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2009-01-05More clearly explain that quote marks are no longer necessary.Tilghman Lesher
(closes issue #13718) Reported by: davidw Patches: 20081020__bug13718.diff.txt uploaded by Corydon76 (license 14) Tested by: blitzrage git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@167176 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-03When parsing environment variable ASTERISK_PROMPT, make sure to proceed to ↵Jeff Peeler
the next character when a non format specifier is used (no %). Otherwise, the while loop looking for the null byte will never exit. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@167125 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-31Mostly just whitespace, but also convert 'CVS' to 'SVN' in a coupleSean Bright
places and fix a few typos I found in the CODING_GUIDELINES. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@167061 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-31Don't forget to free typenameTerry Wilson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@167057 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-31Change some incorrect syntax for pri set debug and correctMark Michelson
an off-by-one error in ss7 set debug command git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@167021 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-31That was weird...Tilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166958 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-31Merged revisions 166953 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r166953 | tilghman | 2008-12-31 13:20:35 -0600 (Wed, 31 Dec 2008) | 5 lines Also inherit the musiconhold class. (Closes #14153) Reported by: Jerry Geis, via the users list. Patch by: me (license 14) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166954 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-30Fix some svn:keywordsTerry Wilson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166908 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-29Update app_queue to deal with the removal of AST_PBX_KEEPALIVEMark Michelson
When placing a call to a queue which ran a gosub on the member's channel, Asterisk would crash every time, stemming from the fact that the member's channel was being hung up unexpectedly when the Gosub completed. The necessary change was pretty much copied and pasted from app_dial's similar changes made last week. I also took the opportunity to change a LOG_DEBUG message in app_dial to use ast_debug. I am guessing this was due to a direct merge from 1.4 that was not corrected to use trunk's preferred syntax. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166861 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-28Fix a typo in the XML documentation of the AUDIOHOOK_INHERIT dialplan function.Eliel C. Sardanons
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166823 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-28Merged revisions 166772 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r166772 | russell | 2008-12-28 09:13:48 -0600 (Sun, 28 Dec 2008) | 4 lines Use strncat() instead of an sprintf() in which source and target buffers overlap http://lists.digium.com/pipermail/asterisk-dev/2008-December/035919.html ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166773 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-24There is no section 22.2.2 in rfc 3261. I believe 26.2.2 is what was meant:Terry Wilson
Note that in the SIPS URI scheme, transport is independent of TLS, and thus "sips:alice@atlanta.com;transport=tcp" and "sips:alice@atlanta.com;transport=sctp" are both valid (although note that UDP is not a valid transport for SIPS). The use of "transport=tls" has consequently been deprecated, partly because it was specific to a single hop of the request. This is a change since RFC 2543. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166731 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-23Allow semicolons and extended characters in user-specified SIP headers.Tilghman Lesher
(closes issue #14110) Reported by: gork Patches: 20081222__bug14110__2.diff.txt uploaded by Corydon76 (license 14) Tested by: gork, putnopvut git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166696 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-23Merged revisions 166093 via svnmerge from Steve Murphy
https://origsvn.digium.com/svn/asterisk/branches/1.4 In order to merge this 1.4 patch into trunk, I had to resolve some conflicts and wait for Russell to make some changes to res_agi. I re-ran all the tests; 39 calls in all, and made fairly careful notes and comparisons: I don't want this to blow up some aspect of asterisk; I completely removed the KEEPALIVE from the pbx.h decls. The first 3 scenarios involving feature park; feature xfer to 700; hookflash park to Park() app call all behave the same, don't appear to leave hung channels, and no crashes. ........ r166093 | murf | 2008-12-19 15:30:32 -0700 (Fri, 19 Dec 2008) | 131 lines This merges the masqpark branch into 1.4 These changes eliminate the need for (and use of) the KEEPALIVE return code in res_features.c; There are other places that use this result code for similar purposes at a higher level, these appear to be left alone in 1.4, but attacked in trunk. The reason these changes are being made in 1.4, is that parking ends a channel's life, in some situations, and the code in the bridge (and some other places), was not checking the result code properly, and dereferencing the channel pointer, which could lead to memory corruption and crashes. Calling the masq_park function eliminates this danger in higher levels. A series of previous commits have replaced some parking calls with masq_park, but this patch puts them ALL to rest, (except one, purposely left alone because a masquerade is done anyway), and gets rid of the code that tests the KEEPALIVE result, and the NOHANGUP_PEER result codes. While bug 13820 inspired this work, this patch does not solve all the problems mentioned there. I have tested this patch (again) to make sure I have not introduced regressions. Crashes that occurred when a parked party hung up while the parking party was listening to the numbers of the parking stall being assigned, is eliminated. These are the cases where parking code may be activated: 1. Feature one touch (eg. *3) 2. Feature blind xfer to parking lot (eg ##700) 3. Run Park() app from dialplan (eg sip xfer to 700) (eg. dahdi hookflash xfer to 700) 4. Run Park via manager. The interesting testing cases for parking are: I. A calls B, A parks B a. B hangs up while A is getting the numbers announced. b. B hangs up after A gets the announcement, but before the parking time expires c. B waits, time expires, A is redialed, A answers, B and A are connected, after which, B hangs up. d. C picks up B while still in parking lot. II. A calls B, B parks A a. A hangs up while B is getting the numbers announced. b. A hangs up after B gets the announcement, but before the parking time expires c. A waits, time expires, B is redialed, B answers, A and B are connected, after which, A hangs up. d. C picks up A while still in parking lot. Testing this throroughly involves acting all the permutations of I and II, in situations 1,2,3, and 4. Since I added a few more changes (ALL references to KEEPALIVE in the bridge code eliimated (I missed one earlier), I retested most of the above cases, and no crashes. H-extension weirdness. Current h-extension execution is not completely correct for several of the cases. For the case where A calls B, and A parks B, the 'h' exten is run on A's channel as soon as the park is accomplished. This is expected behavior. But when A calls B, and B parks A, this will be current behavior: After B parks A, B is hung up by the system, and the 'h' (hangup) exten gets run, but the channel mentioned will be a derivative of A's... Thus, if A is DAHDI/1, and B is DAHDI/2, the h-extension will be run on channel Parked/DAHDI/1-1<ZOMBIE>, and the start/answer/end info will be those relating to Channel A. And, in the case where A is reconnected to B after the park time expires, when both parties hang up after the joyful reunion, no h-exten will be run at all. In the case where C picks up A from the parking lot, when either A or C hang up, the h-exten will be run for the C channel. CDR's are a separate issue, and not addressed here. As to WHY this strange behavior occurs, the answer lies in the procedure followed to accomplish handing over the channel to the parking manager thread. This procedure is called masquerading. In the process, a duplicate copy of the channel is created, and most of the active data is given to the new copy. The original channel gets its name changed to XXX<ZOMBIE> and keeps the PBX information for the sake of the original thread (preserving its role as a call originator, if it had this role to begin with), while the new channel is without this info and becomes a call target (a "peer"). In this case, the parking lot manager thread is handed the new (masqueraded) channel. It will not run an h-exten on the channel if it hangs up while in the parking lot. The h exten will be run on the original channel instead, in the original thread, after the bridge completes. See bug 13820 for our intentions as to how to clean up the h exten behavior. Review: http://reviewboard.digium.com/r/29/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166665 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-23Fix spelling error.Russell Bryant
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166625 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-23Blocked revisions 166592 via svnmergeTilghman Lesher
........ r166592 | tilghman | 2008-12-23 09:35:38 -0600 (Tue, 23 Dec 2008) | 3 lines Compile, even if both DAHDI and Zaptel are not installed. (Closes issue #14120) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166601 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-23Merged revisions 166568 via svnmerge from Mark Michelson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r166568 | mmichelson | 2008-12-23 09:16:26 -0600 (Tue, 23 Dec 2008) | 12 lines Fix a crash resulting from a datastore with inheritance but no duplicate callback The fix for this is to simply set the newly created datastore's data pointer to NULL if it is inherited but has no duplicate callback. (closes issue #14113) Reported by: francesco_r Patches: 14113.patch uploaded by putnopvut (license 60) Tested by: francesco_r ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166569 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-23Merged revisions 166509 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r166509 | tilghman | 2008-12-22 22:05:25 -0600 (Mon, 22 Dec 2008) | 4 lines Use the integer form of condition for integer comparisons. (closes issue #14127) Reported by: andrew ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166533 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-22Always use the value of the AGISIGHUP when running an AGI.Mark Michelson
Prior to this patch, the value of AGISIGUP was not always honored when set on a channel. (closes issue #13711) Reported by: fmueller Patches: 13711.patch uploaded by putnopvut (license 60) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166470 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-22Cosmetic change - don't mix struct initializer styles.Russell Bryant
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166436 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-22Merged revisions 166380 via svnmerge from Mark Michelson
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
2008-12-22Fix a bad typo.Russell Bryant
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166377 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-22Remove some error messages. This is the default handler that is valid to use.Russell Bryant
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166342 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-22Merged revisions 166297 via svnmerge from Russell Bryant
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
2008-12-22Introduce ast_careful_fwrite() and use in AMI to prevent partial writes.Russell Bryant
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
2008-12-22Re-work ref count handling of MoH classes using astobj2 to resolve crashes.Russell Bryant
(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
2008-12-22Record the previous port in the temporary address structure so that the ↵Joshua Colp
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
2008-12-22Fix a file playback crash and explicitly initialize values in func_timeout.cMark Michelson
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
2008-12-22Blocked revisions 166262 via svnmergeRussell Bryant
........ 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
2008-12-22Remove AST_PBX_KEEPALIVE usage from res_agi.Russell Bryant
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
2008-12-20Make a note about formatting the review URL in commit messagesRussell Bryant
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166219 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-19Get rid of an extra space.Mark Michelson
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
2008-12-19Blocked revisions 166157 via svnmergeMark Michelson
........ 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
2008-12-19Remove the verbatim tag from the author lineMark Michelson
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
2008-12-19Adding a new dialplan function AUDIOHOOK_INHERITMark Michelson
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
2008-12-19Add configuration support for half_full DAHDI buffer policyMatthew Fredrickson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166058 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-19Blocked revisions 165991 via svnmergeJeff Peeler
........ 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
2008-12-19Fix the XML documentation for Record().Eliel C. Sardanons
<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
2008-12-19Merged revisions 165889 via svnmerge from Russell Bryant
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
2008-12-19Disable some automatic links generated by doxygen.Russell Bryant
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@165886 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-19Introduce commit message formatting guidelines.Russell Bryant
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
2008-12-18Merged revisions 165796 via svnmerge from Russell Bryant
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
2008-12-18(closes issue #13993)Jeff Peeler
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
2008-12-18Merged revisions 165767 via svnmerge from Tilghman Lesher
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
2008-12-18Numerous documentation updates.Joshua Colp
(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
2008-12-18Fix crashes in res_odbc.Mark Michelson
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
2008-12-18Remove the need for AST_PBX_KEEPALIVE with the GoSub option from Dial.Russell Bryant
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
2008-12-18Merged revisions 165661 via svnmerge from Russell Bryant
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
2008-12-18Fix 2 resource leaks and fix another pipe-to-comma conversionTilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@165658 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-18Merged revisions 165591 via svnmerge from Joshua Colp
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
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