asterisk

Commit Graph

Author	SHA1	Message	Date
Tilghman Lesher	0e51fd84c4	Merged revisions 160791 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r160791 \| tilghman \| 2008-12-03 15:58:21 -0600 (Wed, 03 Dec 2008) \| 9 lines Merged revisions 160770 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r160770 \| tilghman \| 2008-12-03 15:54:07 -0600 (Wed, 03 Dec 2008) \| 2 lines Some compilers warn on null format strings; some don't (caught by buildbot) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@160807 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	3d5b168cbc	Merged revisions 160626 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r160626 \| mmichelson \| 2008-12-03 12:37:46 -0600 (Wed, 03 Dec 2008) \| 16 lines Add some safety measures when using gosub, especially when using the options for app_dial and app_queue to run a gosub when the call is answered. * Check for the existence of the gosub target in gosub_exec. If it is nonexistent, then this will cause errors when we attempt to actually run the gosub, including a definite memory leak and potential crashes. Return an error in this situation * Check the return value of pbx_exec in app_dial and app_queue before attempting to actually run the gosub routine. If there was an error, we should not attempt to run the gosub. * Change a '\|' to a ',' in app_queue. * Add some extra curly braces where they had been missing previously. (closes issue #13548) Reported by: fiddur ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@160627 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	f2168e3a67	Merged revisions 160555 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r160555 \| mmichelson \| 2008-12-03 11:07:09 -0600 (Wed, 03 Dec 2008) \| 11 lines When investigating issue #13548, I found that gosub handling in app_queue was just completely wrong, mostly because the channel operations being performed were being done on the incorrect channel. With this set of changes, a gosub will correctly run on the answering queue member's channel. There are still crash issues which occur if there are dialplan syntax errors, so I cannot yet close the referenced issue. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@160556 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	452f1f9c0d	Merged revisions 156388 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r156388 \| tilghman \| 2008-11-12 15:34:51 -0600 (Wed, 12 Nov 2008) \| 12 lines Merged revisions 156386 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r156386 \| tilghman \| 2008-11-12 15:18:57 -0600 (Wed, 12 Nov 2008) \| 5 lines When using call limits under 1 second, infinite call lengths are allowed, instead. (closes issue #13851) Reported by: ruddy ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@160393 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	d6effd3f11	Merged revisions 156290 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r156290 \| jpeeler \| 2008-11-12 13:11:15 -0600 (Wed, 12 Nov 2008) \| 11 lines Merged revisions 156289 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r156289 \| jpeeler \| 2008-11-12 13:10:12 -0600 (Wed, 12 Nov 2008) \| 3 lines For whatever reason, gcc only warned me about the possible use of an uninitialized variable when compiling 1.6.1. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@160392 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	1a2fee8c28	Merged revisions 156228 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r156228 \| jpeeler \| 2008-11-12 12:32:46 -0600 (Wed, 12 Nov 2008) \| 16 lines Merged revisions 156178 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r156178 \| jpeeler \| 2008-11-12 11:53:44 -0600 (Wed, 12 Nov 2008) \| 8 lines (closes issue #13173) Reported by: pep This change adds an announce_thread responsible for playing announcements to an existing conference. This allows all announcing to be immediately stopped if necessary but more importantly allows other threads that need to play something to not block. There are multiple benefits to this, but the actual bug is for solving the scenario for a channel to be unusable after hang up for the entire duration of the parting announcement. The parting announcement can be extremely long depending on what the user recorded upon joining the conference. Reviewed by Russell on Review Board: http://reviewboard.digium.com/r/25/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@160391 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	107dfe787e	Merged revisions 152969,153122,154264,154268,154366,155399,155863,156166,156295,156690,156756,158066,158082,158540,158602,159276 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r152969 \| tilghman \| 2008-10-30 15:35:46 -0500 (Thu, 30 Oct 2008) \| 10 lines Merged revisions 152958 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r152958 \| tilghman \| 2008-10-30 15:33:28 -0500 (Thu, 30 Oct 2008) \| 3 lines Cannot join detached threads. See http://www.opengroup.org/onlinepubs/000095399/functions/pthread_join.html (Closes issue #13400) ........ ................ r153122 \| tilghman \| 2008-10-31 11:35:21 -0500 (Fri, 31 Oct 2008) \| 10 lines Merged revisions 153114 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r153114 \| tilghman \| 2008-10-31 11:30:32 -0500 (Fri, 31 Oct 2008) \| 3 lines Turn off qualify on uncached realtime peers. (Closes issue #13383) ........ ................ r154264 \| tilghman \| 2008-11-04 12:59:48 -0600 (Tue, 04 Nov 2008) \| 10 lines Recorded merge of revisions 154263 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r154263 \| tilghman \| 2008-11-04 12:58:05 -0600 (Tue, 04 Nov 2008) \| 3 lines Make the monitor thread non-detached, so it can be joined (suggested by Russell on -dev list). ........ ................ r154268 \| rmudgett \| 2008-11-04 13:07:26 -0600 (Tue, 04 Nov 2008) \| 11 lines Merged revisions 154266 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r154266 \| rmudgett \| 2008-11-04 13:01:08 -0600 (Tue, 04 Nov 2008) \| 4 lines JIRA ABE-1703 mISDN sets the channel to the wrong state when it receives the indication AST_CONTROL_RINGING. ........ ................ r154366 \| tilghman \| 2008-11-04 14:51:18 -0600 (Tue, 04 Nov 2008) \| 16 lines Merged revisions 154365 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r154365 \| tilghman \| 2008-11-04 14:49:33 -0600 (Tue, 04 Nov 2008) \| 9 lines On busy systems, it's possible for the values checked within a single line of code to change, unless the structure is locked to ensure a consistent state. (closes issue #13717) Reported by: kowalma Patches: 20081102__bug13717.diff.txt uploaded by Corydon76 (license 14) Tested by: kowalma ........ ................ r155399 \| tilghman \| 2008-11-07 16:28:58 -0600 (Fri, 07 Nov 2008) \| 14 lines Merged revisions 155398 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r155398 \| tilghman \| 2008-11-07 16:27:32 -0600 (Fri, 07 Nov 2008) \| 7 lines Clarify error message. (closes issue #13809) Reported by: denke Patches: 20081104__bug13809.diff.txt uploaded by Corydon76 (license 14) Tested by: denke ........ ................ r155863 \| mmichelson \| 2008-11-10 15:14:44 -0600 (Mon, 10 Nov 2008) \| 22 lines Merged revisions 155861 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r155861 \| mmichelson \| 2008-11-10 15:07:39 -0600 (Mon, 10 Nov 2008) \| 14 lines Channel drivers assume that when their indicate callback is invoked, that the channel on which the callback was called is locked. This patch corrects an instance in chan_agent where a channel's indicate callback is called directly without first locking the channel. This was leading to some observed locking issues in chan_local, but considering that all channel drivers operate under the same expectations, the generic fix in chan_agent is the right way to go. AST-126 ........ ................ r156166 \| russell \| 2008-11-12 11:38:20 -0600 (Wed, 12 Nov 2008) \| 15 lines Merged revisions 156164 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r156164 \| russell \| 2008-11-12 11:29:52 -0600 (Wed, 12 Nov 2008) \| 7 lines Move the sanity check that makes sure "always fork" is not set along with the console option to be after the code that reads options from asterisk.conf. This resolves a situation where Asterisk can start taking up 100% when misconfigured. (Thanks to Bryce Porter (x86 on IRC) for letting me log in to his system to figure out what was causing the 100% CPU problem.) ........ ................ r156295 \| tilghman \| 2008-11-12 13:28:22 -0600 (Wed, 12 Nov 2008) \| 13 lines Merged revisions 156294 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r156294 \| tilghman \| 2008-11-12 13:26:45 -0600 (Wed, 12 Nov 2008) \| 6 lines If the SLA thread is not started, then reload causes a memory leak. (closes issue #13889) Reported by: eliel Patches: app_meetme.c.patch uploaded by eliel (license 64) ........ ................ r156690 \| tilghman \| 2008-11-13 15:30:41 -0600 (Thu, 13 Nov 2008) \| 14 lines Merged revisions 156688 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r156688 \| tilghman \| 2008-11-13 15:24:00 -0600 (Thu, 13 Nov 2008) \| 7 lines Provide more space for all the data which can appear in an originating channel name. (closes issue #13398) Reported by: bamby Patches: manager.c.diff uploaded by bamby (license 430) ........ ................ r156756 \| tilghman \| 2008-11-13 18:43:13 -0600 (Thu, 13 Nov 2008) \| 13 lines Merged revisions 156755 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r156755 \| tilghman \| 2008-11-13 18:41:37 -0600 (Thu, 13 Nov 2008) \| 6 lines ast_waitfordigit() requires that the channel be up, for no good logical reason. This prevents While/EndWhile from working within the "h" extension. Reported by: jgalarneau (for ABE C.2) Fixed by: me ........ ................ r158066 \| mmichelson \| 2008-11-20 11:39:06 -0600 (Thu, 20 Nov 2008) \| 20 lines Merged revisions 158053 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r158053 \| mmichelson \| 2008-11-20 11:33:06 -0600 (Thu, 20 Nov 2008) \| 12 lines Make sure to set the hangup cause on the calling channel in the case that ast_call() fails. For incoming SIP channels, this was causing us to send a 603 instead of a 486 when the call-limit was reached on the destination channel. (closes issue #13867) Reported by: still_nsk Patches: 13867.diff uploaded by putnopvut (license 60) Tested by: blitzrage ........ ................ r158082 \| mmichelson \| 2008-11-20 11:54:31 -0600 (Thu, 20 Nov 2008) \| 24 lines Merged revisions 158071 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r158071 \| mmichelson \| 2008-11-20 11:48:42 -0600 (Thu, 20 Nov 2008) \| 16 lines We don't handle 4XX responses to BYE well. According to section 15 of RFC 3261, we should terminate a dialog if we receive a 481 or 408 in response to our BYE. Since I am aware of at least one phone manufacturer who may sometimes send a 404 as well, I am being liberal and saying that any 4XX response to a BYE should result in a terminated dialog. (closes issue #12994) Reported by: pabelanger Patches: 12994.patch uploaded by putnopvut (license 60) Closes AST-129 ........ ................ r158540 \| russell \| 2008-11-21 16:12:37 -0600 (Fri, 21 Nov 2008) \| 10 lines Merged revisions 158539 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r158539 \| russell \| 2008-11-21 16:05:55 -0600 (Fri, 21 Nov 2008) \| 2 lines When compiling with DEBUG_THREADS, report the real file/func/line for ao2_lock/ao2_unlock ........ ................ r158602 \| tilghman \| 2008-11-21 17:14:11 -0600 (Fri, 21 Nov 2008) \| 12 lines Merged revisions 158600 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r158600 \| tilghman \| 2008-11-21 17:07:46 -0600 (Fri, 21 Nov 2008) \| 5 lines The passed extension may not be the same in the list as the current entry, because we strip spaces when copying the extension into the structure. Therefore, use the copied item to place the item into the list. (found by lmadsen on -dev, fixed by me) ........ ................ r159276 \| tilghman \| 2008-11-25 15:57:59 -0600 (Tue, 25 Nov 2008) \| 14 lines Merged revisions 159269 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r159269 \| tilghman \| 2008-11-25 15:56:48 -0600 (Tue, 25 Nov 2008) \| 7 lines Don't try to send a response on a NULL pvt. (closes issue #13919) Reported by: barthpbx Patches: chan_iax2.c.patch uploaded by eliel (license 64) Tested by: barthpbx ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@160389 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	b0bad74445	Merged revisions 152216,152287,152369,152467,152569,152605 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r152216 \| tilghman \| 2008-10-27 16:34:04 -0500 (Mon, 27 Oct 2008) \| 13 lines Merged revisions 152215 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r152215 \| tilghman \| 2008-10-27 16:32:00 -0500 (Mon, 27 Oct 2008) \| 6 lines Inherit ALL elements of CallerID across a local channel. (closes issue #13368) Reported by: Peter Schlaile Patches: 20080826__bug13368.diff.txt uploaded by Corydon76 (license 14) ........ ................ r152287 \| jpeeler \| 2008-10-27 18:31:39 -0500 (Mon, 27 Oct 2008) \| 10 lines Merged revisions 152286 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r152286 \| jpeeler \| 2008-10-27 18:28:49 -0500 (Mon, 27 Oct 2008) \| 2 lines Buffer policy setting for half is not needed. ........ ................ r152369 \| tilghman \| 2008-10-28 12:07:39 -0500 (Tue, 28 Oct 2008) \| 15 lines Merged revisions 152368 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r152368 \| tilghman \| 2008-10-28 12:04:56 -0500 (Tue, 28 Oct 2008) \| 8 lines Reset all DIAL variables back to blank, in case Dial is called multiple times per call (which could otherwise lead to inconsistent status reports). (closes issue #13216) Reported by: ruddy Patches: 20081014__bug13216.diff.txt uploaded by Corydon76 (license 14) Tested by: ruddy ........ ................ r152467 \| tilghman \| 2008-10-28 17:33:40 -0500 (Tue, 28 Oct 2008) \| 10 lines Merged revisions 152463 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r152463 \| tilghman \| 2008-10-28 17:32:34 -0500 (Tue, 28 Oct 2008) \| 3 lines Quoting in the wrong direction (Fixes AST-107) ........ ................ r152569 \| russell \| 2008-10-29 00:34:26 -0500 (Wed, 29 Oct 2008) \| 15 lines Merged revisions 152539 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r152539 \| russell \| 2008-10-29 00:23:51 -0500 (Wed, 29 Oct 2008) \| 7 lines Fix an incorrect usage of sizeof() (closes issue #13795) Reported by: andrew53 Patches: chan_sip_sizeof.patch uploaded by andrew53 (license 519) ........ ................ r152605 \| murf \| 2008-10-29 00:47:13 -0500 (Wed, 29 Oct 2008) \| 22 lines Merged revisions 152538 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r152538 \| murf \| 2008-10-28 23:19:04 -0600 (Tue, 28 Oct 2008) \| 14 lines A little documentation cross-ref between features and dial and queue... I wasted some time (stupidly) trying to get the one-touch parking stuff working, because it didn't occur to me that I had to also have the corresponding options in the dial command! Duh! (In all this time, I never set this up before!) So, to keep some poor fool from suffering the same fate, I made the features.conf.sample file mention the corresponding opts in dial/queue; and the docs for dial/app specifically mention the corresponding decls in the feature.conf file. I hope this doesn't spoil some vast, eternal plan... ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@160388 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	8411899d44	Merged revisions 147518,147689,148000,148112,148268,148917,148988,149062,149131,149201,149205,149208 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r147518 \| file \| 2008-10-08 09:53:51 -0500 (Wed, 08 Oct 2008) \| 9 lines Merged revisions 147517 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r147517 \| file \| 2008-10-08 11:51:42 -0300 (Wed, 08 Oct 2008) \| 2 lines If we receive DTMF make sure that the state of the speech structure goes back to being not ready. (issue #LUMENVOX-8) ........ ................ r147689 \| kpfleming \| 2008-10-08 17:26:55 -0500 (Wed, 08 Oct 2008) \| 9 lines Merged revisions 147681 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r147681 \| kpfleming \| 2008-10-08 17:22:09 -0500 (Wed, 08 Oct 2008) \| 3 lines when parsing a text configuration option, ensure that the buffer on the stack is actually large enough to hold the legal values of that option, and also ensure that sscanf() knows to stop parsing if it would overrun the buffer (without these changes, specifying "buffers=...,immediate" would overflow the buffer on the stack, and could not have worked as expected) ........ ................ r148000 \| tilghman \| 2008-10-09 14:39:34 -0500 (Thu, 09 Oct 2008) \| 11 lines Merged revisions 147997 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r147997 \| tilghman \| 2008-10-09 14:38:33 -0500 (Thu, 09 Oct 2008) \| 4 lines When blank, callerid name and number should display "unknown caller" in voicemail emails. (Closes issue #13643) ........ ................ r148112 \| mmichelson \| 2008-10-09 18:15:33 -0500 (Thu, 09 Oct 2008) \| 26 lines Merged revisions 146026 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r146026 \| murf \| 2008-10-03 12:12:54 -0500 (Fri, 03 Oct 2008) \| 18 lines (closes issue #13579) Reported by: dwagner (closes issue #13584) Reported by: dwagner Tested by: murf, putnopvut The thought occurred to me that the res= from the extension spawn was ending up being returned from the bridge. "Thou shalt not poison the return value". Made the change and it appears to allow blind xfers to work as normal. If I'm wrong, reopen the bugs. But it looks good to me! Many thanks to putnopvut for helping me reproduce this! ........ ................ r148268 \| tilghman \| 2008-10-10 11:31:31 -0500 (Fri, 10 Oct 2008) \| 14 lines Merged revisions 148257 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r148257 \| tilghman \| 2008-10-10 11:25:31 -0500 (Fri, 10 Oct 2008) \| 7 lines User not notified of temporary greeting, if ODBC storage is in use. (closes issue #13659) Reported by: moliveras Patches: 20081009__bug13659.diff.txt uploaded by Corydon76 (license 14) Tested by: moliveras ........ ................ r148917 \| tilghman \| 2008-10-14 12:46:48 -0500 (Tue, 14 Oct 2008) \| 11 lines Merged revisions 148916 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r148916 \| tilghman \| 2008-10-14 12:41:08 -0500 (Tue, 14 Oct 2008) \| 4 lines Ensure that mail headers are 7-bit clean, even when UTF-8 characters are used in headers like 'Subject' and 'To'. Closes AST-107. ........ ................ r148988 \| tilghman \| 2008-10-14 14:03:44 -0500 (Tue, 14 Oct 2008) \| 9 lines Merged revisions 148987 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r148987 \| tilghman \| 2008-10-14 14:03:08 -0500 (Tue, 14 Oct 2008) \| 2 lines Some compilers warn, some don't. Fixing. ........ ................ r149062 \| tilghman \| 2008-10-14 15:16:48 -0500 (Tue, 14 Oct 2008) \| 13 lines Merged revisions 149061 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r149061 \| tilghman \| 2008-10-14 15:09:06 -0500 (Tue, 14 Oct 2008) \| 6 lines Check correct values in the return of ast_waitfor(); also, get rid of a possible memory leak. (closes issue #13658) Reported by: explidous Patch by: me ........ ................ r149131 \| mmichelson \| 2008-10-14 16:08:48 -0500 (Tue, 14 Oct 2008) \| 15 lines Merged revisions 149130 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r149130 \| mmichelson \| 2008-10-14 15:49:02 -0500 (Tue, 14 Oct 2008) \| 7 lines Don't allow reserved characters to be used in register lines in sip.conf. (closes issue #13570) Reported by: putnopvut ........ ................ r149201 \| mmichelson \| 2008-10-14 17:41:13 -0500 (Tue, 14 Oct 2008) \| 20 lines Merged revisions 149200 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r149200 \| mmichelson \| 2008-10-14 17:40:42 -0500 (Tue, 14 Oct 2008) \| 12 lines Update the queue with the correct number of calls and whether the call was completed within the service level when a transfer takes place. This way, we do not "break" the leastrecent and fewestcalls strategies by not logging a call until after the transferred call has ended. (closes issue #13395) Reported by: Marquis Patches: app_queue.c.transfer.patch uploaded by Marquis (license 32) ........ ................ r149205 \| mmichelson \| 2008-10-14 18:04:44 -0500 (Tue, 14 Oct 2008) \| 20 lines Merged revisions 149204 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r149204 \| mmichelson \| 2008-10-14 18:00:01 -0500 (Tue, 14 Oct 2008) \| 12 lines Add a tolerance period for sync-triggered audiohooks so that if packetization of audio is close (but not equal) we don't end up flushing the audiohooks over small inconsistencies in synchronization. Related to issue #13005, and solves the issue for most people who were experiencing the problem. However, a small number of people are still experiencing the problem on long calls, so I am not closing the issue yet ........ ................ r149208 \| mmichelson \| 2008-10-14 18:15:04 -0500 (Tue, 14 Oct 2008) \| 17 lines Merged revisions 149207 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r149207 \| mmichelson \| 2008-10-14 18:10:26 -0500 (Tue, 14 Oct 2008) \| 9 lines Call register_peer_exten even in the case that the peer's IP/port does not change. (closes issue #13309) Reported by: dimas Patches: v2-13309.patch uploaded by dimas (license 88) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@160387 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	0a9c41ca89	Merged revisions 160208 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r160208 \| tilghman \| 2008-12-01 18:37:21 -0600 (Mon, 01 Dec 2008) \| 10 lines Merged revisions 160207 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r160207 \| tilghman \| 2008-12-01 18:25:16 -0600 (Mon, 01 Dec 2008) \| 3 lines Ensure that Asterisk builds with --enable-dev-mode, even on the latest gcc and glibc. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@160228 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	f798f236fb	Merged revisions 159554 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r159554 \| mmichelson \| 2008-11-26 13:57:11 -0600 (Wed, 26 Nov 2008) \| 19 lines Add some necessary hangup commands in the case that forwarding a call fails 1) Hang up the original destination if the local channel cannot be requested. 2) Hang up the local channel (in addition to the original destination) if ast_call fails when calling the newly created local channel. This prevents channels from sticking around forever in the case of a botched call forward (e.g. to an extension which does not exist). (closes issue #13764) Reported by: davidw Patches: 13764_v2.patch uploaded by putnopvut (license 60) Tested by: putnopvut, davidw ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@159558 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Terry Wilson	fbcd6b9595	Merged revisions 159093 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r159093 \| twilson \| 2008-11-25 10:18:53 -0600 (Tue, 25 Nov 2008) \| 2 lines Add missing variable declaration for PPC code ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@159094 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	ac0c9bbc4d	Merged revisions 157706 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r157706 \| kpfleming \| 2008-11-19 06:42:19 -0600 (Wed, 19 Nov 2008) \| 5 lines make some corrections to the ast_agi_register_multiple(), ast_agi_unregister_multiple() and ast_agi_fdprintf() API calls to be consistent with API guidelines also, move UPGRADE.txt to UPGRADE-1.6.txt and make the new UPGRADE.txt contain information about upgrading between Asterisk 1.6 releases ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@157738 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	35a7e1deca	Merged revisions 157306 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r157306 \| mmichelson \| 2008-11-18 12:31:08 -0600 (Tue, 18 Nov 2008) \| 20 lines Merged revisions 157305 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r157305 \| mmichelson \| 2008-11-18 12:25:55 -0600 (Tue, 18 Nov 2008) \| 12 lines Fix a crash in the end_bridge_callback of app_dial and app_followme which would occur at the end of an attended transfer. The error occurred because we initially stored a pointer to an ast_channel which then was hung up due to a masquerade. This commit adds a "fixup" callback to the bridge_config structure to allow for end_bridge_callback_data to be changed in the case that a new channel pointer is needed for the end_bridge_callback. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@157307 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	4fbe80cc63	This is the 1.6.0 version of revision 156883 of trunk. This is different in that it preserves the case-sensitiveness of processing queues from configuration. closes issue #13703 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@156889 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	bd31630ab0	Merged revisions 156817 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r156817 \| mmichelson \| 2008-11-14 09:20:03 -0600 (Fri, 14 Nov 2008) \| 18 lines Merged revisions 156816 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r156816 \| mmichelson \| 2008-11-14 09:18:59 -0600 (Fri, 14 Nov 2008) \| 10 lines If the prompt to reenter a voicemail password timed out, it resulted in the password not being saved, even if the input matched what you gave when first prompted to enter a new password. This is because the return value of ast_readstring was checked, but not checked properly. This bug was discovered by Jared Smith during an Asterisk training course. Thanks for reporting it! ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@156818 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	221444dd9c	Merged revisions 156169 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r156169 \| mmichelson \| 2008-11-12 11:41:56 -0600 (Wed, 12 Nov 2008) \| 15 lines Merged revisions 156167 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r156167 \| mmichelson \| 2008-11-12 11:38:33 -0600 (Wed, 12 Nov 2008) \| 7 lines When doing some tests, I was having a crash at the end of every call if an attended transfer occurred during the call. I traced the cause to the CDR on one of the channels being NULL. murf suggested a check in the end bridge callback to be sure the CDR is non-NULL before proceeding, so that's what I'm adding. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@156170 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	62eefbec73	Don't blow up if we get NULL when trying to parse out the full name field (fixed for Jared in the training room) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@156012 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	0591cc3029	Merged revisions 155554 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r155554 \| seanbright \| 2008-11-08 20:27:00 -0500 (Sat, 08 Nov 2008) \| 14 lines Merged revisions 155553 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r155553 \| seanbright \| 2008-11-08 20:08:07 -0500 (Sat, 08 Nov 2008) \| 6 lines Use static functions here instead of nested ones. This requires a small change to the ast_bridge_config struct as well. To understand the reason for this change, see the following post: http://gcc.gnu.org/ml/gcc-help/2008-11/msg00049.html ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@155555 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	ede179faec	Merge revision 153709 from trunk ------------------------------------------------------------------------ r153709 \| kpfleming \| 2008-11-02 17:34:39 -0600 (Sun, 02 Nov 2008) \| 3 lines instead of trying to forcibly load res_agi when app_stack is loaded (even if the administrator didn't want it loaded), use GCC weak symbols to determine whether it was loaded already or not; if it was loaded, then use it. ------------------------------------------------------------------------ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@153745 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	fa635ea4b0	port gcc 4.3.x warning fixes from trunk to this branch git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@153743 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Terry Wilson	e23be17786	Merged revisions 153181 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r153181 \| twilson \| 2008-10-31 13:55:33 -0500 (Fri, 31 Oct 2008) \| 5 lines Recent CDR fixes moved execution of the 'h' exten into the bridging code, so variables that were set after ast_bridge_call was called would not show up in the 'h' exten. Added a callback function to handle setting variables, etc. from w/in the bridging code. Calls back into a nested function within the function calling ast_bridge_call (closes issue #13793) Reported by: greenfieldtech ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@153265 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	84dcf8e6b1	Merged revisions 152646 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r152646 \| mmichelson \| 2008-10-29 15:53:53 -0500 (Wed, 29 Oct 2008) \| 9 lines If there was no named defined in a voicemail.conf mailbox entry, then app_directory would crash when attempting to read that entry from the file. We now check for the NULL or empty string properly so that there will be no crash. (closes issue #13804) Reported by: bluecrow76 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@152647 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Terry Wilson	969bef1dea	Small modification to putnopvut's patch to fix this issue. Thanks for all the help, putnopvut! (closes issue #12884) Reported by: bcnit Patches: 12884v4-1.6.0-branch.patch uploaded by otherwiseguy (license 396) Tested by: otherwiseguy git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@152644 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	8bcea43631	Merged revisions 152134 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r152134 \| tilghman \| 2008-10-27 11:24:11 -0500 (Mon, 27 Oct 2008) \| 4 lines Oops, only delete the ARG variables once upon release. The following section would have removed them again (removing variables from 2 stack frames, instead of just one). ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@152157 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	eb87d6dac7	Merged revisions 148200 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r148200 \| seanbright \| 2008-10-09 20:42:13 -0400 (Thu, 09 Oct 2008) \| 12 lines Don't include logger.h in asterisk.h by default as it is causing problems building app_voicemail. Instead, include it where it is needed. This turned out to be a relatively minor issue because other headers include logger.h as well. Need to test -addons before merging this back to 1.6.0. (closes issue #13605) Reported by: tomo1657 Patches: 13605_seanbright.diff uploaded by seanbright (license 71) Tested by: mmichelson ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@148204 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	2b03afb74b	Somehow we got conflict markers checked in! Might need a 1.6.0.1 sooner than we'd like. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@148201 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	60399ff4ac	Merged revisions 148144 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r148144 \| mmichelson \| 2008-10-09 18:30:47 -0500 (Thu, 09 Oct 2008) \| 10 lines Read the callerid in the correct order and make sure to read the Urgent flag value from the IMAP headers. (closes issue #13652) Reported by: jaroth Patches: imapheaders.patch uploaded by jaroth (license 50) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@148147 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	d3edd930a9	Merged revisions 147194 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r147194 \| seanbright \| 2008-10-07 12:52:02 -0400 (Tue, 07 Oct 2008) \| 10 lines Merged revisions 147193 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r147193 \| seanbright \| 2008-10-07 12:48:30 -0400 (Tue, 07 Oct 2008) \| 2 lines Make 'imapsecret' an alias to 'imappassword' in voicemail.conf. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@147195 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	4d5405c74d	Merged revisions 147050 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r147050 \| seanbright \| 2008-10-07 08:01:36 -0400 (Tue, 07 Oct 2008) \| 8 lines Make sure to compare the correct number of characters when special-casing our DAHDI operator mode stuff. Technically, it would work fine, as 'DAH' is currently unique amongst our channel technologies, but as Jared points out: <@jsmith> Sure... as long as the technology starts whith DAH.... but it could be DAHDOO! ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@147051 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	9f9e461e48	Merged revisions 145428 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r145428 \| tilghman \| 2008-10-01 10:44:06 -0500 (Wed, 01 Oct 2008) \| 7 lines Initializing buffer prevents a segfault when arguments are incomplete. (closes issue #13471) Reported by: alecdavis Patches: 20080916__bug13471.diff.txt uploaded by Corydon76 (license 14) Tested by: alecdavis ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@145429 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Steve Murphy	561fd11a1f	Merged revisions 144569 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r144569 \| murf \| 2008-09-25 16:21:28 -0600 (Thu, 25 Sep 2008) \| 14 lines (closes issue #13557) Reported by: nickpeirson The user attached a patch, but the license is not yet recorded. I took the liberty of finding and replacing ALL index() calls with strchr() calls, and that involves more than just main/pbx.c; chan_oss, app_playback, func_cut also had calls to index(), and I changed them out. 1.4 had no references to index() at all. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@144570 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	73e498b461	Merged revisions 143405 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r143405 \| tilghman \| 2008-09-17 15:57:58 -0500 (Wed, 17 Sep 2008) \| 13 lines Merged revisions 143404 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r143404 \| tilghman \| 2008-09-17 15:55:47 -0500 (Wed, 17 Sep 2008) \| 6 lines When callerid is blank, we want to use "unknown caller" in those cases, too. (closes issue #13486) Reported by: tomo1657 Patches: 20080917__bug13486.diff.txt uploaded by Corydon76 (license 14) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@143406 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	5302fa36b8	Recorded merge of revisions 143031 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r143031 \| tilghman \| 2008-09-13 08:54:15 -0500 (Sat, 13 Sep 2008) \| 8 lines Repair IAXVAR implementation so that it works again (regression?) (closes issue #13354) Reported by: adomjan Patches: 20080828__bug13354.diff.txt uploaded by Corydon76 (license 14) 20080829__bug13354__1.6.0.diff.txt uploaded by Corydon76 (license 14) Tested by: Corydon76, adomjan ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@143032 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	1b94ff5b0e	Merged revisions 142745 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r142745 \| tilghman \| 2008-09-12 11:38:55 -0500 (Fri, 12 Sep 2008) \| 12 lines Merged revisions 142744 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r142744 \| tilghman \| 2008-09-12 11:38:02 -0500 (Fri, 12 Sep 2008) \| 4 lines Missing merge from 1.2 fixes errant exit on DTMF, only when language is Italian (cf commit 34242) (Closes issue #7353) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@142746 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Steve Murphy	3c1c24500b	Merged revisions 142676 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r142676 \| murf \| 2008-09-11 22:50:48 -0600 (Thu, 11 Sep 2008) \| 40 lines Merged revisions 142675 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r142675 \| murf \| 2008-09-11 22:29:34 -0600 (Thu, 11 Sep 2008) \| 29 lines Tested by: sergee, murf, chris-mac, andrew, KNK This is a "second attempt" to restore the previous "endbeforeh" behavior in 1.4 and up. In order to capture information concerning all the legs of transfers in all their infinite combinations, I was forced to this particular solution by a chain of logical necessities, the first being that I was not allowed to rewrite the CDR mechanism from the ground up! This change basically leaves the original machinery alone, which allows IVR and local channel type situations to generate CDR's as normal, but a channel flag can be set to suppress the normal running of the h exten. That flag would be set by the code that runs the h exten from the ast_bridge_call routine, to prevent the h exten from being run twice. Also, a flag in the ast_bridge_config struct passed into ast_bridge_call can be used to suppress the running of the h exten in that routine. This would happen, for instance, if you use the 'g' option in the Dial app. Running this routine 'early' allows not only the CDR() func to be used in the h extension for reading CDR variables, but also allows them to be modified before the CDR is posted to the backends. While I dearly hope that this patch overcomes all problems, and introduces no new problems, reality suggests that surely someone will have problems. In this case, please re-open 13251 (or 13289), and we'll see if we can't fix any remaining issues. ** trunk note: some code to suppress the h exten being run from app_queue was added; for the 'continue' option available only in trunk/1.6.x. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@142677 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	fac93e8478	Fix app_queue's device state callback so that it can correctly parse custom device states (and any other device which does not contain a '/'). 1.6.1 will be getting this patch as well, but trunk is going to get a much more massive patch by bbryant which does some very nice overhauling of some structures in app_queue. (closes issue #12979) Reported by: sigxcpu Patches: 12979.patch uploaded by putnopvut (license 60) Tested by: sigxcpu git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@142090 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	dc4224c694	Merged revisions 140975 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r140975 \| mmichelson \| 2008-09-03 15:16:12 -0500 (Wed, 03 Sep 2008) \| 4 lines Fix some locking order issues in app_queue. This was brought up by atis on IRC a while ago. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@140976 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	08966a95de	Merged revisions 140566 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r140566 \| russell \| 2008-09-02 10:11:53 -0500 (Tue, 02 Sep 2008) \| 2 lines Update instructions for getting libresample ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@140567 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Steve Murphy	ee980c4a33	Merged revisions 139627 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r139627 \| murf \| 2008-08-22 16:03:13 -0600 (Fri, 22 Aug 2008) \| 59 lines Merged revisions 139347 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r139347 \| murf \| 2008-08-21 17:03:50 -0600 (Thu, 21 Aug 2008) \| 47 lines (closes issue #13251) Reported by: sergee Tested by: murf THis is a bold move for a static release fix, but I wouldn't have made it if I didn't feel confident (at least a bit confident) that it wouldn't mess everyone up. The reasoning goes something like this: 1. We simply cannot do anything with CDR's at the current point (in pbx.c, after the __ast_pbx_run loop). It's way too late to have any affect on the CDRs. The CDR is already posted and gone, and the remnants have been cleared. 2. I was very much afraid that moving the running of the 'h' extension down into the bridge code (where it would be now practical to do it), would result in a lot more calls to the 'h' exten, so I implemented it as another exten under another name, but found, to my pleasant surprise, that there was a 1:1 correspondence to the running of the 'h' exten in the pbx_run loop, and the new spot at the end of the bridge. So, I ifdef'd out the current 'h' loop, and moved it into the bridge code. The only difference I can see is the stuff about the AST_PBX_KEEPALIVE, and hopefully, if this is still an important decision point, I can replicate it if there are complaints. To be perfectly honest, the KEEPALIVE situation is not totally clear to me, and how it relates to a post-bridge situation is less clear. I suspect the users will point out everything in total clarity if this steps on anyone's toes! 3. I temporarily swap the bridge_cdr into the channel before running the 'h' exten, which makes it possible for users to edit the cdr before it goes out the door. And, of course, with the endbeforehexten config var set, the users can also get at the billsec/duration vals. After the h exten finishes, the cdr is swapped back and processing continues as normal. Please, all who deal with CDR's, please test this version of Asterisk, and file bug reports as appropriate! ........ I also made a little fix to the app_dial's 'e' option, that is related to my updates. ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@139628 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	7d1bda1c07	Merged revisions 139215 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r139215 \| russell \| 2008-08-20 17:16:36 -0500 (Wed, 20 Aug 2008) \| 19 lines Merged revisions 139213 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r139213 \| russell \| 2008-08-20 17:14:35 -0500 (Wed, 20 Aug 2008) \| 11 lines Fix a crash in the ChanSpy application. The issue here is that if you call ChanSpy and specify a spy group, and sit in the application long enough looping through the channel list, you will eventually run out of stack space and the application with exit with a seg fault. The backtrace was always inside of a harmless snprintf() call, so it was tricky to track down. However, it turned out that the call to snprintf() was just the biggest stack consumer in this code path, so it would always be the first one to hit the boundary. (closes issue #13338) Reported by: ruddy ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@139216 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	a63274bf29	Manually add revision 138887 from trunk to the 1.6.0 branch. I had misunderstood the policy for when to merge to 1.6.0 since it moved to rc status. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@138890 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	ac72baa7b5	Merged revisions 138687 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r138687 \| mmichelson \| 2008-08-18 15:04:10 -0500 (Mon, 18 Aug 2008) \| 18 lines Merged revisions 138685 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r138685 \| mmichelson \| 2008-08-18 15:01:14 -0500 (Mon, 18 Aug 2008) \| 10 lines Change the inequalities used in app_queue with regards to timeouts from being strict to non-strict for more accuracy. (closes issue #13239) Reported by: atis Patches: app_queue_timeouts_v2.patch uploaded by atis (license 242) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@138688 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jason Parker	84d0f2de87	Merged revisions 137496 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r137496 \| qwell \| 2008-08-13 15:05:50 -0500 (Wed, 13 Aug 2008) \| 6 lines Add FAXMODE variable with what fax transport was used. (closes issue #13252) Patches: v1-13252.patch uploaded by dimas (license 88) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@137497 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	a6df06e84e	Merged revisions 136784 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r136784 \| mmichelson \| 2008-08-08 10:31:31 -0500 (Fri, 08 Aug 2008) \| 3 lines Fix compilation for ODBC voicemail ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@136785 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	4a5ac6487f	This is weird. Either SVN or vim tabbed a bunch of functions over one level during a merge. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@136724 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	745f28879f	Merged revisions 136722 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r136722 \| mmichelson \| 2008-08-07 18:39:50 -0500 (Thu, 07 Aug 2008) \| 3 lines Remove one last batch of debug messages ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@136723 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	5567deefc9	Merged revisions 136715 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r136715 \| mmichelson \| 2008-08-07 17:25:50 -0500 (Thu, 07 Aug 2008) \| 18 lines Merging the imap_consistency_trunk branch to trunk. For an explanation of what "imap_consistency" is, please see svn revision 134223 to the 1.4 branch. Coincidentally, this also fixes a recent bug report regarding the inability to save messages to the new folder when using IMAP storage since they will would be flagged as "seen" and not be recognized as new messages. (closes issue #13234) Reported by: jaroth ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@136719 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	f67e68de8e	Merged revisions 136489 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r136489 \| tilghman \| 2008-08-07 11:55:57 -0500 (Thu, 07 Aug 2008) \| 15 lines Merged revisions 136488 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r136488 \| tilghman \| 2008-08-07 11:50:47 -0500 (Thu, 07 Aug 2008) \| 7 lines Update persistent state on all exit conditions. (closes issue #12916) Reported by: sgenyuk Patches: app_queue.patch.txt uploaded by neutrino88 (license 297) Tested by: sgenyuk, aragon ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@136490 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Steve Murphy	638711f573	Merged revisions 135821 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r135821 \| murf \| 2008-08-05 17:45:32 -0600 (Tue, 05 Aug 2008) \| 42 lines Merged revisions 135799 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r135799 \| murf \| 2008-08-05 17:13:20 -0600 (Tue, 05 Aug 2008) \| 34 lines (closes issue #12982) Reported by: bcnit Tested by: murf I discovered that also, in the previous bug fixes and changes, the cdr.conf 'unanswered' option is not being obeyed, so I fixed this. And, yes, there are two 'answer' times involved in this scenario, and I would agree with you, that the first answer time is the time that should appear in the CDR. (the second 'answer' time is the time that the bridge was begun). I made the necessary adjustments, recording the first answer time into the peer cdr, and then using that to override the bridge cdr's value. To get the 'unanswered' CDRs to appear, I purposely output them, using the dial cmd to mark them as DIALED (with a new flag), and outputting them if they bear that flag, and you are in the right mode. I also corrected one small mention of the Zap device to equally consider the dahdi device. I heavily tested 10-sec-wait macros in dial, and without the macro call; I tested hangups while the macro was running vs. letting the macro complete and the bridge form. Looks OK. Removed all the instrumentation and debug. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@135822 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago

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