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Author	SHA1	Message	Date
Mark Michelson	61c0d20d20	Merged revisions 166380 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r166380 | mmichelson | 2008-12-22 14:56:29 -0600 (Mon, 22 Dec 2008) | 36 lines Fix a deadlock relating to channel locks and autoservice It has been discovered that if a channel is locked prior to a call to ast_autoservice_stop, then it is likely that a deadlock will occur. The reason is that the call to ast_autoservice_stop has a check built into it to be sure that the thread running autoservice is not currently trying to manipulate the channel we are about to pull out of autoservice. The autoservice thread, however, cannot advance beyond where it currently is, though, because it is trying to acquire the lock of the channel for which autoservice is attempting to be stopped. The gist of all this is that a channel MUST NOT be locked when attempting to stop autoservice on the channel. In this particular case, the channel was locked by a call to ast_read. A call to ast_exists_extension led to autoservice being started and stopped due to the existence of dialplan switches. It may be that there are future commits which handle the same symptoms but in a different location, but based on my looks through the code, it is very rare to see a construct such as this one. (closes issue #14057) Reported by: rtrauntvein Patches: 14057v3.patch uploaded by putnopvut (license 60) Tested by: rtrauntvein Review: http://reviewboard.digium.com/r/107/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166382 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	cf25187ac4	Fix a bad typo. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166377 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	7166156658	Remove some error messages. This is the default handler that is valid to use. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166342 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	50387142b7	Merged revisions 166297 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r166297 | russell | 2008-12-22 11:22:56 -0600 (Mon, 22 Dec 2008) | 2 lines Fix up timeout handling in ast_carefulwrite(). ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166317 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	c2999a8366	Introduce ast_careful_fwrite() and use in AMI to prevent partial writes. ... This patch introduces a function to do careful writes on a file stream which will handle timeouts and partial writes. It is currently used in AMI to address the issue that has been reported. However, there are probably a few other places where this could be used. (closes issue #13546) Reported by: srt Tested by: russell http://reviewboard.digium.com/r/104/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166282 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	77b1fe0ceb	Re-work ref count handling of MoH classes using astobj2 to resolve crashes. ... (closes issue #13566) Reported by: igorcarneiro Tested by: russell Review: http://reviewboard.digium.com/r/106/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166273 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	29b658f040	Record the previous port in the temporary address structure so that the comparison does not treat the host as having changed even if it did not. This would have been uninitialized before and would have led to a baddddd port. ... (closes issue #13628) Reported by: pananix Patches: bug13628.patch uploaded by jpeeler (license 325) Tested by: file, blitzrage git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166268 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	9f7ce9da41	Fix a file playback crash and explicitly initialize values in func_timeout.c ... A crash was brought up on the bugtracker. The first run through valgrind was full of legitimate complaints of uninitialized values in func_timeout when setting a response timeout. These were fixed but the crash persisted. A second run through showed the real problem. The reference counting used for filestreams was incorrect because there were some missing increments when a frame was read from a format module. (closes issue #14118) Reported by: blitzrage Patches: 14118v2.patch uploaded by putnopvut (license 60) Tested by: blitzrage git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166267 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	8556f306de	Blocked revisions 166262 via svnmerge ... ........ r166262 | russell | 2008-12-22 08:45:27 -0600 (Mon, 22 Dec 2008) | 7 lines Re-work ref count handling of MoH classes using astobj2 to resolve crashes. (closes issue #13566) Reported by: igorcarneiro Tested by: russell Review: http://reviewboard.digium.com/r/106/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166263 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	7e72821959	Remove AST_PBX_KEEPALIVE usage from res_agi. ... This patch removes the usage of AST_PBX_KEEPALIVE from res_agi. The only usage was for the AGI command, "asyncagi break". This patch removes this feature. Normally, a feature would not be removed like this. However, this code is broken and usage of it will result in a memory leak. Usage of this feature will make the AGI code return a result of AST_PBX_KEEPALIVE. The PBX handler assumes that another thread has assumed ownership of the channel. The channel thread will exit without destroying the channel. Unfortunately, _no_ thread has ownership of the channel at this point. There are a couple of serious problems here: 1) The only way to recover the caller is to issue a channel redirect. This will work, but this will be done with a masquerade, and the old ast_channel structure will be lost. 2) Until the channel redirect happens, there is no code servicing the channel. That means nothing is reading audio or handling events coming from the channel. This is very bad. The recommended way to get this same "break" functionality is to issue the redirect while the channel is still being handled by the AGI code. That way, there will be no memory leak, and there will be no period of time that the channel is not being serviced. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166258 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	894c91afe0	Make a note about formatting the review URL in commit messages ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166219 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	e015e6f404	Get rid of an extra space. ... I don't know how this crept back in when I had already fixed it earlier git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166162 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	b0cc15d519	Blocked revisions 166157 via svnmerge ... ........ r166157 | mmichelson | 2008-12-19 17:34:57 -0600 (Fri, 19 Dec 2008) | 9 lines Backport of AUDIOHOOK_INHERIT for Asterisk 1.4 (closes issue #13538) Reported by: mbit Patches: 13538.patch uploaded by putnopvut (license 60) Tested by: putnopvut ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166158 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	c4ea017532	Remove the verbatim tag from the author line ... I could have sworn I already did that before, though... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166095 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	9733b30ff0	Adding a new dialplan function AUDIOHOOK_INHERIT ... This function is being added as a method to allow for an audiohook to move to a new channel during a channel masquerade. The most obvious use for such a facility is for MixMonitor when a transfer is performed. Prior to the addition of this functionality, if a channel running MixMonitor was transferred by another party, then the recording would stop once the transfer had completed. By using AUDIOHOOK_INHERIT, you can make MixMonitor continue recording the call even after the transfer has completed. It has also been determined that since this is seen by most as a bug fix and is not an invasive change, this functionality will also be backported to 1.4 and merged into the 1.6.0 branches, even though they are feature-frozen. (closes issue #13538) Reported by: mbit Patches: 13538.patch uploaded by putnopvut (license 60) Tested by: putnopvut Review: http://reviewboard.digium.com/r/102/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166092 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Matthew Fredrickson	775033301a	Add configuration support for half_full DAHDI buffer policy ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166058 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	b59f7d77d5	Blocked revisions 165991 via svnmerge ... ........ r165991 | jpeeler | 2008-12-19 13:48:00 -0600 (Fri, 19 Dec 2008) | 15 lines (closes issue #13480) Reported by: tzafrir Replace a bunch of if defined checks for Zaptel/DAHDI through several new defines in dahdi_compat.h. This removes a lot of code duplication. Example from bug: #ifdef HAVE_ZAPTEL fd = open("/dev/zap/pseudo", O_RDWR); #else fd = open("/dev/dahdi/pseudo", O_RDWR); #endif is replaced with: fd = open(DAHDI_FILE_PSEUDO, O_RDRW); ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@165992 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Eliel C. Sardanons	340d22ab39	Fix the XML documentation for Record(). ... <value> tags inside <variable> elements must have CDATA and no another XML node. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@165954 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	37d8f255e4	Merged revisions 165889 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r165889 | russell | 2008-12-19 09:03:02 -0600 (Fri, 19 Dec 2008) | 9 lines Ensure that the chanspy datastore is fully initialized. This patch resolved some random crash issues observed by a user on a BSD system (closes issue #14111) Reported by: ys Patches: app_chanspy.c.diff uploaded by ys (license 281) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@165890 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	79fe1aa0c6	Disable some automatic links generated by doxygen. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@165886 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	2450098785	Introduce commit message formatting guidelines. ... This documents the recommended outline to use for commit message. It also covers information on special tags that can be used in commit messages, as well as the template functionality that is available on bugs.digium.com. Review: http://reviewboard.digium.com/r/96/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@165883 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	1cb4baade2	Merged revisions 165796 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r165796 | russell | 2008-12-18 15:39:25 -0600 (Thu, 18 Dec 2008) | 11 lines Make ast_carefulwrite() be more careful. This patch handles some additional cases that could result in partial writes to the file description. This was done to address complaints about partial writes on AMI. (issue #13546) (more changes needed to address potential problems in 1.6) Reported by: srt Tested by: russell Review: http://reviewboard.digium.com/r/99/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@165801 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	4e4093ab48	(closes issue #13993) ... Reported by: mika Add ActionID response to ping if sent with request. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@165798 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	a117714b88	Merged revisions 165767 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r165767 | tilghman | 2008-12-18 15:14:47 -0600 (Thu, 18 Dec 2008) | 8 lines Add mutexes around accesses to the IMAP library interface. This prevents certain crashes, especially when shared mailboxes are used. (closes issue #13653) Reported by: howardwilkinson Patches: asterisk-1.4.21.2-appvoicemail-sharedimap-lock.patch uploaded by howardwilkinson (license 590) Tested by: jpeeler ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@165797 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	654ea55a65	Numerous documentation updates. ... (closes issue #13970) Reported by: pkempgen Patches: __20081217_cli_usage_fixes.patch.txt uploaded by blitzrage (license 10) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@165792 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	221694480c	Fix crashes in res_odbc. ... The variable "class" was being set NULL just prior to being dereferenced in an ao2_link call. I have moved the setting of the variable to NULL until after the ao2_link call. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@165724 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	50a25ac847	Remove the need for AST_PBX_KEEPALIVE with the GoSub option from Dial. ... This is part of an effort to completely remove AST_PBX_KEEPALIVE and other similar return codes from the source. While this usage was perfectly safe, there are others that are problematic. Since we know ahead of time that we do not want to PBX to destroy the channel, the PBX API has been changed so that information can be provided as an argument, instead, thus removing the need for the KEEPALIVE return value. Further changes to get rid of KEEPALIVE and related code is being done by murf. There is a patch up for that on review 29. Review: http://reviewboard.digium.com/r/98/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@165723 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	8cc50d4677	Merged revisions 165661 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r165661 | russell | 2008-12-18 12:52:18 -0600 (Thu, 18 Dec 2008) | 7 lines Set the process group ID on the MOH process so that all children will get killed (closes issue #14099) Reported by: caspy Patches: res_musiconhold.c.patch.killpg.try2 uploaded by caspy (license 645) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@165662 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	665b55e6f8	Fix 2 resource leaks and fix another pipe-to-comma conversion ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@165658 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	549fcd78a1	Merged revisions 165591 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r165591 | file | 2008-12-18 13:11:42 -0400 (Thu, 18 Dec 2008) | 4 lines Only care about a compatible codec for early bridging if we are actually bridging to another channel. If we are not we actually want to bring the audio back to us. (closes issue #13545) Reported by: davidw ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@165599 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	a2c557f3a1	Fix reference counts of the class and add an assertion to the end. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@165541 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	afba7b0a36	Blocked revisions 165537 via svnmerge ... ........ r165537 | file | 2008-12-18 12:07:37 -0400 (Thu, 18 Dec 2008) | 6 lines Do not crash if we are not passed in a followme id. (closes issue #14106) Reported by: ys Patches: app_followme.c.2.diff uploaded by ys (license 281) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@165538 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Eliel C. Sardanons	344a37f2a7	Remove duplicate code from the ast_str API. We now use __AST_STR_* to ... access 'struct ast_str' members, but this must only be used inside the API implementation. (closes issue #14098) Reported by: eliel Patches: ast_str.patch uploaded by eliel (license 64) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@165502 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	9e3ecac5ab	Add a \todo note for app_originate. ... Jared Smith suggested that we add a way to be able to set variables and functions on the outbound channel. I think that it's a great idea, so I have added it as a todo so that it gets done at some point. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@165469 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	aecde42abb	Add a new application, Originate. ... (closes issue #14075) Reported by: rcasas Patches: app_originate.c uploaded by rcasas (license 641), heavily modified by me Tested by: russell Review: http://reviewboard.digium.com/r/95/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@165433 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	08ae164b58	Add RECORD_STATUS variable, as requested on the -users list. ... Patch by me (license 14) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@165397 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	6c459b1b58	Fix a refcount leak in res_odbc ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@165330 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	7c1bd94231	Fix the build ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@165326 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	6c521ba21f	Oops, broke trunk ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@165325 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	f09b0b3a83	Merged revisions 165317 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r165317 | tilghman | 2008-12-17 15:14:37 -0600 (Wed, 17 Dec 2008) | 4 lines Reverse the fix from issue #6176 and add proper handling for that issue. (Closes issue #13962, closes issue #13363) Fixed by myself (license 14) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@165319 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	a7829044ec	Merged revisions 165255 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r165255 | mmichelson | 2008-12-17 14:51:38 -0600 (Wed, 17 Dec 2008) | 7 lines Fix some memory leaks found while looking at how realtime configs are handled. Also cleaned up some coding guidelines violations in app_realtime.c, mostly related to spacing ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@165318 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Steve Murphy	834782960b	This patch is here committed to satisfy the ... buildbot, who has a problem with the const. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@165254 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Terry Wilson	647c8f2222	Polycom phones close the connection after reading a little bit of the firmware files, we should stop sending in that case. Also, make that case print out a debug statement instead of a scary WARNING. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@165219 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	4534957e81	Call proxy_update so that the IP address gets populated. Sending stuff to 0.0.0.0 is silly! ... (closes issue #14055) Reported by: chris-mac git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@165216 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Matthew Nicholson	91192e30c5	This patch adds a new 'ignoresdpversion' option to sip.conf. When this is ... enabled (either globally or for a specific peer), chan_sip will treat any SDP data it receives as new data and update the media stream accordingly. By default, Asterisk will only modify the media stream if the SDP session version received is different from the current SDP session version. This option is required to interoperate with devices that have non-standard SDP session version implementations (observed by toc on the bug tracker with Microsoft OCS which always uses 0 as the session version). http://reviewboard.digium.com/r/94/ (closes issue #13958) Reported by: toc Tested by: toc git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@165180 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	0ab0896516	argsep is used as an attribute for an argument, as well ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@165145 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	93a51114cb	And actually assign the function to a pointer... ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@165143 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	d0671d8d30	Use the create_vm_state_from_user function in a place where ... it was not being used before. Also, I've moved the urgent folder check in messagecount() up a bit so that the flow is a bit better. This was something I noticed while taking a look at issue #13973, although I don't think this is the underlying cause of the issue. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@165142 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	cacecc2971	ignore this copied file ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@165108 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Steve Murphy	1a37ee4025	A possibly "horrible fix" for a "horribly broken" ... situation. As stuff shifts around in the asterisk code, the miscellaneous inclusions from the standalone stuff gets broken. There's no easy fix for this situation. I made sure that everything in utils builds without problem ***AND*** that aelparse runs the regressions correctly with the following make menuselect options both on and off: DONT_OPTIMIZE DEBUG_THREADS DEBUG_CHANNEL_LOCKS MALLOC_DEBUG MTX_PROFILE DEBUG_SCHEDULER DEBUG_THREADLOCALS DETECT_DEADLOCKS CHANNEL_TRACE I think from now on, I'm going to #undef all these features in the various utils native files; I guess I could do the same for the copied-in files, surrounded by STANDALONE ifdef. A standalone isn't going to care about threads, mutexes, etc. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@165071 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago