asterisk

Commit Graph

Author	SHA1	Message	Date
Terry Wilson	ebaf59a656	Opaquification for ast_format structs in struct ast_channel Review: https://reviewboard.asterisk.org/r/1770/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356573 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	57f42bd74f	ast_channel opaquification of pointers and integral types Review: https://reviewboard.asterisk.org/r/1753/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356042 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	34c55e8e7c	Opaquify char * and char[] in ast_channel Review: https://reviewboard.asterisk.org/r/1733/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354968 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Joshua Colp	afdd96712c	Make the 'c' option to MeetMe work even if the 'q' option is used. (closes issue ASTERISK-17053) Reported by: justdave git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354084 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	99cae5b750	Opaquify channel stringfields Continue channel opaque-ification by wrapping all of the stringfields. Eventually, we will restrict what can actually set these variables, but the purpose for now is to hide the implementation and keep people from adding code that directly accesses the channel structure. Semantic changes will follow afterward. Review: https://reviewboard.asterisk.org/r/1661/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352348 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	04da92c379	Replace direct access to channel name with accessor functions There are many benefits to making the ast_channel an opaque handle, from increasing maintainability to presenting ways to kill masquerades. This patch kicks things off by taking things a field at a time, renaming the field to '__do_not_use_${fieldname}' and then writing setters/getters and converting the existing code to using them. When all fields are done, we can move ast_channel to a C file from channel.h and lop off the '__do_not_use_'. This patch sets up main/channel_interal_api.c to be the only file that actually accesses the ast_channel's fields directly. The intent would be for any API functions in channel.c to use the accessor functions. No more monkeying around with channel internals. We should use our own APIs. The interesting changes in this patch are the addition of channel_internal_api.c, the moving of the AST_DATA stuff from channel.c to channel_internal_api.c (note: the AST_DATA stuff will have to be reworked to use accessor functions when ast_channel is really opaque), and some re-working of the way channel iterators/callbacks are handled so as to avoid creating fake ast_channels on the stack to pass in matching data by directly accessing fields (since "name" is a stringfield and the fake channel doesn't init the stringfields, you can't use the ast_channel_name_set() function). I went with ast_channel_name(chan) for a getter, and ast_channel_name_set(chan, name) for a setter. The majority of the grunt-work for this change was done by writing a semantic patch using Coccinelle ( http://coccinelle.lip6.fr/ ). Review: https://reviewboard.asterisk.org/r/1655/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350223 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	c04f4d72fd	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) ........ Merged revisions 350023 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 350024 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350025 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	8e94432d9a	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) ........ Merged revisions 347369 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 347383 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@347395 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	dc05ce5e4f	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) ........ Merged revisions 344439 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 344440 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@344441 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	c1647ab33a	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) ........ Merged revisions 344102 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 344103 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@344104 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kevin P. Fleming	784bbf70d7	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. ........ Merged revisions 342990 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 342991 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@342992 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	796ed62f47	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 ........ Merged revisions 340470 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 340471 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@340472 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	e218748ac1	Merged revisions 337120 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r337120 \| mjordan \| 2011-09-20 17:49:36 -0500 (Tue, 20 Sep 2011) \| 28 lines 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/trunk@337124 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Olle Johansson	73424f128e	Merged revisions 336042 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r336042 \| oej \| 2011-09-15 14:46:38 +0200 (Tor, 15 Sep 2011) \| 12 lines 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/trunk@336043 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	39fe851e79	Merged revisions 331644 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r331644 \| jrose \| 2011-08-12 11:18:57 -0500 (Fri, 12 Aug 2011) \| 9 lines 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/trunk@331717 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jason Parker	1a8069abe2	Merged revisions 331579 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r331579 \| qwell \| 2011-08-11 16:54:54 -0500 (Thu, 11 Aug 2011) \| 13 lines 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/trunk@331580 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	4ea4b7e1ab	Merged revisions 328771 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.10 ................ r328771 \| kmoore \| 2011-07-19 10:46:54 -0500 (Tue, 19 Jul 2011) \| 18 lines 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/trunk@328772 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Leif Madsen	a525edea59	Merged revisions 328247 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.10 ................ r328247 \| lmadsen \| 2011-07-14 16:25:31 -0400 (Thu, 14 Jul 2011) \| 14 lines 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/trunk@328259 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Leif Madsen	a5770c43f0	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	15 years ago
Richard Mudgett	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	15 years ago
Richard Mudgett	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	15 years ago
Russell Bryant	6df3b851e3	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	15 years ago
Russell Bryant	d05e5281da	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	15 years ago
Richard Mudgett	a45d2f29c6	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	15 years ago
Sean Bright	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	15 years ago
Paul Belanger	7c3d14957b	Formatting change, remove red blobs git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@316054 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Olle Johansson	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	15 years ago
Jonathan Rose	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	15 years ago
Brett Bryant	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	15 years ago
David Vossel	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	15 years ago
Jeff Peeler	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	15 years ago
Paul Belanger	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
David Vossel	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
Sean Bright	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
Sean Bright	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
Sean Bright	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
Sean Bright	07b49f3adf	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
Sean Bright	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
Russell Bryant	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
Jason Parker	74e0a87776	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
Russell Bryant	2b056c97cd	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
Tilghman Lesher	72dc402f1f	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
Tilghman Lesher	758a671219	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
Andrew Parisio	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
Brett Bryant	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
Brett Bryant	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
Jean Galarneau	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
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	15 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	15 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	15 years ago

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474 Commits (9ed6de9fd254d4b51c493a8841fa1368ea4c0949)