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463 Commits (283f63c594d0c4327c51fdae0587a6a9f1cb15d3)
Author | SHA1 | Message | Date |
---|---|---|---|
|
cf4d5e585a |
Revert previous commit
New feature should be added into trunk, unfortunately it is too late for the Asterisk 10 branch. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@337640 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
944cdaa94d |
Merged revisions 337118 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r337118 | mjordan | 2011-09-20 17:38:54 -0500 (Tue, 20 Sep 2011) | 21 lines Fix for incorrect voicemail duration in external notifications This patch fixes an issue where the voicemail duration was being reported with a duration significantly less than the actual sound file duration. Voicemails that contained mostly silence were reporting the duration of only the sound in the file, as opposed to the duration of the file with the silence. This patch fixes this by having two durations reported in the __ast_play_and_record family of functions - the sound_duration and the actual duration of the file. The sound_duration, which is optional, now reports the duration of the sound in the file, while the actual full duration of the file is reported in the duration parameter. This allows the voicemail applications to use the sound_duration for minimum duration checking, while reporting the full duration to external parties if the voicemail is kept. (issue ASTERISK-2234) (closes issue ASTERISK-16981) Reported by: Mary Ciuciu, Byron Clark, Brad House, Karsten Wemheuer, KevinH Tested by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1443 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@337120 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
9795ec3ef0 |
Meetme: Introducing a new option "k" to kill a conference if there's only a single member left.
When using Meetme as a modular call bridge from third party applications, it's handy to make it behave like a normal call bridge. When the second to last person exists, the last person will be kicked out of the conference when this option is enabled. (closes issue ASTERISK-18234) Review: https://reviewboard.asterisk.org/r/1376/ Patch by oej, sponsored by ClearIT, Solna, Sweden git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@336042 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
c64bc780da |
Merged revisions 331635 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r331635 | jrose | 2011-08-12 10:49:17 -0500 (Fri, 12 Aug 2011) | 1 line Fixes 32bit compilation warnings brought on by 331634 in app_dial and app_meetme ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@331644 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
369ab7e566 |
Merged revisions 331578 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r331578 | qwell | 2011-08-11 16:46:39 -0500 (Thu, 11 Aug 2011) | 6 lines Use proper values for 64-bit option flags. Also, reusing bits es no bueno, so change the value of a duplicate. (issue ASTERISK-18239) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@331579 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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Merged revisions 328770 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r328770 | kmoore | 2011-07-19 10:43:32 -0500 (Tue, 19 Jul 2011) | 11 lines MeetMe requests a PIN twice in some circumstances If a call to MeetMe includes both the dynamic(D) and always request PIN(P) options, MeetMe will ask for the PIN two times: once for creating the conference and once for entering the conference. This behavior was introduced in rev 311616 when adding the CONFFLAG_ALWAYSPROMPT option to the logic branch controlling PIN entry for joining a conference. (closes AST-601) Review: https://reviewboard.asterisk.org/r/1305/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.10@328771 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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Merged revisions 328209 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r328209 | lmadsen | 2011-07-14 16:13:06 -0400 (Thu, 14 Jul 2011) | 6 lines Introduce <support_level> tags in MODULEINFO. This change introduces MODULEINFO into many modules in Asterisk in order to show the community support level for those modules. This is used by changes committed to menuselect by Russell Bryant recently (r917 in menuselect). More information about the support level types and what they mean is available on the wiki at https://wiki.asterisk.org/wiki/display/AST/Asterisk+Module+Support+States ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.10@328247 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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Merged revisions 324176 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r324176 | lmadsen | 2011-06-17 14:38:40 -0400 (Fri, 17 Jun 2011) | 2 lines Fix typo in documentation. Pointed out by Vlad Povorozniuc ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@324177 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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0096238b52 |
Merged revisions 320823 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r320823 | rmudgett | 2011-05-25 12:06:38 -0500 (Wed, 25 May 2011) | 18 lines The AMI Newstate event contains different information between v1.4 and v1.8. The addition of connected line support in v1.8 changes the behavior of the channel caller ID somewhat. The channel caller ID value no longer time shares with the connected line ID on outgoing call legs. The timing of some AMI events/responses output the connected line ID as caller ID. These party ID's are now separate. * The ConnectedLineNum and ConnectedLineName headers were added to many AMI events/responses if the CallerIDNum/CallerIDName headers were also present. (closes issue #18252) Reported by: gje Tested by: rmudgett Review: https://reviewboard.asterisk.org/r/1227/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@320825 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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091fcbce3f |
Merged revisions 320237 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r320237 | rmudgett | 2011-05-20 15:49:03 -0500 (Fri, 20 May 2011) | 27 lines Merged revisions 320236 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r320236 | rmudgett | 2011-05-20 15:44:54 -0500 (Fri, 20 May 2011) | 20 lines Merged revisions 320235 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r320235 | rmudgett | 2011-05-20 15:38:22 -0500 (Fri, 20 May 2011) | 13 lines The meetme CLI command completion leaves conferences mutex locked. When issuing a meetme kick CLI command and an invalid (non-existent) conference number is specified, pressing Tab leaves the conferences mutex locked and, therefore, all conferences deadlock. Add missing unlock. (closes issue #19336) Reported by: zvision Patches: app_meetme.diff uploaded by zvision (license 798) ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@320238 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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Merged revisions 317969 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r317969 | russell | 2011-05-06 16:49:01 -0500 (Fri, 06 May 2011) | 10 lines Use the right variable to print the time in a debug message. The original patch also increased some buffer sizes, but that was already done in this version. (closes issue #17034) Reported by: sysreq Patches: asterisk-issue-17034.patch uploaded by sysreq (license 1009) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317970 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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Merged revisions 317967 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r317967 | russell | 2011-05-06 16:38:54 -0500 (Fri, 06 May 2011) | 2 lines Fix some more "set but unused" compiler warnings. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317968 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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Merged revisions 316831 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r316831 | rmudgett | 2011-05-04 13:51:40 -0500 (Wed, 04 May 2011) | 9 lines Wait for leader with Music On Hold allows crosstalk between participants. Parenthesis in the wrong position. Regression from issue #14365 when expanding conference flags to use 64 bits. (closes issue #18418) Reported by: MrHanMan Tested by: rmudgett ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@316832 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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c596329564 |
Merged revisions 316476 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r316476 | seanbright | 2011-05-03 22:34:01 -0400 (Tue, 03 May 2011) | 17 lines Merged revisions 316475 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r316475 | seanbright | 2011-05-03 22:23:01 -0400 (Tue, 03 May 2011) | 10 lines Honor the C option to MeetMe when L is passed. This fixes a case that r304773 and friends missed. (closes issue #17317) Reported by: var Patches: meetme-continue-on-l_16218.diff uploaded by var (license 1227) Tested by: seanbright ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@316477 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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7c3d14957b |
Formatting change, remove red blobs
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@316054 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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0622568f15 |
Add explanation of strange flag setup in app_meetme (stolen from Mark's message to asterisk-dev)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@314158 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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5af547a619 |
Minor change to 'L' option for meetme to include some verb statements for the option.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@312756 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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c31d7b21ea |
Merged revisions 311615 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r311615 | bbryant | 2011-03-23 17:54:11 -0400 (Wed, 23 Mar 2011) | 8 lines This patch fixes a bug with MeetMe behavior where the 'P' option for always prompting for a pin is ignored for the first caller. (closes issue #18070) Reported by: mav3rick Review: https://reviewboard.asterisk.org/r/1132/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@311616 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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7902813301 |
Merged revisions 311497 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r311497 | dvossel | 2011-03-22 10:25:24 -0500 (Tue, 22 Mar 2011) | 9 lines Merged revisions 311496 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r311496 | dvossel | 2011-03-22 10:24:45 -0500 (Tue, 22 Mar 2011) | 2 lines Fixes memory leak in MeetMe AMI action ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@311498 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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8f7982f280 |
Add new manager action MeetmeListRooms.
From the submitter: I've added a new manager action to list only the active conferences on an Asterisk system. It shows the same data displayed when you run a 'meetme list' on the Asterisk CLI. (closes issue #17905) Reported by: rcasas Patches: app_meetme.c.patch uploaded by rcasas (license 641) Review: https://reviewboard.asterisk.org/r/874/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307359 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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3556e4c2d4 |
Replace ast_log(LOG_DEBUG, ...) with ast_debug()
(closes issue #18556) Reported by: kkm Review: https://reviewboard.asterisk.org/r/1071/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306258 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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c26c190711 |
Asterisk media architecture conversion - no more format bitfields
This patch is the foundation of an entire new way of looking at media in Asterisk. The code present in this patch is everything required to complete phase1 of my Media Architecture proposal. For more information about this project visit the link below. https://wiki.asterisk.org/wiki/display/AST/Media+Architecture+Proposal The primary function of this patch is to convert all the usages of format bitfields in Asterisk to use the new format and format_cap APIs. Functionally no change in behavior should be present in this patch. Thanks to twilson and russell for all the time they spent reviewing these changes. Review: https://reviewboard.asterisk.org/r/1083/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306010 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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cc2c9442f6 |
Merged revisions 304777 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r304777 | seanbright | 2011-01-29 13:09:37 -0500 (Sat, 29 Jan 2011) | 22 lines 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/trunk@304778 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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ed1ee072b8 |
Merged revisions 304774 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r304774 | seanbright | 2011-01-29 12:54:43 -0500 (Sat, 29 Jan 2011) | 16 lines 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/trunk@304775 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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e229e9f010 |
Merged revisions 304730 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r304730 | seanbright | 2011-01-29 12:15:27 -0500 (Sat, 29 Jan 2011) | 22 lines 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/trunk@304772 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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Merged revisions 304727 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r304727 | seanbright | 2011-01-29 11:28:27 -0500 (Sat, 29 Jan 2011) | 16 lines 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/trunk@304728 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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c5cf436a92 |
Merged revisions 304683 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r304683 | seanbright | 2011-01-28 17:54:23 -0500 (Fri, 28 Jan 2011) | 16 lines 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/trunk@304684 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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092134399c |
Merged revisions 303549 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r303549 | russell | 2011-01-24 14:51:37 -0600 (Mon, 24 Jan 2011) | 45 lines 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/trunk@303551 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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Merged revisions 301090 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r301090 | qwell | 2011-01-07 14:53:02 -0600 (Fri, 07 Jan 2011) | 15 lines Merged revisions 301089 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r301089 | qwell | 2011-01-07 14:52:00 -0600 (Fri, 07 Jan 2011) | 8 lines Initialize useropts/adminopts in case there is no column in the realtime DB. (closes issue #18182) Reported by: dimas Patches: v1-18182.patch uploaded by dimas (license 88) Tested by: dimas ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@301091 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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Merged revisions 297245 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r297245 | russell | 2010-12-02 07:20:19 -0600 (Thu, 02 Dec 2010) | 20 lines 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/trunk@297248 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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Merged revisions 296787 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r296787 | tilghman | 2010-11-30 13:12:48 -0600 (Tue, 30 Nov 2010) | 2 lines DOC: Conference number can be omitted; if omitted, all users in a meetme are listed. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@296788 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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Merged revisions 296467 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r296467 | tilghman | 2010-11-27 04:40:22 -0600 (Sat, 27 Nov 2010) | 12 lines 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/trunk@296468 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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935930d8a3 |
Meetme use voicemail greet for join/leave announce
Added option v(mailbox@[context]) which tells MeetMe where to look for a users greet file. If one does not exist it clears the v option and defers to the functionality of i/I as/if set by the MeetMe() command. Review: https://reviewboard.asterisk.org/r/1009/ (closes issue #18297) Reported by: parisioa Patches: meetme_final_patch_v.diff uploaded by parisioa (license 1153) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@296249 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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e8de16e970 |
Merged revisions 287760 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r287760 | bbryant | 2010-09-20 20:00:23 -0400 (Mon, 20 Sep 2010) | 30 lines Merged revisions 287759 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r287759 | bbryant | 2010-09-20 19:58:26 -0400 (Mon, 20 Sep 2010) | 23 lines Merged revisions 287758 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r287758 | bbryant | 2010-09-20 19:57:08 -0400 (Mon, 20 Sep 2010) | 16 lines Fix misvalidation of meetme pins in conjunction with the 'a' MeetMe flag. When using the 'a' MeetMe flag and having a user and admin pin setup for your conference, using the user pin would gain you admin priviledges. Also, when no user pin was set, an admin pin was, the 'a' MeetMe flag wasn't used, and the user tried to enter a conference then they were still prompted for a pin and forced to hit #. (closes issue #17908) Reported by: kuj Patches: pins_2.patch uploaded by kuj (license 1111) Tested by: kuj Review: [full review board URL with trailing slash] ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@287763 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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0c63db0483 |
Merged revisions 285533 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r285533 | bbryant | 2010-09-08 16:58:43 -0400 (Wed, 08 Sep 2010) | 15 lines Merged revisions 285532 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r285532 | bbryant | 2010-09-08 16:56:12 -0400 (Wed, 08 Sep 2010) | 8 lines Fixes a bug with MeetMe where after announcing the amount of time left in a conference, if music on hold was playing, it doesn't restart. (closes issue #17408) Reported by: sysreq Patches: asterisk-issue-17408_fixed.patch uploaded by sysreq (license 1009) Tested by: sysreq ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@285534 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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0a5c0dd75e |
Merged revisions 280346 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r280346 | jeang | 2010-07-29 11:07:16 -0500 (Thu, 29 Jul 2010) | 17 lines Merged revisions 280345 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r280345 | jeang | 2010-07-29 11:01:35 -0500 (Thu, 29 Jul 2010) | 10 lines Merged revisions 280341 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r280341 | jeang | 2010-07-29 10:52:31 -0500 (Thu, 29 Jul 2010) | 2 lines Fix a dsp structure leak occuring when a local channel is put into a meetme conference, then masquaraded away. ABE-2422 ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@280416 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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9bb8dc67e7 |
Ensure realtime conferences are treated the same as static conferences when trying to find an empty one.
Also, parse the useropts properly, when retrieving from realtime, and add them to the existing flags. (closes issue #17502) Reported by: kenji Patches: 20100720__issue17502.diff.txt uploaded by tilghman (license 14) Tested by: kenji git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278463 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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b4e18d5660 |
Add load priority order, such that preload becomes unnecessary in most cases
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278132 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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ec37ffbdaf |
ast_callerid restructuring
The purpose of this patch is to eliminate struct ast_callerid since it has turned into a miscellaneous collection of various party information. Eliminate struct ast_callerid and replace it with the following struct organization: struct ast_party_name { char *str; int char_set; int presentation; unsigned char valid; }; struct ast_party_number { char *str; int plan; int presentation; unsigned char valid; }; struct ast_party_subaddress { char *str; int type; unsigned char odd_even_indicator; unsigned char valid; }; struct ast_party_id { struct ast_party_name name; struct ast_party_number number; struct ast_party_subaddress subaddress; char *tag; }; struct ast_party_dialed { struct { char *str; int plan; } number; struct ast_party_subaddress subaddress; int transit_network_select; }; struct ast_party_caller { struct ast_party_id id; char *ani; int ani2; }; The new organization adds some new information as well. * The party name and number now have their own presentation value that can be manipulated independently. ISDN supplies the presentation value for the name and number at different times with the possibility that they could be different. * The party name and number now have a valid flag. Before this change the name or number string could be empty if the presentation were restricted. Most channel drivers assume that the name or number is then simply not available instead of indicating that the name or number was restricted. * The party name now has a character set value. SIP and Q.SIG have the ability to indicate what character set a name string is using so it could be presented properly. * The dialed party now has a numbering plan value that could be useful to have available. The various channel drivers will need to be updated to support the new core features as needed. They have simply been converted to supply current functionality at this time. The following items of note were either corrected or enhanced: * The CONNECTEDLINE() and REDIRECTING() dialplan functions were consolidated into func_callerid.c to share party id handling code. * CALLERPRES() is now deprecated because the name and number have their own presentation values. * Fixed app_alarmreceiver.c write_metadata(). The workstring[] could contain garbage. It also can only contain the caller id number so using ast_callerid_parse() on it is silly. There was also a typo in the CALLERNAME if test. * Fixed app_rpt.c using ast_callerid_parse() on the channel's caller id number string. ast_callerid_parse() alters the given buffer which in this case is the channel's caller id number string. Then using ast_shrink_phone_number() could alter it even more. * Fixed caller ID name and number memory leak in chan_usbradio.c. * Fixed uninitialized char arrays cid_num[] and cid_name[] in sig_analog.c. * Protected access to a caller channel with lock in chan_sip.c. * Clarified intent of code in app_meetme.c sla_ring_station() and dial_trunk(). Also made save all caller ID data instead of just the name and number strings. * Simplified cdr.c set_one_cid(). It hand coded the ast_callerid_merge() function. * Corrected some weirdness with app_privacy.c's use of caller presentation. Review: https://reviewboard.asterisk.org/r/702/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276347 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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6535a1d0ed |
Merged revisions 275773 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r275773 | jpeeler | 2010-07-12 15:34:51 -0500 (Mon, 12 Jul 2010) | 12 lines Make user removals and traversals thread safe in meetme. Race conditions present in meetme involving the user list where a lack of locking has the potential for a user to be removed during a traversal or as in the case of the reporter after checking if the list is empty could cause a crash. Fixing this was done by convering the userlist to an ao2 container. (closes issue #17390) Reported by: Vince Review: https://reviewboard.asterisk.org/r/746/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276074 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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7eafb1a763 |
When creating a conference for a unit test, it is not mandatory to open a
dahdi pseudo channel, so if we fail doing it, continue creating the conference. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275509 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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2fdf43f9fc |
Get more information about the Bamboo test failures
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275312 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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d6011adab4 |
Weird, no output and Bamboo still fails...
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275227 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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384681e182 |
Add some diagnostic feedback to our data tests
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275172 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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da8450323f |
Kill some startup warnings and errors and make some messages more helpful in tracking down the source.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275105 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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a1b89a6a50 |
Implement AstData API data providers as part of the GSOC 2010 project,
midterm evaluation. Review: https://reviewboard.asterisk.org/r/757/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274727 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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b840ef081e |
Merged revisions 273474 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r273474 | jpeeler | 2010-07-01 15:19:16 -0500 (Thu, 01 Jul 2010) | 14 lines Allow admin user to join conference without using admin mode and no user pin. Configuring the conference in meetme.conf like the following: conf => 2345,,6666 did not prompt for pin when used without admin mode. This meant that the conference could not be joined as an admin even if the user knew the correct pin. The original bug report was submitted claiming that the blank user pin should deny entry into the conference. I think a better way to handle this would be with a feature enhancement that used the following syntax: conf => 2345,X,6666 - where X denotes no acceptable pin allowed (closes issue #15704) Reported by: modelnine ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@273522 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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bd9ff2829e |
Merged revisions 273354 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r273354 | jpeeler | 2010-07-01 10:05:43 -0500 (Thu, 01 Jul 2010) | 12 lines Ensure channel placed in meetme in ringing state is properly hung up. An outgoing channel placed in meetme while still ringing which was then hung up would not exit meetme and the channel was not properly destroyed. Specifically checking for this scenario by looking at the appropriate control frames resolves the issue. (closes issue #15871) Reported by: Ivan Patches: meetme_congestion_trunk_v2.patch uploaded by Ivan (license 229) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@273355 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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90c850b5b1 |
Fix previous merge. ast_test_flag != ast_test_flag64
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272259 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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affec518d6 |
Merged revisions 272255 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r272255 | pabelanger | 2010-06-23 16:57:01 -0400 (Wed, 23 Jun 2010) | 12 lines First caller into a dynamic conference now enter pin once. If MeetMe is configured to use dynamic conference numbers, then the first caller (which creates the conference) had to enter the PIN number twice. (closes issue #15878) Reported by: shawkris Patches: issue15878.patch uploaded by pabelanger (license 224) Tested by: pabelanger ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272257 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |