asterisk

Commit Graph

Author	SHA1	Message	Date
Richard Mudgett	93601856b6	Merged revision 310986 from https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier .......... r310986 \| rmudgett \| 2011-03-16 13:56:28 -0500 (Wed, 16 Mar 2011) \| 28 lines Dial() o option broke when connected line feature added. The patch restores the o option behavior and adds the ability to specify the CallerID. The Dial o and f options are complementary to each other. The o option stores the CallerID on the outgoing channel as the channel's CallerID. The f option forces the CallerID sent by the outgoing channel. o(x) - The argument 'x' is optional. If not present, then specify that the CallerID that was present on the calling channel be stored as the CallerID on the called channel. This was the behavior of Asterisk 1.0 and earlier. If present, then specify the CallerID stored on the called channel. Note that o(${CALLERID(all)}) is similar to option o without parameters. f(x) - The argument 'x' is optional and its presence changes the behavior of this option. If not present, then force the outgoing CallerID on a call-forward or deflection to the dialplan extension for this Dial() using a dialplan 'hint'. For example, some PSTNs do not allow CallerID to be set to anything other than the numbers assigned to you. If present, then force the outgoing CallerID to 'x'. Patches: jira_abe_2752_dial_fo_options.patch uploaded by rmudgett (license 664) Tested by: rmudgett JIRA ABE-2752 JIRA SWP-3096 .......... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@311295 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	ef01ba5ff2	This fixes a nasty chanspy bug which was causing a channel leak every time a spied on channel made a call. In addition to the above, it makes certain channel destruction occurs so that applications don't get stuck waiting for datastore destruction while monitored by chanspy. (closes issue #18742) Reported by: jkister Tested by: jkister, jcovert, jrose Review: http://reviewboard.digium.internal/r/106/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@311197 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Tilghman Lesher	15641c348e	Merged revisions 310141 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r310141 \| tilghman \| 2011-03-09 23:51:37 -0600 (Wed, 09 Mar 2011) \| 12 lines Merged revisions 310140 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r310140 \| tilghman \| 2011-03-09 23:38:44 -0600 (Wed, 09 Mar 2011) \| 5 lines Initialize column size to 0 to deal with a potential UnixODBC bug on 64-bit systems. (closes issue #18295) Reported by: pruiz ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@310142 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	4ad0ddf5e3	Merged revisions 309857 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r309857 \| jrose \| 2011-03-07 16:04:44 -0600 (Mon, 07 Mar 2011) \| 15 lines Merged revisions 309856 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r309856 \| jrose \| 2011-03-07 16:02:12 -0600 (Mon, 07 Mar 2011) \| 8 lines Bug fix for MixMonitor involving filenames with '.' not in the extension Closes issue #18391) Reported by: pabelanger Patches: bugfix.patch uploaded by jrose (license 1225) Tested by: jrose ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@309858 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
David Ruggles	d5e1774082	Merged revisions 309356 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r309356 \| diruggles \| 2011-03-03 19:42:28 -0500 (Thu, 03 Mar 2011) \| 16 lines Merged revisions 309355 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r309355 \| diruggles \| 2011-03-03 19:34:13 -0500 (Thu, 03 Mar 2011) \| 9 lines fix small memory leak fix small memory leak caused by a string allocation that wasn't freed (closes issue #18907) Reported by: andy11 Patches: asterisk_trunk-app_externalivr-leak.patch uploaded by andy11 (license 1224) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@309403 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jason Parker	c8ef3e081b	Merged revisions 308007 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r308007 \| qwell \| 2011-02-15 17:33:24 -0600 (Tue, 15 Feb 2011) \| 17 lines Merged revisions 308002 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r308002 \| qwell \| 2011-02-15 17:32:20 -0600 (Tue, 15 Feb 2011) \| 10 lines Fix regression that changed behavior of queues when ringing a queue member. This reverts r298596, which was to fix a highly bizarre and contrived issue with a queue member that called into his own queue being transferred back into his own queue. I couldn't reproduce that issue in any way. I think one of the other recent transfer fixes actually fixed this. (closes issue #18747) Reported by: vrban ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@308010 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	227c620866	Don't crash when forcing caller id. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@307962 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Tilghman Lesher	ff43beaa2d	Calling a gosub routine defined in AEL from Dial/Queue ceased to work. A bug in AEL did not distinguish between the "s" extension generated by AEL and an "s" extension that was required to exist by the chan_dahdi (or another channel) that was not supplied with a starting extension. Therefore, AEL made incorrect assumptions about what commands were permissable in the context. This was fixed by making AEL generate a different extension name. However, Dial and Queue make additional assumptions about the name of the default gosub extension. Therefore, they needed to be brought into line with a "macro" rendered by AEL (as a gosub), without breaking traditional dialplans written without the aid of AEL. Related to (issue #18480) Reported by: nivek (closes issue #18729) Reported by: kkm Patches: 20110209__issue18729.diff.txt uploaded by tilghman (license 14) 018729-dial-queue-gosub-try3.patch uploaded by kkm (license 888) Tested by: kkm git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@307750 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jeff Peeler	49c4800686	Merged revisions 306966 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r306966 \| jpeeler \| 2011-02-08 13:41:21 -0600 (Tue, 08 Feb 2011) \| 9 lines Merged revisions 306965 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r306965 \| jpeeler \| 2011-02-08 13:40:58 -0600 (Tue, 08 Feb 2011) \| 1 line fix this line again ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@306967 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jeff Peeler	dad67ad1a4	Merged revisions 306961 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r306961 \| jpeeler \| 2011-02-08 13:25:10 -0600 (Tue, 08 Feb 2011) \| 15 lines Merged revisions 306960 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r306960 \| jpeeler \| 2011-02-08 13:18:50 -0600 (Tue, 08 Feb 2011) \| 9 lines Backup file storing message duration is not used with IMAP_STORAGE, remove code. The message duration is stored in the body of the email when using IMAP_STORAGE, so nothing needs to happen with the backup file. (closes issue #18718) Reported by: kerframil ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@306962 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jeff Peeler	59502582b3	Merged revisions 306865 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r306865 \| jpeeler \| 2011-02-08 10:21:25 -0600 (Tue, 08 Feb 2011) \| 9 lines Merged revisions 306864 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r306864 \| jpeeler \| 2011-02-08 10:19:17 -0600 (Tue, 08 Feb 2011) \| 1 line make this safer and fully correct, pointed out by Steve Davis ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@306866 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jason Parker	e8bd6696b5	Merged revisions 306346 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r306346 \| qwell \| 2011-02-04 13:21:43 -0600 (Fri, 04 Feb 2011) \| 9 lines Don't fallthrough to 'unknown' in the 'ringing' case. This could cause improper exits from the queue. (closes issue #18499) Reported by: zaltar Patches: app_queue.patch uploaded by zaltar (license 1148) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@306356 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	cee1db213b	Don't send redirecting updates to the caller if the dialplan forked the call. Each fork in the dial could be redirected and confuse the caller. For ISDN the DivLeg1 and DivLeg3 messages would get confused because ISDN redirects calls in sequence not in parallel. * Also fixed a formatting inconsistency in app_dial.c and make a warning message more useful about what frame type could not be written. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@306324 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	a785544090	Merged revisions 305889 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r305889 \| rmudgett \| 2011-02-02 18:15:07 -0600 (Wed, 02 Feb 2011) \| 17 lines Merged revisions 305888 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r305888 \| rmudgett \| 2011-02-02 18:02:43 -0600 (Wed, 02 Feb 2011) \| 8 lines Minor AST_FRAME_TEXT related issues. * Include the null terminator in the buffer length. When the frame is queued it is copied. If the null terminator is not part of the frame buffer length, the receiver could see garbage appended onto it. * Add channel lock protection with ast_sendtext(). * Fixed AMI SendText action ast_sendtext() return value check. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@305923 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Andrew Latham	69e83f1a72	Replacing doc/* and asterisk.pdf with wiki links Adding links to http(s)://wiki.asterisk.org git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@305838 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jason Parker	1a5122534c	Merged revisions 305253 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r305253 \| qwell \| 2011-01-31 16:59:34 -0600 (Mon, 31 Jan 2011) \| 17 lines Merged revisions 305252 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r305252 \| qwell \| 2011-01-31 16:56:54 -0600 (Mon, 31 Jan 2011) \| 10 lines Prevent a crash when dialing a technology with no destination (ex: Dial(SIP/)) chan_iax2 and other channel drivers already had code to prevent this. The attempt that app_dial was making to prevent it was not correct, so I fixed that. (closes issue #18371) Reported by: gbour Patches: 18371.patch uploaded by gbour (license 1162) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@305254 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	b27fc05f06	Merged revisions 304978 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r304978 \| tilghman \| 2011-01-31 01:25:14 -0600 (Mon, 31 Jan 2011) \| 9 lines Merged revisions 304952 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r304952 \| tilghman \| 2011-01-31 00:54:45 -0600 (Mon, 31 Jan 2011) \| 2 lines Fix compilation when ODBC_STORAGE is defined. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@304985 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Andrew Latham	b7d7fc94c2	Add Function and Application Relationships to documentation Add and extend the see-also sections to the documentation for applications and functions in an effort to expand the online documentation of the wiki. Also check for and update any links to moved documentation in the doc folder. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@304908 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Sean Bright	b0c9f29c72	Merged revisions 304776 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r304776 \| seanbright \| 2011-01-29 13:08:14 -0500 (Sat, 29 Jan 2011) \| 15 lines If we fail to allocate our announcement objects, make sure we don't leak objects. The majority of this patch was committed already in r304726 and r304729. (issue #18225) Reported by: kenji (issue #18444) Reported by: junky (closes issue #18343) Reported by: kobaz Patches: meetme-refs.diff uploaded by kobaz (license 834) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@304777 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Sean Bright	07bbfff4eb	Merged revisions 304773 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r304773 \| seanbright \| 2011-01-29 12:51:28 -0500 (Sat, 29 Jan 2011) \| 9 lines When we pass the S() or L() options to MeetMe, make sure that we honor C as well. Without this patch, if the user was kicked from the conference via the S() or L() mechanism, we would just hang up on them even if we also passed C (continue in dialplan when kicked). With this patch we honor the C flag in those cases. (closes issue #17317) Reported by: var ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@304774 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Sean Bright	4ba774c116	Merged revisions 304729 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r304729 \| seanbright \| 2011-01-29 12:01:51 -0500 (Sat, 29 Jan 2011) \| 15 lines Make sure that we unref the correct object when ejecting the most recent caller. Currently, when we kick the last user to enter, we decrement our own reference count which results in a crash when we kick another user or when we exit the conference ourselves. This will fix #18225 in 1.8 and trunk, but that particular bug does not exist in 1.6.2. (closes issue #18225) Reported by: kenji Patches: issue18225.patch uploaded by seanbright (license 71) Tested by: seanbright ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@304730 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Sean Bright	05116e68f4	Merged revisions 304726 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r304726 \| seanbright \| 2011-01-29 11:26:57 -0500 (Sat, 29 Jan 2011) \| 9 lines Fix user reference leak in MeetMe. We were unlinking the user from the conferences user container, but not decrementing the reference count of the user as well, resulting in a leak. (closes issue #18444) Reported by: junky Tested by: seanbright ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@304727 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Sean Bright	6f4332b4cc	Merged revisions 304659,304682 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r304659 \| seanbright \| 2011-01-28 16:22:09 -0500 (Fri, 28 Jan 2011) \| 5 lines Don't leak references if we can't create a pseudo channel for mixing in MeetMe. If there was a problem allocating a pseudo channel when building our meetme, we weren't destroying our user container or destroying the mutexes that we created. ........ r304682 \| seanbright \| 2011-01-28 17:38:05 -0500 (Fri, 28 Jan 2011) \| 2 lines Revert part of the previous commit that snuck in. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@304683 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	b18db77287	Merged revisions 303677 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r303677 \| jpeeler \| 2011-01-25 10:59:28 -0600 (Tue, 25 Jan 2011) \| 26 lines Merged revisions 303676 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r303676 \| jpeeler \| 2011-01-25 10:58:29 -0600 (Tue, 25 Jan 2011) \| 20 lines Fix voicemail sequencing for file based storage. A previous change was made to account for when the number of voicemail messages exceeds the max limit to be handled properly, but it caused gaps in the messages to not be properly handled. This has now been resolved. In later non 1.4 branches, it appears that resequencing wasn't even occurring due from what appears and accidental code removal. (closes issue #18498) Reported by: JJCinAZ Patches: bug18498v2.patch uploaded by jpeeler (license 325) (closes issue #18486) Reported by: bluefox Patches: bug18486.patch uploaded by jpeeler (license 325) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@303678 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	cfc893a5bc	Merged revisions 303548 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r303548 \| russell \| 2011-01-24 14:49:53 -0600 (Mon, 24 Jan 2011) \| 38 lines Merged revisions 303546 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r303546 \| russell \| 2011-01-24 14:32:21 -0600 (Mon, 24 Jan 2011) \| 31 lines Fix channel redirect out of MeetMe() and other issues with channel softhangup. Mantis issue #18585 reports that a channel redirect out of MeetMe() stopped working properly. This issue includes a patch that resolves the issue by removing a call to ast_check_hangup() from app_meetme.c. I left that in my patch, as it doesn't need to be there. However, the rest of the patch fixes this problem with or without the change to app_meetme. The key difference between what happens before and after this patch is the effect of the END_OF_Q control frame. After END_OF_Q is hit in ast_read(), ast_read() will return NULL. With the ast_check_hangup() removed, app_meetme sees this which causes it to exit as intended. Checking ast_check_hangup() caused app_meetme to exit earlier in the process, and the target of the redirect saw the condition where ast_read() returned NULL. Removing ast_check_hangup() works around the issue in app_meetme, but doesn't solve the issue if another application did the same thing. There are also other edge cases where if an application finishes at the same time that a redirect happens, the target of the redirect will think that the channel hung up. So, I made some changes in pbx.c to resolve it at a deeper level. There are already places that unset the SOFTHANGUP_ASYNCGOTO flag in an attempt to abort the hangup process. My patch extends this to remove the END_OF_Q frame from the channel's read queue, making the "abort hangup" more complete. This same technique was used in every place where a softhangup flag was cleared. (closes issue #18585) Reported by: oej Tested by: oej, wedhorn, russell Review: https://reviewboard.asterisk.org/r/1082/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@303549 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	309a50c3bd	Merged revisions 303008 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r303008 \| jpeeler \| 2011-01-20 11:07:44 -0600 (Thu, 20 Jan 2011) \| 14 lines Merged revisions 303007 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r303007 \| jpeeler \| 2011-01-20 11:04:08 -0600 (Thu, 20 Jan 2011) \| 8 lines Add new queue strategy to preserve behavior for when queue members moved to ao2. Add queue strategy called "rrordered" to mimic old behavior from when queue members were stored in a linked list. ABE-2707 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@303009 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	239ab4d4d3	Merged revisions 302920 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r302920 \| russell \| 2011-01-20 10:11:58 -0600 (Thu, 20 Jan 2011) \| 2 lines Resolve a compiler warning. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@302921 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Leif Madsen	5c7e815a99	Merged revisions 302917 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r302917 \| lmadsen \| 2011-01-20 09:42:05 -0600 (Thu, 20 Jan 2011) \| 8 lines Option L() is milliseconds, not seconds. > Change the verbose output of option L() to say milliseconds and not seconds > as the value is in milliseconds. > > (closes issue #18264) > Reported by: jacco > Patches: > app_dial_patch.txt uploaded by lmadsen (license 10) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@302918 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Sean Bright	90f6681ad4	Merged revisions 302833 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r302833 \| seanbright \| 2011-01-19 18:47:22 -0500 (Wed, 19 Jan 2011) \| 7 lines Support greetingsfolder as documented in voicemail.conf.sample. (closes issue #17870) Reported by: edhorton Patches: __20100816-app_voicemail-greetingsfolder-support.txt uploaded by lmadsen (license 10) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@302834 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Paul Belanger	d5fa54dee5	Merged revisions 301176 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r301176 \| pabelanger \| 2011-01-08 16:58:24 -0500 (Sat, 08 Jan 2011) \| 7 lines Indicate log level argument for Log() is not optional (closes issue #18586) Reported by: kshumard Patches: app_verbose.c.patch uploaded by kshumard (license 92) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@301177 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jason Parker	a6b8200be6	Merged revisions 301089 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r301089 \| qwell \| 2011-01-07 14:52:00 -0600 (Fri, 07 Jan 2011) \| 8 lines Initialize useropts/adminopts in case there is no column in the realtime DB. (closes issue #18182) Reported by: dimas Patches: v1-18182.patch uploaded by dimas (license 88) Tested by: dimas ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@301090 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	4b0d83c5e3	Merged revisions 301046 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r301046 \| jpeeler \| 2011-01-07 13:57:42 -0600 (Fri, 07 Jan 2011) \| 8 lines Fix regression causing forwarding voicemails to not work with file storage. I had actually already fixed this in 295200 in 1.4 and thought it wasn't missing in the other branches for some reason. (closes issue #18358) Reported by: cabal95 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@301047 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	908b3848d0	Merged revisions 300951 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r300951 \| jpeeler \| 2011-01-07 11:23:37 -0600 (Fri, 07 Jan 2011) \| 14 lines Merged revisions 300918 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r300918 \| jpeeler \| 2011-01-07 11:13:21 -0600 (Fri, 07 Jan 2011) \| 7 lines Ensure good bye prompt in voicemail is played at the correct time. Specifically in the case of timing out but not leaving voicemail nothing should be heard. And when leaving voicemail it should be heard. ABE-2647 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@300955 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	d514f9c3a8	Quote arguments, just in case there's a space in a pathname. (Diagnosed by pabelanger on #asterisk-dev, fixed by me.) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@299989 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Paul Belanger	1c673d24ed	Merged revisions 299864 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r299864 \| pabelanger \| 2010-12-28 13:51:13 -0500 (Tue, 28 Dec 2010) \| 2 lines Documentation typo ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@299865 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	59eff79358	Merged revisions 298684 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r298684 \| jpeeler \| 2010-12-16 17:30:59 -0600 (Thu, 16 Dec 2010) \| 9 lines Merged revisions 298683 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r298683 \| jpeeler \| 2010-12-16 17:29:30 -0600 (Thu, 16 Dec 2010) \| 2 lines After recording only silence for a voicemail prepending, restore backup files. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@298685 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	b064838468	Merged revisions 298597 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r298597 \| jpeeler \| 2010-12-16 14:49:33 -0600 (Thu, 16 Dec 2010) \| 14 lines Merged revisions 298596 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r298596 \| jpeeler \| 2010-12-16 14:46:52 -0600 (Thu, 16 Dec 2010) \| 7 lines Fix improper hangup when doing an attended transfer to queue. Had to indicate ringing in wait_for_answer so the attended transfer code would not try and hang up the local channel it created, which would kill the call. ABE-2624 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@298598 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	461c3de2ed	Merged revisions 297713 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r297713 \| tilghman \| 2010-12-06 18:21:50 -0600 (Mon, 06 Dec 2010) \| 15 lines Merged revisions 297689 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r297689 \| tilghman \| 2010-12-06 18:07:37 -0600 (Mon, 06 Dec 2010) \| 8 lines Don't create a Local channel if the target extension does not exist. (closes issue #18126) Reported by: junky Patches: followme.diff uploaded by junky (license 177) (partially restructured by me to avoid a possible memory leak) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@297733 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	3433890c9a	Merged revisions 297229 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r297229 \| russell \| 2010-12-02 07:16:47 -0600 (Thu, 02 Dec 2010) \| 13 lines Merged revisions 297228 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r297228 \| russell \| 2010-12-02 07:16:15 -0600 (Thu, 02 Dec 2010) \| 6 lines Add "DAHDI" to a couple of app_meetme error messages. This is in response to some questions on IRC. To the user, there was nothing that made it obvious that this error had anything to do with DAHDI not being loaded. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@297245 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	38b81d2772	Merged revisions 296869 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r296869 \| jpeeler \| 2010-11-30 18:24:58 -0600 (Tue, 30 Nov 2010) \| 11 lines Merged revisions 296868 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r296868 \| jpeeler \| 2010-11-30 18:23:19 -0600 (Tue, 30 Nov 2010) \| 4 lines Properly restore backup information file when hanging up during message prepending. ABE-2654 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@296870 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	82ee0bc14e	DOC: Conference number can be omitted; if omitted, all users in a meetme are listed. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@296787 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	b4f92dec2c	Merged revisions 296466 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r296466 \| tilghman \| 2010-11-27 04:39:01 -0600 (Sat, 27 Nov 2010) \| 5 lines 18 characters is too short for most date/times (20 is the usual, but we add more in case of greater precision). (closes issue #18369) Reported by: tnakonz ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@296467 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	30a7e71c27	Merged revisions 296001 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r296001 \| russell \| 2010-11-24 11:03:16 -0600 (Wed, 24 Nov 2010) \| 45 lines Merged revisions 296000 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r296000 \| russell \| 2010-11-24 10:48:39 -0600 (Wed, 24 Nov 2010) \| 38 lines Handle failures building translation paths more effectively. The problem scenario occurred on a heavily loaded system that was using the codec_dahdi module and exceeded the hardware transcoding capacity. The failure mode at that point was not good. The report came in to us as an Asterisk lock-up. The "core show locks" shows a ton of threads locked up (but no obvious deadlock). Upon deeper investigation, when the system is in this state, the CPU was maxed out. The CPU was being consumed by the Asterisk logger spewing messages on every audio frame for calls set up after transcoder capacity was reached. The purpose of this patch is to make Asterisk handle failures to create a translation path in a more graceful manner. If we can't translate, then the call just needs to be dropped, as it's not going to work. These are the changes: 1) In set_format() of channel.c (which is called by set_read_format() and set_write_format()), it was ignoring if ast_translator_build_path() failed and returned NULL. It now pays attention to that case and returns a result reflecting failure. With this change in place, the bridging code will immediately detect a failure and end the bridge instead of proceeding to try to bridge frames that can't be translated and making channel drivers freak out by sending them frames in a format they weren't expecting. 2) In ast_indicate_data() of channel.c, failure of ast_playtones_start() was ignored. It is now reflected in the return value of the function. This didn't turn out to have any affect on the bug, but seemed like a good change to leave in. 3) In app_dial(), when only sending a call to a single endpoint, it will attempt to do some bridging of its own of early audio. It uses make_compatible() when it's going to do this. However, it ignored failure from make compatible. So, even with the fix from #1, if there was early audio going through app_dial, there would still be a period of invalid frames passing through. After detecting failure here, Dial() exits. ABE-2658 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@296002 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	c08103f033	Merged revisions 295843 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r295843 \| rmudgett \| 2010-11-22 13:28:23 -0600 (Mon, 22 Nov 2010) \| 53 lines Merged revisions 295790 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r295790 \| rmudgett \| 2010-11-22 12:46:26 -0600 (Mon, 22 Nov 2010) \| 46 lines The channel redirect function (CLI or AMI) hangs up the call instead of redirecting the call. To recreate the problem: 1) Party A calls Party B 2) Invoke CLI "channel redirect" command to redirect channel call leg associated with A. 3) All associated channels are hung up. Note that if the CLI command were done on the channel call leg associated with B it works. This regression was a result of the fix for issue #16946 (https://reviewboard.asterisk.org/r/740/). The regression affects all features that use an async goto to execute the dialplan because of an external event: Channel redirect, AMI redirect, SIP REFER, and FAX detection. The struct ast_channel._softhangup code is a mess. The variable is used for several purposes that do not necessarily result in the call being hung up. I have added doxygen comments to describe how the various _softhangup bits are used. I have corrected all the places where the variable was tested in a non-bit oriented manner. The primary fix is the new AST_CONTROL_END_OF_Q frame. It acts as a weak hangup request so the soft hangup requests that do not normally result in a hangup do not hangup. JIRA SWP-2470 JIRA SWP-2489 (closes issue #18171) Reported by: SantaFox (closes issue #18185) Reported by: kwemheuer (closes issue #18211) Reported by: zahir_koradia (closes issue #18230) Reported by: vmarrone (closes issue #18299) Reported by: mbrevda (closes issue #18322) Reported by: nerbos Review: https://reviewboard.asterisk.org/r/1013/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@295866 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Brett Bryant	ddb80391f6	Patch for deadlock from ordering issue between channel/queue locks in app_queue (set_queue_variables). (closes issue #18031) Reported by: rain Review: https://reviewboard.asterisk.org/r/1018/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@295670 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	f1abd401b9	Merged revisions 294910 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r294910 \| jpeeler \| 2010-11-12 15:14:23 -0600 (Fri, 12 Nov 2010) \| 4 lines Return correct error code if lock path fails. The recent changes to open_mailbox actually caused it to be fixed, but let's be consistent. Reported by alecdavis in asterisk-dev. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@294911 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	06ac20454e	Merged revisions 294904 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r294904 \| jpeeler \| 2010-11-12 14:51:15 -0600 (Fri, 12 Nov 2010) \| 23 lines Merged revisions 294903 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r294903 \| jpeeler \| 2010-11-12 14:49:09 -0600 (Fri, 12 Nov 2010) \| 16 lines Fix regression causing abort in voicemail after opening a mailbox with no mesgs. In order to be more safe, some error handling code was changed to respect more error conditions including the potential memory allocation failure for deleted and heard message tracking introduced in 293004. However, last_message_index returns -1 for zero messages (perhaps as expected) and was triggering the stricter error checking. Because last_message_index is only called directly in one place, just return 0 from open_mailbox (for file based storage) when no messages are detected unless a real error has occurred. (closes issue #18240) Reported by: leobrown Patches: bug18240.1-6-2.diff.txt uploaded by alecdavis (license 585) Tested by: pabelanger ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@294905 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	6cbda6ed92	Merged revisions 293118 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r293118 \| jpeeler \| 2010-10-26 13:33:24 -0500 (Tue, 26 Oct 2010) \| 36 lines Merged revisions 293004 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r293004 \| jpeeler \| 2010-10-25 17:55:28 -0500 (Mon, 25 Oct 2010) \| 29 lines Fix inprocess_container in voicemail to correctly restrict max messages. The comparison function logic was off, so the number of sessions for a given mailbox were not being incremented properly. This problem caused the maximum number of messages per folder to not be respected when simultaneously leaving multiple voicemails just below the threshold. These problems should be fixed by the above, but just in case: Fixed resequence_mailbox to rely on the actual number of detected number of files in a directory rather than just assuming only 10 messages more than the maximum had been left. Also if more messages than the maximum are deleted they are actually removed now. The second purpose of this commit should have been separated out probably, but is related to the above. Again, if the number of messages in a given voicemail folder exceeds the maximum set limit make sure to allocate enough space for the deleted and heard index tracking array. A few random fixes: There was a forgotten decrement of the inprocess count in imap_store_file. When using IMAP storage, do not look in the directory where file based storage messages may still reside and influence the message count. Ensure to use only the first format in sendmail. ABE-2516 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@293119 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Paul Belanger	83ed33746e	Application not properly unregister in voicemail (closes issue #18128) Reported by: junky Patches: vm_unregister.diff uploaded by junky (license 177) Tested by: pabelanger, lmadsen git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@292436 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Paul Belanger	d4a74bd2ae	Merged revisions 292412 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r292412 \| pabelanger \| 2010-10-20 20:05:45 -0400 (Wed, 20 Oct 2010) \| 17 lines Merged revisions 292411 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r292411 \| pabelanger \| 2010-10-20 20:00:51 -0400 (Wed, 20 Oct 2010) \| 10 lines Record priv-recordintro as sln, not gsm This removes the gsm->sln step when transcoding priv-recordintro. (closes issue #18176) Reported by: pabelanger Patches: chan_sip.diff uploaded by pabelanger (license 224) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@292413 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago

1 2 3 4 5 ...

3863 Commits (7cf95da39acad3057a8976b3850bf6693e95b2ee)