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2010-05-04Add new possible value to autopause option to allow members to be autopaused ↵Mark Michelson
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
2010-05-04Merged revisions 260923 via svnmerge from Jeff Peeler
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
2010-05-03Add new admin features to meetme: Roll call, eject all, mute all, record ↵Jeff Peeler
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
2010-04-30Fix logic reversal error when queue callers join the queue.Mark Michelson
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
2010-04-28Merged revisions 259664 via svnmerge from Jeff Peeler
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
2010-04-22Pass interactive = 0 and fix a compile error.Eliel C. Sardanons
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@258595 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-04-22Asterisk data retrieval API.Eliel C. Sardanons
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
2010-04-21Merged revisions 258432 via svnmerge from Jeff Peeler
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
2010-04-21Added MixMonitorMute manager commandJulian Lyndon-Smith
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
2010-04-20Merged revisions 258029 via svnmerge from Jeff Peeler
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
2010-04-19Bad merge fixTilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@257883 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-04-16Merged revisions 257686 via svnmerge from Dwayne M. Hubbard
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
2010-04-12Remove silly debug message that is not useful.Leif Madsen
(issue #17159) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256860 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-04-09Merge Call completion support into trunk.Mark Michelson
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
2010-04-03Consolidate ast_channel.cid.cid_rdnis into ast_channel.redirecting.from.number.Richard Mudgett
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
2010-04-03Using the Dial application f option when the call is forwarded will likely ↵Richard Mudgett
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: https://origsvn.digium.com/svn/asterisk/trunk@256103 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-04-02Export MEETMEBOOKID and fix pin-less conferences with realtime conferencesRussell Bryant
(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: https://origsvn.digium.com/svn/asterisk/trunk@256019 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-04-02Allow symbol export filtering to work properly on platforms that have symbol ↵Kevin P. Fleming
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: https://origsvn.digium.com/svn/asterisk/trunk@255906 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-03-31Recorded merge of revisions 255591 via svnmerge from Tilghman Lesher
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: https://origsvn.digium.com/svn/asterisk/trunk@255592 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-03-31Add documentation clarifying when 't' and 'T' can be used.Leif Madsen
(closes issue #17021) Reported by: kovzol Tested by: lmadsen, kovzol, davidw, ebroad git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@255504 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-03-29This patch adds custom device state handling for ConfBridge conferences,Jared Smith
matching the devstate handling of the MeetMe conferences. Review: https://reviewboard.asterisk.org/r/572/ Closes issue #16972 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@255281 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-03-27We need to inclde sys/wait.h on OpenBSD to get WEXITSTATUS.Sean Bright
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@255158 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-03-24Allow configuration of minsecs and nextaftercmd per mailbox.Jeff Peeler
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: https://origsvn.digium.com/svn/asterisk/trunk@254321 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-03-23Remove unused structure member in app_queue.Sean Bright
(closes issue #15494) Reported by: makoto git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@254045 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-03-20Include sys/wait.h on FreeBSD to get the WEXITSTATUS() macro.Russell Bryant
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@253539 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-03-20Resolve compiler warnings on FreeBSD.Russell Bryant
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@253538 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-03-18Change usage of pipe to comma in UserEvent docs.Leif Madsen
Change the example usage of pipe as a separator to comma in the UserEvent documentation. (closes issue #16961) Reported by: jlpedrosa git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@253345 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-03-16Mask out previous arguments on each nested invocation of Gosub.Tilghman Lesher
(closes issue #16758) Reported by: wdoekes Patches: 20100316__issue16758.diff.txt uploaded by tilghman (license 14) Review: https://reviewboard.asterisk.org/r/561/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@252976 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-03-15Resolve a crash in SLATrunk when the specified trunk doesn't exist.Sean Bright
Reported by philipp64 in #asterisk-dev. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@252623 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-03-12Don't override a user option with the global option.Tilghman Lesher
(closes issue #16849) Reported by: ip-rob Patches: 20100311__issue16849.diff.txt uploaded by tilghman (license 14) Tested by: ip-rob git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@251989 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-03-11Because ExecIf needs to reprocess arguments, it's best if we don't remove ↵Tilghman Lesher
quotes during parsing. (closes issue #16905) Reported by: ip-rob Patches: 20100303__issue16905.diff.txt uploaded by tilghman (license 14) Tested by: ip-rob git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@251884 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-03-11If the argument to the system application is quoted, ensure we remove the ↵Tilghman Lesher
quotes before trying to execute. (closes issue #16842) Reported by: ip-rob Patches: 20100310__issue16842.diff.txt uploaded by tilghman (license 14) Tested by: ip-rob git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@251877 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-03-11Add supporting code for app-directory pause option.Alec L Davis
Since 1.6.1 CLI help reports that option p(n) 'initial pause' is available. Supporting code was never implemented. (closes issue #16751) Reported by: alecdavis Patches: directory_pause.trunk.diff.txt uploaded by alecdavis (license 585) Tested by: alecdavis Review: https://reviewboard.asterisk.org/r/481/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@251779 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-03-10Be less ambiguous in Record() app docs.Leif Madsen
For some reason the documentation for the 'k' application in trunk and 1.6.2 is different than 1.6.0 and 1.6.1, so I'm setting them all to match. The wording in 1.6.2 and trunk was ambiguous, so you could interpret the wording the mean that recording would continue upon hangup indefinitely, or you could interpret it to mean that the recorded data would not be discarded upon hangup. This change makes it clear we mean the latter, and not the former. Came from a discussion in #asterisk on IRC. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@251680 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-03-05Fix app_followme playing wrong sound files.Jeff Peeler
Fixes regression introduced in 140167 that uses the wrong variable names. (closes issue #16930) Reported by: ianc Patches: fix_reload_followme.diff uploaded by ianc (license 998) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@250979 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-03-05Missing quote in ODBC query.Tilghman Lesher
(closes issue #16953) Reported by: elguero Patches: app_voicemail-odbc-syntax-fix.diff uploaded by elguero (license 37) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@250913 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-03-03Removed cdrflags from ast_channel structure.Richard Mudgett
Only chan_dahdi set a value in cdrflags. Everyone else just copied it around the system. Noone cared about any value it may have contained. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@250565 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-03-03Updated CHANGES file to mention res_fax and res_fax_spandsp.Matthew Nicholson
Also fixed MODULEINFO depends and conflicts for app_fax, res_fax, and res_fax_spandsp. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@250302 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-03-02adds 'p' option to PickupChanDavid Vossel
The 'p' option allows the PickupChan app to pickup a ringing phone by looking for the first match to a partial channel name rather than requiring a full match. (closes issue #16613) Reported by: syspert Patches: pickipbycallid.patch uploaded by syspert (license 938) pickupbycallerid_v2.patch uploaded by dvossel (license 671) Tested by: dvossel, syspert git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@250141 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-03-02Fix literal values wrapped in documentation.Leif Madsen
(closes issue #16145) Reported by: tilghman git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@249950 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-03-02revert ability to exit echo appAlec L Davis
caused a regression, as only supported VOICE, not VIDEO etc. (issue #16880) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@249947 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-03-02Fix several XML documentation validate errors.Leif Madsen
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@249892 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-03-02fix build by checking result of symlink in test_voicemail_vmsaynameJeff Peeler
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@249891 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-03-02Add new application VMSayName for use with voicemail.Jeff Peeler
VMSayName that will play the recorded name of the voicemail user if it exists, otherwise will play the mailbox number. A unit test has been written to verify correct functionality called test_voicemail_vmsayname. (closes issue #14973) Reported by: ghjm Review: https://reviewboard.asterisk.org/r/530/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@249889 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-03-02fixes ability to exit echo appAlec L Davis
when called from a ISDN channel, null frames prevent '#' exit. Now only echo back VOICE and DTMF frames (issue #16880) Reported by: alecdavis Patches: echo_exit.diff.txt uploaded by alecdavis (license 585) Tested by: alecdavis git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@249801 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-03-01Merged revisions 249671 via svnmerge from Sean Bright
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r249671 | seanbright | 2010-03-01 14:35:01 -0500 (Mon, 01 Mar 2010) | 11 lines Fix crash in app_voicemail related to message counting. We were passing a 'struct inprocess **' and treating it like a 'struct inprocess *' causing a segfault. (closes issue #16921) Reported by: whardier Patches: 20100301_issue16921.patch uploaded by seanbright (license 71) Tested by: whardier ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@249672 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-03-01Constify a bit of app_voicemail, to make ODBC and IMAP compile once again.Tilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@249623 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-02-28Fix unit test that Alec Davis broke.Tilghman Lesher
(closes issue #16927) Reported by: alecdavis git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@249491 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-02-28make unit test check for NULL folder, which then defaults to INBOXAlec L Davis
previous test, gave false level of assurance that code was healthy. (issue #16927) Reported by: alecdavis Patches: based on app_voicemail_test.diff.txt uploaded by alecdavis (license 585) Tested by: alecdavis git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@249449 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-02-28Properly document voicemail API documents. Also fix a crash reported via ↵Tilghman Lesher
the -dev list. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@249405 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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