3766 Commits (73c617efd3732a2ee18a0df3ae4efd80a13be4a0)

Author	SHA1	Message	Date
Tilghman Lesher	45a4bf35c2	The switch fallthrough could create some errorneous situations, so best to force directly to the default case. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@273714 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tzafrir Cohen	c613897d1c	Fix various typos reported by Lintian ... (Also fix the typos in the comments) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@273641 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	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
Jeff Peeler	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
Matthew Nicholson	cb22af3ec5	Merged revisions 272367 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 This version of the patch only adds AgentComplete for attended transfers. It was already present for blind transfers. ........ r272367 | mnicholson | 2010-06-23 17:33:51 -0500 (Wed, 23 Jun 2010) | 8 lines Send AgentComplete manager events in the event of blind and attended transfers. (closes issue #16819) Reported by: elbriga Patches: app_queue.diff uploaded by elbriga (license 482) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272368 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Paul Belanger	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	16 years ago
Paul Belanger	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	16 years ago
Terry Wilson	2bcef29e11	Don't start the sla thread unless we realy need it ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272146 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Terry Wilson	7938510af9	Make sure reload updates SLA config ... Even if there are no stations or trunks defined, we need to start the sla thread to make sure we get the reload event. Also, when doing a reload we need to remove the existing trunks and stations or they end up hanging around. (closes issue #16818) Reported by: mbonin Patches: sla_reload.patch uploaded by twilson (license 396) Tested by: twilson git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272109 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	63fd368411	Add new application for declining counting words in multiple languages. ... (closes issue #16869) Reported by: chappell Patches: app_say_counted-20100317.c uploaded by chappell (license 8) Tested by: chappell git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271520 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Paul Belanger	531290385c	option w[(secs)] incorrectly capitalized in xmldoc ... (closes issue #17516) Reported by: karlfife git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271089 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	f3a9392542	Don't pass null to manager_event() ... (closes issue #17087) Reported by: bklang Patches: app-fax-null-sprintf1.diff uploaded by mnicholson (license 96) Tested by: bklang git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@269083 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Leif Madsen	c672763af8	Fix some doxygen warnings. ... (closes issue #17336) Reported by: snuffy Patches: doxygen-fixes1.diff uploaded by snuffy (license 35) Tested by: russell git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268969 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	afd4454c44	Generic Advice of Charge. ... Asterisk Generic AOC Representation - Generic AOC encode/decode routines. (Generic AOC must be encoded to be passed on the wire in the AST_CONTROL_AOC frame) - AST_CONTROL_AOC frame type to represent generic encoded AOC data - Manager events for AOC-S, AOC-D, and AOC-E messages Asterisk App Support - app_dial AOC-S pass-through support on call setup - app_queue AOC-S pass-through support on call setup AOC Unit Tests - AOC Unit Tests for encode/decode routines - AOC Unit Test for manager event representation. SIP AOC Support - Pass-through of generic AOC-D and AOC-E messages to snom phones via the snom AOC specification. - Creation of chan_sip page3 flags for the addition of the new 'snom_aoc_enabled' sip.conf option. IAX AOC Support - Natively supports AOC pass-through through the use of the new AST_CONTROL_AOC frame type DAHDI AOC Support - ETSI PRI full AOC Pass-through support - 'aoc_enable' chan_dahdi.conf option for independently enabling pass-through of AOC-S, AOC-D, AOC-E. - 'aoce_delayhangup' option for retrieving AOC-E on disconnect. - DAHDI A() dial string option for requesting AOC services. example usage: ;requests AOC-S, AOC-D, and AOC-E on call setup exten=>1111,1,Dial(DAHDI/g1/1112/A(s,d,e)) Review: https://reviewboard.asterisk.org/r/552/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267096 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	266db9fa8c	Silence a compiler warning. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267093 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	b0357dcc3e	Support setting locale per-mailbox (changes date/time languages for email, pager messages). ... (closes issue #14333) Reported by: klaus3000 Patches: 20090515__issue14333.diff.txt uploaded by tilghman (license 14) app_voicemail.c-svn-trunk-rev211675-patch.txt uploaded by klaus3000 (license 65) Tested by: klaus3000 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@266828 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Terry Wilson	ffbb85bb4d	Set app and appdata fields when a Dial is redirected ... (closes issue #17204) Reported by: one47 Tested by: twilson, one47 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@266786 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	70a1bf3142	Remove redundant ast_conntected_line_free call. ... This wouldn't cause any problems, but it's certainly not needed either. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@266098 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	9ed82007f1	Merged revisions 265610 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r265610 | mnicholson | 2010-05-25 11:48:19 -0500 (Tue, 25 May 2010) | 8 lines Don't mark the cdr records of unanswered queue calls with "NOANSWER". This restores the behavior prior to r258670. (closes issue #17334) Reported by: jvandal Patches: queue-cdr-fixes1.diff uploaded by mnicholson (license 96) Tested by: aragon, jvandal ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@265611 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	cba378d847	Allow SendDTMF to play digits to a specified channel. ... Patch supplied by reporter was modified to use autoservice and prevent a potential channel ref leak but is otherwise as the reporter uploaded it. (closes issue #17182) Reported by: rcasas Patches: app_senddtmf.c.patch_trunk uploaded by rcasas (license 641) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@265453 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	4e38beb960	Make app_rpt.c able to compile again. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@265367 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	1225ee831c	Merged revisions 265089 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r265089 | mmichelson | 2010-05-21 15:59:14 -0500 (Fri, 21 May 2010) | 8 lines Don't hang up on a queue caller if the file we attempt to play does not exist. This also fixes a documentation mistake in file.h that made my original attempt to correct this problem not work correctly. (closes issue #17061) Reported by: RoadKill ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@265090 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	a5bee137f9	Error message fix. ... (closes issue #17356) Reported by: kenner Patches: app_stack.c.diff uploaded by kenner (license 1040) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@264752 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	3d1f005fed	Dial and queue connected line update macro not always run when expected. ... The connected line update macro would not get run if the connected line number string was empty. The number could be empty if the connected line update did not update a number but the name. It should be run if there was an AST_CONTROL_CONNECTED_LINE frame received for pending dials and queues. Renamed and added some more comments for some confusing identifiers directly connected to the related code. Also fixed a memory leak in app_queue. Review: https://reviewboard.asterisk.org/r/669/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@264669 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	d38c6459f5	Merged revisions 264334 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r264334 | mnicholson | 2010-05-19 15:01:38 -0500 (Wed, 19 May 2010) | 5 lines Set quieted flag when receiving a dtmf tone during playback in speechbackground. (closes issue #16966) Reported by: asackheim ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@264335 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	94df424e1d	Merged revisions 263769 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r263769 | jpeeler | 2010-05-18 13:54:58 -0500 (Tue, 18 May 2010) | 10 lines Modify directory name reading to be interrupted with operator or pound escape. In the case of accidentally entering the wrong first three letters for the reading, users could be very frustrated if the name listing is very long. This allows interrupting the reading by pressing 0 or #. 0 will attempt to execute a configured operator (o) extension and # will exit and proceed in the dialplan. ABE-2200 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263807 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	fa8e44f232	With IMAP backend, messages in INBOX were counted twice for MWI. ... (closes issue #17135) Reported by: edhorton Patches: 20100513__issue17135.diff.txt uploaded by tilghman (license 14) 17135_2.diff uploaded by ebroad (license 878) Tested by: edhorton, ebroad git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263589 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	b5d5cc565f	Enhancements to connected line and redirecting work. ... From reviewboard: Digium has a commercial customer who has made extensive use of the connected party and redirecting information present in later versions of Asterisk Business Edition and which is to be in the upcoming 1.8 release. Through their use of the feature, new problems and solutions have come about. This patch adds several enhancements to maximize usage of the connected party and redirecting information functionality. First, Asterisk trunk already had connected line interception macros. These macros allow you to manipulate connected line information before it was sent out to its target. This patch adds the same feature except for redirecting information instead. Second, the ast_callerid and ast_party_id structures have been enhanced to provide a "tag." This tag can be set with func_callerid, func_connectedline, func_redirecting, and in the case of DAHDI, mISDN, and SIP channels, can be set in a configuration file. The idea behind the callerid tag is that it can be set to whatever value the administrator likes. Later, when running connected line and redirecting macros, the admin can read the tag off the appropriate structure to determine what action to take. You can think of this sort of like a channel variable, except that instead of having the variable associated with a channel, the variable is associated with a specific identity within Asterisk. Third, app_dial has two new options, s and u. The s option lets a dialplan writer force a specific caller ID tag to be placed on the outgoing channel. The u option allows the dialplan writer to force a specific calling presentation value on the outgoing channel. Fourth, there is a new control frame subclass called AST_CONTROL_READ_ACTION added. This was added to correct a very specific situation. In the case of SIP semi-attended (blond) transfers, the party being transferred would not have the opportunity to run a connected line interception macro to possibly alter the transfer target's connected line information. The issue here was that during a blond transfer, the SIP transfer code has no bridged channel on which to queue the connected line update. The way this was corrected was to add this new control frame subclass. Now, we queue an AST_CONTROL_READ_ACTION frame on the channel on which the connected line interception macro should be run. When ast_read is called to read the frame, ast_read responds by calling a callback function associated with the specific read action the control frame describes. In this case, the action taken is to run the connected line interception macro on the transferee's channel. Review: https://reviewboard.asterisk.org/r/652/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263541 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	a0b12a5666	Merged revisions 262662 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r262662 | dvossel | 2010-05-12 12:00:04 -0500 (Wed, 12 May 2010) | 11 lines fixes app_meetme dsp error We attempted to detect silence after translating a frame from signed linear. This caused a flooding of errors. To resolve this the code to detect silence was moved before the translation. (closes issue #17133) Reported by: jsdyer ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262744 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	1d7a548ae6	Ensure the arguments are initialized. Also miscellaneous CG cleanup. ... (closes issue #16576) Reported by: uxbod Patches: 20100505__issue16576.diff.txt uploaded by tilghman (license 14) Tested by: uxbod git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262656 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	c84e7f83c8	Merged revisions 262321 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r262321 | tilghman | 2010-05-11 12:22:07 -0500 (Tue, 11 May 2010) | 2 lines Fix issue #17302 a slightly different way (mad props to Qwell) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262330 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	62067caaab	fixes PickupChan application ... (closes issue #16863) Reported by: schern Patches: app_directed_pickup.c.patch uploaded by schern (license 995) for_trunk.diff uploaded by cjacobsen (license 1029) Tested by: Graber, cjacobsen, lathama, rickead2000, dvossel git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262240 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Alec L Davis	dd3343c33d	VoicemailMain and VMauthenticate, allow escape to the 'a' extension when a single '*' is entered ... Where a site uses VoicemailMain(mailbox) the users have to be at their own extension to clear their voicemail, they have no way of escaping VoicemailMain to allow entry of new boxnumber. This patch, allows a site to include to 'a' priority in the VoicemailMain context, to allow an escape. If the 'a' priority doesn't exist in the context that VoicemailMain was called from then it acts as the old behaviour. Reported by: alecdavis Tested by: alecdavis Patch vm_a_extension.diff2.txt uploaded by alecdavis (license 585) Review: https://reviewboard.asterisk.org/r/489/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262005 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	8312f25b13	Merged revisions 261735 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r261735 | jpeeler | 2010-05-06 15:10:59 -0500 (Thu, 06 May 2010) | 8 lines Only allow the operator key to be accepted after leaving a voicemail. Or rather disallow the operator key from being accepted when not offered, such as after finishing a recording from within the mailbox options menu. ABE-2121 SWP-1267 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@261736 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Paul Belanger	d7ff67179d	'queue reset stats' erroneously clears wrapuptime configuration. ... Resets each member's lastcall to 0 now. (closes issue #17262) Reported by: rain Patches: wrapuptime_reset_fix.diff uploaded by rain (license 327) Tested by: rain git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@261232 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	fc652b869a	Add new possible value to autopause option to allow members to be autopaused in all queues. ... See the CHANGES file and queues.conf.sample for more details. (closes issue #17008) Reported by: jlpedrosa Patches: queues.autopause_en_review.diff uploaded by jlpedrosa (license 1002) Review: https://reviewboard.asterisk.org/r/581/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@261051 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	9db934a869	Merged revisions 260923 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r260923 | jpeeler | 2010-05-04 13:46:46 -0500 (Tue, 04 May 2010) | 12 lines Voicemail transfer to operator should occur immediately, not after main menu. There were two scenarios in the advanced options that while using the operator=yes and review=yes options, the transfer occurred only after exiting the main menu (after sending a reply or leaving a message for an extension). Now after the audio is processed for the reply or message the transfer occurs immediately as expected. ABE-2107 ABE-2108 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@260924 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	8ddd92f823	Add new admin features to meetme: Roll call, eject all, mute all, record in-conf ... This patch adds the following in-conference admin DTMF features: *81 - Roll call (or simply user count if INTROUSER isn't enabled) *82 - Eject all non-admins *83 - Mute/unmute all non-admins *84 - Start recording the conference on the fly FWIW, this code uses newly recorded prompts. (closes issue #16379) Reported by: rfinnie Patches: meetme-enhancements-232771-v1.patch uploaded by rfinnie (license 940) modified slightly by me git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@260757 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	2dcb4df6d8	Fix logic reversal error when queue callers join the queue. ... When a specific position is specified for the queue, the idea was that the caller cannot be placed ahead of higher-priority callers. Unfortunately, the logic was reversed so that the caller could ONLY be placed ahead of higher priority callers. Discovered while writing a unit test. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@260344 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	dc9295da58	Merged revisions 259664 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r259664 | jpeeler | 2010-04-28 12:13:29 -0500 (Wed, 28 Apr 2010) | 4 lines Do not play goodbye prompt after timeout of message review. ABE-2124 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@259672 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Eliel C. Sardanons	78edf881d5	Pass interactive = 0 and fix a compile error. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@258595 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Eliel C. Sardanons	a753e8878b	Asterisk data retrieval API. ... This module implements an abstraction for retrieving and exporting asterisk data. Developed by: Brett Bryant <brettbryant@gmail.com> Eliel C. Sardanons (LU1ALY) <eliels@gmail.com> For the Google Summer of code 2009 Project. Documentation can be found in doxygen format and inside the header include/asterisk/data.h Review: https://reviewboard.asterisk.org/r/275/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@258517 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	e0e32a3bd8	Merged revisions 258432 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r258432 | jpeeler | 2010-04-21 16:45:36 -0500 (Wed, 21 Apr 2010) | 8 lines Fix looping forever when no input received in certain voicemail menu scenarios. Specifically, prompting for an extension (when leaving or forwarding a message) or when prompting for a digit (when saving a message or changing folders). ABE-2122 SWP-1268 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@258433 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Julian Lyndon-Smith	d85650e4aa	Added MixMonitorMute manager command ... Added a new manager command to mute/unmute MixMonitor audio on a channel. Added a new feature to audiohooks so that you can mute either read / write (or both) types of frames - this allows for MixMonitor to mute either side of the conversation without affecting the conversation itself. (closes issue #16740) Reported by: jmls Review: https://reviewboard.asterisk.org/r/487/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@258190 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	31338f9671	Merged revisions 258029 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r258029 | jpeeler | 2010-04-20 11:16:33 -0500 (Tue, 20 Apr 2010) | 11 lines Play correct prompt when voicemail store failure occurs after attempted forward. If a user's mailbox was full and a message was attempted to be forwarded to said box, warnings on the console would indicate failure. However, the played prompt was that of success (vm-msgsaved). Now storage failure is taken into account and the correct prompt (vm-mailboxfull) is played when appropriate. ABE-2123 SWP-1262 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@258065 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	990ccdd05f	Bad merge fix ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@257883 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Dwayne M. Hubbard	77868073a8	Merged revisions 257686 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r257686 | dhubbard | 2010-04-16 16:15:43 -0500 (Fri, 16 Apr 2010) | 21 lines Make the mixmonitor thread process audio frames faster Mantis issue 17078 reports MixMonitor recordings have shorter durations than the call duration. This was because the mixmonitor thread was not processing frames from the audiohook fast enough. The mixmonitor thread would slowly fall behind the most recent audio frame and when the channel hangs up, the mixmonitor thread would exit without processing the same number of frames as the channel; leaving the mixmonitor recording shorter than actual call duration. This revision fixes this issue by moving the ast_audiohook_trigger_wait() and the subsequent audiohook.status check into the block where the ast_audiohook_read_frame() function returns NULL. (closes issue #17078) Reported by: geoff2010 Patches: dw-M17078.patch uploaded by dhubbard (license 733) Tested by: dhubbard, geoff2010 Review: https://reviewboard.asterisk.org/r/611/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@257713 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Leif Madsen	875014bdd4	Remove silly debug message that is not useful. ... (issue #17159) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256860 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	e24661fd18	Merge Call completion support into trunk. ... From Reviewboard: CCSS stands for Call Completion Supplementary Services. An admittedly out-of-date overview of the architecture can be found in the file doc/CCSS_architecture.pdf in the CCSS branch. Off the top of my head, the big differences between what is implemented and what is in the document are as follows: 1. We did not end up modifying the Hangup application at all. 2. The document states that a single call completion monitor may be used across multiple calls to the same device. This proved to not be such a good idea when implementing protocol-specific monitors, and so we ended up using one monitor per-device per-call. 3. There are some configuration options which were conceived after the document was written. These are documented in the ccss.conf.sample that is on this review request. For some basic understanding of terminology used throughout this code, see the ccss.tex document that is on this review. This implements CCBS and CCNR in several flavors. First up is a "generic" implementation, which can work over any channel technology provided that the channel technology can accurately report device state. Call completion is requested using the dialplan application CallCompletionRequest and can be canceled using CallCompletionCancel. Device state subscriptions are used in order to monitor the state of called parties. Next, there is a SIP-specific implementation of call completion. This method uses the methods outlined in draft-ietf-bliss-call-completion-06 to implement call completion using SIP signaling. There are a few things to note here: * The agent/monitor terminology used throughout Asterisk sometimes is the reverse of what is defined in the referenced draft. * Implementation of the draft required support for SIP PUBLISH. I attempted to write this in a generic-enough fashion such that if someone were to want to write PUBLISH support for other event packages, such as dialog-state or presence, most of the effort would be in writing callbacks specific to the event package. * A subportion of supporting PUBLISH reception was that we had to implement a PIDF parser. The PIDF support added is a bit minimal. I first wrote a validation routine to ensure that the PIDF document is formatted properly. The rest of the PIDF reading is done in-line in the call-completion-specific PUBLISH-handling code. In other words, while there is PIDF support here, it is not in any state where it could easily be applied to other event packages as is. Finally, there are a variety of ISDN-related call completion protocols supported. These were written by Richard Mudgett, and as such I can't really say much about their implementation. There are notes in the CHANGES file that indicate the ISDN protocols over which call completion is supported. Review: https://reviewboard.asterisk.org/r/523 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256528 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	a5a0a5f867	Consolidate ast_channel.cid.cid_rdnis into ast_channel.redirecting.from.number. ... SWP-1229 ABE-2161 * Ensure chan_local.c:local_call() will not leak cid.cid_dnid when copying. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256104 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago