asterisk

Commit Graph

Author	SHA1	Message	Date
Richard Mudgett	ba26e7e249	Coverity Report: Fix issues for error type REVERSE_INULL (deprecated modules) * Fix only issue pointed out by deprecated_REVERSE_INULL.txt for app_meetme.c in find_user(). * Change use of %i to %d in sscanf() in find_user(). The use of %i gives unexpected parsing because it can accept hex, octal, and decimal integer formats. * Changed other uses of %i in app_meetme() to use %d for consistency. (issue ASTERISK-19648) Reported by: Matt Jordan git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@367906 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	680b627906	Send relative path named recordings to the meetme directory instead of sounds Prior to this patch, no effort was made to parse the path name to determine a proper destination for recordings of MeetMe's r option. This fixes that. Review: https://reviewboard.asterisk.org/r/1846/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@362079 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	304af5d7cc	Fix memory leak when using MeetMeAdmin 'e' option with user specified A memory leak/reference counting leak occurs if the MeetMeAdmin 'e' command (eject last user that joined) is used in conjunction with a specified user. Regardless of the command being executed, if a user is specified for the command, MeetMeAdmin will look up that user. Because the 'e' option kicks the last user that joined, as opposed to the one specified, the reference to the user specified by the command would be leaked when the user variable was assigned to the last user that joined. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@361558 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	3560b4a9b5	Fix MusicOnHold in MeetMe so that it always uses the class if it's been defined There were a few instances of restarting music on hold in meetme that would cause Asterisk to revert to the default class of music on hold for no adequate reason. Review: https://reviewboard.asterisk.org/r/1844/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@361269 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	abd7fa4476	Make the MeetMeAdmin N command (mute all nonadmins) not mute admins (Closes Issue ASTERISK-19335) Reported by: Johan Wilfer Review: https://reviewboard.asterisk.org/r/1843/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@361090 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	cf9df098bd	Prevent SLA settings from getting wiped out on reload If SLA was reloaded without the config file being changed, current settings got wiped out before the SLA reload code decided it wasn't going to reload the file since nothing was changed. Moving the settings reset later in the reload process fixes this. (closes issue AST-744) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@350023 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	547aec88fb	Fix: Meetme recording variables from realtime DB use null entries over channel variables Meetme would attempt to substitute the realtime values of RECORDING_FILE and RECORDING_FORMAT from the meetme db entry instead of using the channel variable set for those variables in spite of those database entries being NULL or even lacking a column to represent them. (closes issue ASTERISK-18873) Reported by: Byron Clark Patches: ASTERISK-18873-1.patch uploaded by Byron Clark (license 6157) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@347369 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	b17a694234	Fix another incorrect case with meetme's PIN logic and add documentation This fixes an issue where a user of a dynamic conference was asked for a PIN twice. This also adds documentation to assist in future modifications to the piece of code responsible for PIN checking. (closes issue AST-670) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@344439 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Kinsey Moore	8e2a8d16df	Fix pin parameter behavior regression in MeetMe The last time this code was touched (by me), a subtlety was missed based on the difference between needing to check a pin's validity and the need to prompt for a pin. (closes issue ASTERISK-18488) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@344102 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Kevin P. Fleming	8daf83a53b	Modify comments in MeetMe application documentation about DAHDI. The MeetMe application documentation has some comments about usage of DAHDI, and they were a bit outdated relative to modern DAHDI releases. This patch changes the comment to just tell the user that a functional DAHDI timing source is required, and no longer mention 'dahdi_dummy', since that module does not exist in current DAHDI releases. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@342990 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	4051609b5c	Update MeetMe p and X option documentation when interacting with the s option. ASTERISK-12175 changed the p and X options to not interfere with the s option when they are used together. It makes more sense for the s option to have priority for the DTMF '*' key since it cannot change its activation code. Otherwise, you could not use option s with the p or X options. JIRA AST-671 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@340470 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Matthew Jordan	f13c3b3fd2	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/1.8@337118 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Paul Belanger	28952b7ea5	Meetme should have 'core' support level (closes issue ASTERISK-18542) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@335714 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jonathan Rose	427d1167cd	Fixes 32bit compilation warnings brought on by 331634 in app_dial and app_meetme git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@331635 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jason Parker	0688f632dc	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/1.8@331578 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Kinsey Moore	9769bb7d34	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/1.8@328770 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Leif Madsen	fc0ea9d188	Revert changes to defaultenabled state for modules in Asterisk 1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@328446 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Leif Madsen	d4938a111e	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/1.8@328209 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Leif Madsen	211af7820d	Fix typo in documentation. Pointed out by Vlad Povorozniuc git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@324176 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	59a41188a8	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/branches/1.8@320823 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	bf153ba6a4	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/branches/1.8@320237 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	10e40660c7	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/branches/1.8@317969 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	91c8e4d297	Fix some more "set but unused" compiler warnings. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@317967 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	63579a892c	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/branches/1.8@316831 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Sean Bright	aa43b12c24	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/branches/1.8@316476 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jonathan Rose	f6f5340777	Merged revisions 312762 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r312762 \| jrose \| 2011-04-05 09:11:36 -0500 (Tue, 05 Apr 2011) \| 1 line Backporting trunk change to add verbosity to 'L' option in meetme ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@312765 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Brett Bryant	51ce432d07	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/branches/1.8@311615 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	a00e99ec56	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/branches/1.8@311497 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
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
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
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
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
Brett Bryant	71cbbd60de	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/branches/1.8@287760 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Brett Bryant	69e4f34914	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/branches/1.8@285533 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jean Galarneau	ae2e66e707	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/branches/1.8@280346 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	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	16 years ago
Tilghman Lesher	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	16 years ago
Richard Mudgett	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	16 years ago
Jeff Peeler	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	16 years ago
Eliel C. Sardanons	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	16 years ago
Tilghman Lesher	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	16 years ago
Tilghman Lesher	d6011adab4	Weird, no output and Bamboo still fails... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275227 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	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	16 years ago
Tilghman Lesher	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	16 years ago

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