r302412 | seanbright | 2011-01-19 10:31:39 -0500 (Wed, 19 Jan 2011) | 10 lines Use appropriate type for requested format in chan_local. We were passing and storing the requested format as an int instead of format_t resulting in truncation. (closes issue #18238) Reported by: whizemen Patches: 0018238_speex16.patch uploaded by whizemen (license 1143) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@302413 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 302314 via svnmerge from Matthew Nicholson
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r302314 | mnicholson | 2011-01-18 15:43:21 -0600 (Tue, 18 Jan 2011) | 18 lines Merged revisions 302313 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r302313 | mnicholson | 2011-01-18 15:40:03 -0600 (Tue, 18 Jan 2011) | 11 lines Merged revisions 302311 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r302311 | mnicholson | 2011-01-18 15:35:03 -0600 (Tue, 18 Jan 2011) | 4 lines URI encode the user part of the contact header. ABE-2705 ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@302315 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
2011-01-17Merged revisions 293493 via svnmerge from Terry Wilson
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r293493 | twilson | 2010-11-01 09:58:00 -0500 (Mon, 01 Nov 2010) | 14 lines Only offer codecs both sides support for directmedia When using directmedia, Asterisk needs to limit the codecs offered to just the ones that both sides recognize, otherwise they may end up sending audio that the other side doesn't understand. (closes issue #17403) Reported by: one47 Patches: sip_codecs_simplified4 uploaded by one47 (license 23) Tested by: one47, falves11 Review: https://reviewboard.asterisk.org/r/967/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@302048 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-17Merged revisions 302005 via svnmerge from Terry Wilson
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r302005 | twilson | 2011-01-17 09:04:59 -0600 (Mon, 17 Jan 2011) | 2 lines Document "encryption" option in sip.conf.sample ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@302006 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-14Merged revisions 301946 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r301946 | rmudgett | 2011-01-14 15:09:57 -0600 (Fri, 14 Jan 2011) | 13 lines Deadlock between dahdi_request() and pri_dchannel() processing an incomming call. The sig_pri_new_ast_channel() is called with the channel private lock held when pri_dchannel() calls it and no channel private lock held when dahdi_request() calls it. The use of pri_grab() in sig_pri_new_ast_channel() could leave the channel private lock held when it returns if the lock was not held before calling it. Make sig_pri_new_ast_channel() just lock the PRI span lock instead of using pri_grab(). It is safe to do this because dahdi_request() does not have the channel private lock and the deadlock potential with the PRI span lock is only between pri_dchannel() and other threads. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@301947 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-14Merged revisions 301851 via svnmerge from Brett Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r301851 | bbryant | 2011-01-14 15:11:55 -0500 (Fri, 14 Jan 2011) | 6 lines Changing previous revisions 301845/301847 to use ast_sockaddr_setnull() instead of setting the field manually to avoid uninitialized data. Review: https://reviewboard.asterisk.org/r/1076/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@301858 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-14Add relationships to function documentation.Andrew Latham
Fix amatuer type mistake git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@301850 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-14Merged revisions 301845 via svnmerge from Brett Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r301845 | bbryant | 2011-01-14 14:35:23 -0500 (Fri, 14 Jan 2011) | 9 lines Fix for a consistent MulticastRTP channel driver crash due to use of unitilized data. (closes issue #18290) (closes issue #18602) Reported by: voipgate, wybecom Review: https://reviewboard.asterisk.org/r/1076/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@301847 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-14Add relationships to function documentation.Andrew Latham
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@301846 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-14Merged revisions 301790 via svnmerge from Jeff Peeler
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r301790 | jpeeler | 2011-01-14 11:32:52 -0600 (Fri, 14 Jan 2011) | 42 lines Resolve deadlock involving REFER. Two fixes: 1) One must always have the private unlocked before calling pbx_builtin_setvar_helper to not invalidate locking order since it locks the channel. 2) Unlock the channel before calling pbx_find_extension, which starts and stops autoservice during the lookup. The problem scenario as illustrated by the reporter: Thread: do_monitor ----------------------- handle_request_do handle_incoming handle_request_refer ast_parking_ext_valid pbx_find_extension ast_autoservice_stop while (chan_list_state == as_chan_list_state) { usleep(1000); } Thread: autoservice_run ----------------------- autoservice_run chan = ast_waitfor_n ast_waitfor_nandfds ast_waitfor_nandfds_classic / simple / complex (depending on your system) ast_channel_lock(c[x]); handle_request_do and schedule_process_request_queue locks the owner if it exists. The autoservice thread is waiting for the channel lock, which wasn't ever released since the do_monitor thread was waiting for autoservice operations to complete. Solved by unlocking the channel but keeping a reference to guarantee safety. (closes issue #18403) Reported by: jthurman Patches: 20110103-blind_deadlock.diff uploaded by jthurman (license 614) issue18403.patch uploaded by jpeeler (license 325) Tested by: jthurman ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@301791 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-13Merged revisions 301731 via svnmerge from Leif Madsen
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r301731 | lmadsen | 2011-01-13 11:01:43 -0600 (Thu, 13 Jan 2011) | 15 lines Merged revisions 301730 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r301730 | lmadsen | 2011-01-13 11:01:11 -0600 (Thu, 13 Jan 2011) | 7 lines Add static entry for split Polycom 332 firmware. (closes issue #18607) Reported by: cjacobsen Patches: polycom_331.diff uploaded by cjacobsen (license 1029) Tested by: lathama ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@301732 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 301683 via svnmerge from Terry Wilson
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r301683 | twilson | 2011-01-12 15:19:48 -0600 (Wed, 12 Jan 2011) | 15 lines Merged revisions 301682 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r301682 | twilson | 2011-01-12 15:05:02 -0600 (Wed, 12 Jan 2011) | 9 lines Don't reject all SUBSCRIBE auth requests When merging another SUBSCRIBE fix from 1.4, some braces were put in the wrong place. This patch fixes that. (closes issue #18597) Reported by: thsgmbh ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@301684 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
2011-01-12fix wrong text of rerun menuselect after user interface warningStefan Schmidt
the warning, if no user interface for menuselect warning was found is not right. you have to rerun configure before make menuselect after installing a proper user interface. (closes issue 0018594) Reported by: Dovid git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@301445 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 301311 via svnmerge from Paul Belanger
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r301311 | pabelanger | 2011-01-11 14:16:06 -0500 (Tue, 11 Jan 2011) | 9 lines Merged revisions 301310 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r301310 | pabelanger | 2011-01-11 14:14:31 -0500 (Tue, 11 Jan 2011) | 2 lines Fix a logic issue when passing context ARG ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@301319 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
2011-01-09Merged revisions 301221 via svnmerge from Paul Belanger
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r301221 | pabelanger | 2011-01-09 16:40:34 -0500 (Sun, 09 Jan 2011) | 21 lines Merged revisions 301220 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r301220 | pabelanger | 2011-01-09 16:38:24 -0500 (Sun, 09 Jan 2011) | 14 lines SOUND_CACHE_DIR now defaults to empty Sounds files included in the Asterisk tarball were being ignored and re-downloaded. Users wanting to cache the files can still override the setting using the --with-sounds-cache option. (closes issue #18589) Reported by: pabelanger Patches: issue18589.patch uploaded by pabelanger (license 224) Tested by: pabelanger Review: https://reviewboard.asterisk.org/r/1074/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@301222 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-08Merged revisions 301177 via svnmerge from Paul Belanger
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r301177 | pabelanger | 2011-01-08 17:00:12 -0500 (Sat, 08 Jan 2011) | 14 lines Merged revisions 301176 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r301176 | pabelanger | 2011-01-08 16:58:24 -0500 (Sat, 08 Jan 2011) | 7 lines Indicate log level argument for Log() is not optional (closes issue #18586) Reported by: kshumard Patches: app_verbose.c.patch uploaded by kshumard (license 92) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@301178 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-08Merged revisions 301134 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r301134 | rmudgett | 2011-01-07 19:11:31 -0600 (Fri, 07 Jan 2011) | 7 lines The DTMF attended transfer feature cannot callback a chan_dahdi BRI phone. The DAHDI ISDN channel name is not dialable. Make a channel name like DAHDI/i3/400-12 dialable when the sequence number is stripped off of the name. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@301135 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-07Merged revisions 301090 via svnmerge from Jason Parker
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r301090 | qwell | 2011-01-07 14:53:02 -0600 (Fri, 07 Jan 2011) | 15 lines Merged revisions 301089 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r301089 | qwell | 2011-01-07 14:52:00 -0600 (Fri, 07 Jan 2011) | 8 lines Initialize useropts/adminopts in case there is no column in the realtime DB. (closes issue #18182) Reported by: dimas Patches: v1-18182.patch uploaded by dimas (license 88) Tested by: dimas ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@301091 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-07Merged revisions 301047 via svnmerge from Jeff Peeler
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r301047 | jpeeler | 2011-01-07 13:58:30 -0600 (Fri, 07 Jan 2011) | 15 lines Merged revisions 301046 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r301046 | jpeeler | 2011-01-07 13:57:42 -0600 (Fri, 07 Jan 2011) | 8 lines Fix regression causing forwarding voicemails to not work with file storage. I had actually already fixed this in 295200 in 1.4 and thought it wasn't missing in the other branches for some reason. (closes issue #18358) Reported by: cabal95 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@301048 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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