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2010-06-08Fix some doxygen warnings.lmadsen2-6/+19
(closes issue #17336) Reported by: snuffy Patches: doxygen-fixes1.diff uploaded by snuffy (license 35) Tested by: russell git-svn-id: http://svn.digium.com/svn/asterisk/trunk@268969 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-02Generic Advice of Charge.rmudgett2-0/+60
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: http://svn.digium.com/svn/asterisk/trunk@267096 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-02Silence a compiler warning.russell1-1/+3
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@267093 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-01Support setting locale per-mailbox (changes date/time languages for email, ↵tilghman1-7/+15
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: http://svn.digium.com/svn/asterisk/trunk@266828 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-01Set app and appdata fields when a Dial is redirectedtwilson1-0/+2
(closes issue #17204) Reported by: one47 Tested by: twilson, one47 git-svn-id: http://svn.digium.com/svn/asterisk/trunk@266786 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-26Remove redundant ast_conntected_line_free call.mmichelson1-1/+0
This wouldn't cause any problems, but it's certainly not needed either. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@266098 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-25Merged revisions 265610 via svnmerge from mnicholson1-7/+0
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: http://svn.digium.com/svn/asterisk/trunk@265611 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-24Allow SendDTMF to play digits to a specified channel.mmichelson1-2/+18
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: http://svn.digium.com/svn/asterisk/trunk@265453 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-24Make app_rpt.c able to compile again.rmudgett1-35/+35
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@265367 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-21Merged revisions 265089 via svnmerge from mmichelson1-0/+4
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: http://svn.digium.com/svn/asterisk/trunk@265090 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-20Error message fix.tilghman1-1/+1
(closes issue #17356) Reported by: kenner Patches: app_stack.c.diff uploaded by kenner (license 1040) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@264752 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-20Dial and queue connected line update macro not always run when expected.rmudgett2-43/+66
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: http://svn.digium.com/svn/asterisk/trunk@264669 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-19Merged revisions 264334 via svnmerge from mnicholson1-0/+1
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: http://svn.digium.com/svn/asterisk/trunk@264335 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-18Merged revisions 263769 via svnmerge from jpeeler1-14/+22
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: http://svn.digium.com/svn/asterisk/trunk@263807 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-17With IMAP backend, messages in INBOX were counted twice for MWI.tilghman1-20/+26
(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: http://svn.digium.com/svn/asterisk/trunk@263589 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-17Enhancements to connected line and redirecting work.mmichelson2-8/+80
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: http://svn.digium.com/svn/asterisk/trunk@263541 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-12Merged revisions 262662 via svnmerge from dvossel1-5/+4
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: http://svn.digium.com/svn/asterisk/trunk@262744 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-12Ensure the arguments are initialized. Also miscellaneous CG cleanup.tilghman1-30/+39
(closes issue #16576) Reported by: uxbod Patches: 20100505__issue16576.diff.txt uploaded by tilghman (license 14) Tested by: uxbod git-svn-id: http://svn.digium.com/svn/asterisk/trunk@262656 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-11Merged revisions 262321 via svnmerge from tilghman1-0/+1
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: http://svn.digium.com/svn/asterisk/trunk@262330 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-10fixes PickupChan applicationdvossel1-2/+2
(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: http://svn.digium.com/svn/asterisk/trunk@262240 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-07VoicemailMain and VMauthenticate, allow escape to the 'a' extension when a ↵alecdavis1-0/+44
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: http://svn.digium.com/svn/asterisk/trunk@262005 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-06Merged revisions 261735 via svnmerge from jpeeler1-1/+4
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: http://svn.digium.com/svn/asterisk/trunk@261736 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-05'queue reset stats' erroneously clears wrapuptime configuration.pabelanger1-1/+1
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: http://svn.digium.com/svn/asterisk/trunk@261232 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-04Add new possible value to autopause option to allow members to be autopaused ↵mmichelson1-5/+54
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: http://svn.digium.com/svn/asterisk/trunk@261051 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-04Merged revisions 260923 via svnmerge from jpeeler1-7/+8
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: http://svn.digium.com/svn/asterisk/trunk@260924 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-03Add new admin features to meetme: Roll call, eject all, mute all, record ↵jpeeler1-3/+158
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: http://svn.digium.com/svn/asterisk/trunk@260757 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-30Fix logic reversal error when queue callers join the queue.mmichelson1-1/+1
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: http://svn.digium.com/svn/asterisk/trunk@260344 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-28Merged revisions 259664 via svnmerge from jpeeler1-2/+2
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: http://svn.digium.com/svn/asterisk/trunk@259672 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-22Pass interactive = 0 and fix a compile error.eliel1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@258595 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-22Asterisk data retrieval API.eliel2-0/+482
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: http://svn.digium.com/svn/asterisk/trunk@258517 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-21Merged revisions 258432 via svnmerge from jpeeler1-2/+10
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: http://svn.digium.com/svn/asterisk/trunk@258433 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-21Added MixMonitorMute manager commandjmls1-1/+91
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: http://svn.digium.com/svn/asterisk/trunk@258190 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-20Merged revisions 258029 via svnmerge from jpeeler1-20/+39
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: http://svn.digium.com/svn/asterisk/trunk@258065 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-19Bad merge fixtilghman1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@257883 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-16Merged revisions 257686 via svnmerge from dhubbard1-5/+6
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: http://svn.digium.com/svn/asterisk/trunk@257713 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-12Remove silly debug message that is not useful.lmadsen1-1/+0
(issue #17159) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@256860 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-09Merge Call completion support into trunk.mmichelson1-9/+98
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: http://svn.digium.com/svn/asterisk/trunk@256528 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-03Consolidate ast_channel.cid.cid_rdnis into ast_channel.redirecting.from.number.rmudgett4-32/+31
SWP-1229 ABE-2161 * Ensure chan_local.c:local_call() will not leak cid.cid_dnid when copying. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@256104 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-03Using the Dial application f option when the call is forwarded will likely ↵rmudgett1-1/+1
crash. Fix app_dial.c:do_forward() OPT_FORCECLID setting cid.cid_num with a stack allocated string instead of a heap allocated string. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@256103 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-02Export MEETMEBOOKID and fix pin-less conferences with realtime conferencesrussell1-0/+13
(closes issue #16866) Reported by: DEA Patches: rt-meetme-options.txt uploaded by DEA (license 3) Tested by: DEA Review: https://reviewboard.asterisk.org/r/582/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@256019 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-02Allow symbol export filtering to work properly on platforms that have symbol ↵kpfleming2-20/+20
prefixes. Some platforms prefix externally-visible symbols in object files generated from C sources (most commonly, '_' is the prefix). On these platforms, the existing symbol export filtering process ends up suppressing all the symbols that are supposed to be left visible. This patch allows the prefix string to be supplied to the top-level Makefile in the LINKER_SYMBOL_PREFIX variable, and then generates the linker scripts as required to include the prefix supplied. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@255906 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-31Recorded merge of revisions 255591 via svnmerge from tilghman1-20/+142
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r255591 | tilghman | 2010-03-31 14:09:46 -0500 (Wed, 31 Mar 2010) | 15 lines Ensure line terminators in email are consistent. Fixes an issue with certain Mail Transport Agents, where attachments are not interpreted correctly. (closes issue #16557) Reported by: jcovert Patches: 20100308__issue16557__1.4.diff.txt uploaded by tilghman (license 14) 20100308__issue16557__1.6.0.diff.txt uploaded by tilghman (license 14) 20100308__issue16557__trunk.diff.txt uploaded by tilghman (license 14) Tested by: ebroad, zktech Reviewboard: https://reviewboard.asterisk.org/r/544/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@255592 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-31Add documentation clarifying when 't' and 'T' can be used.lmadsen1-2/+4
(closes issue #17021) Reported by: kovzol Tested by: lmadsen, kovzol, davidw, ebroad git-svn-id: http://svn.digium.com/svn/asterisk/trunk@255504 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-29This patch adds custom device state handling for ConfBridge conferences,jsmith1-0/+8
matching the devstate handling of the MeetMe conferences. Review: https://reviewboard.asterisk.org/r/572/ Closes issue #16972 git-svn-id: http://svn.digium.com/svn/asterisk/trunk@255281 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-27We need to inclde sys/wait.h on OpenBSD to get WEXITSTATUS.seanbright1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@255158 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-24Allow configuration of minsecs and nextaftercmd per mailbox.jpeeler1-9/+192
Previously only configurable globally. A unit test has also been written to provide protection against parse failures for supported mailbox options. (closes issue #16864) Reported by: kobaz Patches: voicemail2.patch uploaded by kobaz (license 834) Review: https://reviewboard.asterisk.org/r/555/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@254321 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-23Remove unused structure member in app_queue.seanbright1-2/+0
(closes issue #15494) Reported by: makoto git-svn-id: http://svn.digium.com/svn/asterisk/trunk@254045 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-20Include sys/wait.h on FreeBSD to get the WEXITSTATUS() macro.russell1-0/+3
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@253539 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-20Resolve compiler warnings on FreeBSD.russell2-4/+4
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@253538 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-18Change usage of pipe to comma in UserEvent docs.lmadsen1-1/+1
Change the example usage of pipe as a separator to comma in the UserEvent documentation. (closes issue #16961) Reported by: jlpedrosa git-svn-id: http://svn.digium.com/svn/asterisk/trunk@253345 f38db490-d61c-443f-a65b-d21fe96a405b
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