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2010-05-03Add new admin features to meetme: Roll call, eject all, mute all, record ↵jpeeler1-0/+2
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
2010-04-22Asterisk data retrieval API.eliel1-0/+4
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
2010-04-21IAXpeers output now matches SIPpeers format for manager (AMI).lmadsen1-0/+2
(closes issue #17100) Reported by: secesh Tested by: pabelanger Review: https://reviewboard.asterisk.org/r/594/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@258344 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-21Added CHANGES entry for new MixMonitorMute AMI command.jmls1-0/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@258227 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-09Merge Call completion support into trunk.mmichelson1-0/+7
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 An admittedly out-of-date overview of the architecture can be found in the file doc/CCSS_architecture.pdf in the CCSS branch.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
2010-04-09func_srv and explicit specification of a remote IP for SIP.mmichelson1-0/+7
From Review Board: There are two interrelated changes here. First, there is the introduction of func_srv. This adds two new read-only dialplan functions, SRVQUERY and SRVRESULT. They work very similarly to the ENUMQUERY and ENUMRESULT functions, except that this allows one to query SRV records instead. In order to facilitate this work, I added a couple of new API calls to srv.h. ast_srv_get_record_count tells the number of records returned by an SRV lookup. This number is calculated at the time of the SRV lookup. ast_srv_get_nth_record allows one to get a numbered SRV record. Second, there is the modification to chan_sip that allows one to specify a hostname or IP address (along with a port) to send an outgoing INVITE to when dialing a SIP peer. This goes hand-in-hand with func_srv. You can query SRV records and then use the host and port from the results to dial via a specific host instead of what is configured in sip.conf. Review: https://reviewboard.asterisk.org/r/608 SWP-1200 git-svn-id: http://svn.digium.com/svn/asterisk/trunk@256485 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-29This patch adds custom device state handling for ConfBridge conferences,jsmith1-0/+2
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
2010-03-24Allow configuration of minsecs and nextaftercmd per mailbox.jpeeler1-0/+1
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
2010-03-23Change per-file debug and verbose levels to be per-module, the waykpfleming1-0/+11
users expect them to work. 'core set debug' and 'core set verbose' can optionally change the level for a specific filename; however, this is actually for a specific source file name, not the module that source file is included in. With examples like chan_sip, chan_iax2, chan_misdn and others consisting of multiple source files, this will not lead to the behavior that users expect. If they want to set the debug level for chan_sip, they want it set for all of chan_sip, and not to have to also set it for reqresp_parser and other files that comprise the chan_sip module. This patch changes this functionality to be module-name based instead of file-name based. To make this work, some Makefile modifications were required to ensure that the AST_MODULE definition is present in each object file produced for each module as well. Review: https://reviewboard.asterisk.org/r/574/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@253917 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-05PITCH_SHIFT dialplan functiondvossel1-0/+2
The PITCH_SHIFT function can be used on a channel to independently modify the pitch of both rx and tx audio streams. Now you can improve your conference calls by assigning a random pitch effect to everyone entering a meetme room, or just make your day more interesting by making your co-workers sound funny. These are just some of the numerious practical uses for this function. Enjoy! https://reviewboard.asterisk.org/r/526/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@251038 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-03Add new config option to control AMI alarm event reporting in chan_dahdi.jpeeler1-0/+2
New config parameter "reportalarms" added in chan_dahdi.conf which supports the following possible values: "channels": report each channel alarms (current behavior, default for backward compatibility) "spans": report an "SpanAlarm" event when the span of any configured channel is alarmed "all": report channel and span alarms (aggregated behavior) "none": do not report any alarms (closes issue #16709) Reported by: nahuelgreco Patches: chan_dahdi.c.reportalarms.patch uploaded by nahuelgreco (license 162) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@250392 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-03Updated CHANGES file to mention res_fax and res_fax_spandsp.mnicholson1-0/+7
Also fixed MODULEINFO depends and conflicts for app_fax, res_fax, and res_fax_spandsp. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@250302 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-02adds 'p' option to PickupChandvossel1-0/+2
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: http://svn.digium.com/svn/asterisk/trunk@250141 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-02Add new application VMSayName for use with voicemail.jpeeler1-0/+2
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: http://svn.digium.com/svn/asterisk/trunk@249889 f38db490-d61c-443f-a65b-d21fe96a405b
2010-02-18Add a new manager event for our buddies status.phsultan1-0/+2
The new JabberStatus event gives a concise view of the status change to the AMI clients. Thanks fiddur! (closes issue #16760) Reported by: fiddur Patches: 244498.2.diff uploaded by fiddur (license 678) Tested by: fiddur, phsultan git-svn-id: http://svn.digium.com/svn/asterisk/trunk@247500 f38db490-d61c-443f-a65b-d21fe96a405b
2010-02-17Add support for GROUP_MATCH_COUNT regex matching on categoryjpeeler1-0/+1
Current support for regex matching was previously only available on the group. Also, error reporting for regex failures has been added. In addition to this feature enhancement a unit test has been written to check the regular expression logic to ensure the count operation is working as expected. (closes issue #16642) Reported by: kobaz Patches: groupmatch2.patch uploaded by kobaz (license 834) Review: https://reviewboard.asterisk.org/r/503/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@247295 f38db490-d61c-443f-a65b-d21fe96a405b
2010-02-17addition of dynamic parkinglots featuredvossel1-0/+11
This feature allows for parkinglots to be created dynamically within the dialplan. Thanks to all who were involved with getting this patch written and tested! (closes issue #15135) Reported by: IgorG Patches: features.dynamic_park.v3.diff uploaded by IgorG (license 20) 2009090400_dynamicpark.diff.txt uploaded by mvanbaak (license 7) dynamic_parkinglot.diff uploaded by dvossel (license 671) Tested by: eliel, IgorG, acunningham, mvanbaak, zktech Review: https://reviewboard.asterisk.org/r/352/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@247248 f38db490-d61c-443f-a65b-d21fe96a405b
2010-02-12Updated doc for OSP lookup application.transnexus1-0/+3
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@246382 f38db490-d61c-443f-a65b-d21fe96a405b
2010-02-03Add some additional option support for non-default parking lots.jpeeler1-0/+2
The options are: parkedcallparking, parkedcallhangup, parkedcallrecording, and parkedcalltransfers. Previously these options were only available for the default parking lot. (closes issue #16641) Reported by: bluecrow76 Patches: asterisk-1.6.2.1-features.c.diff uploaded by bluecrow76 (license 270) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@244598 f38db490-d61c-443f-a65b-d21fe96a405b
2010-02-02Properly respect GOSUB_RESULT as to what to do with the master channel.tilghman1-0/+2
Previously, we would parse GOSUB_RESULT, but not actually do anything with it. Also, allow GOSUB_RETVAL to be inherited back across a peer/master channel. (closes issue #16687) Reported by: bklang Patches: app_dial-preserve-gosub_retval.patch uploaded by bklang (license 919) (with modifications) (closes issue #16686) Reported by: bklang Patches: app_dial-respect-gosub_result.patch uploaded by bklang (license 919) (with modifications) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@244393 f38db490-d61c-443f-a65b-d21fe96a405b
2010-01-27expand code based appreviation of AST_CONFIG_DIR to configuration directoryjpeeler1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@243652 f38db490-d61c-443f-a65b-d21fe96a405b
2010-01-27Add new option to asterisk.conf (lockconfdir) to protect conf dir during reloadsjpeeler1-0/+2
(closes issue #16358) Reported by: raarts Patches: lockconfdir.diff uploaded by raarts (license 937) modified by me git-svn-id: http://svn.digium.com/svn/asterisk/trunk@243551 f38db490-d61c-443f-a65b-d21fe96a405b
2010-01-19Create iterative method for querying SRV results, and use that for finding ↵tilghman1-0/+5
AGI servers. (closes issue #14775) Reported by: _brent_ Patches: 20091215__issue14775.diff.txt uploaded by tilghman (license 14) hagi-5.patch uploaded by brent (license 388) Tested by: _brent_ Reviewboard: https://reviewboard.asterisk.org/r/378/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@241188 f38db490-d61c-443f-a65b-d21fe96a405b
2010-01-18Make HASHes inheritable across channel creation.tilghman1-0/+3
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@241012 f38db490-d61c-443f-a65b-d21fe96a405b
2010-01-18transmit_silence_during_record replaced by transmit_silencedvossel1-0/+4
In asterisk.conf, transmit_silence_during_record has been removed in favor of using only the transmit_silence option. The transmit_silence_during_record option remains a valid option in asterisk.conf, but has been removed from the sample config and noted in CHANGES. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@240971 f38db490-d61c-443f-a65b-d21fe96a405b
2010-01-05Add a missing part of the connected line work into trunk.mmichelson1-0/+4
Part of the work done for connected line was to add an optional argument to the 'f' option to allow for the connected party information of the outgoing channel to be set to the argument provided. This was overlooked during the merge of the work to trunk and is being added back now. The CHANGES file has also been updated to note this change. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@237803 f38db490-d61c-443f-a65b-d21fe96a405b
2010-01-05Spell "aficionado" like someone who isn't stupid.mmichelson1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@237802 f38db490-d61c-443f-a65b-d21fe96a405b
2009-12-23Update CHANGES to reflect new QUEUE_MEMBER option, "ready"dvossel1-0/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@236312 f38db490-d61c-443f-a65b-d21fe96a405b
2009-12-23update CHANGES to reflect new 'R' app_queue option plus a minor optimization ↵dvossel1-0/+4
to the feature patch (issue #16384) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@236306 f38db490-d61c-443f-a65b-d21fe96a405b
2009-12-22update CHANGES to reflect the addition of the test frameworkdvossel1-0/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@236028 f38db490-d61c-443f-a65b-d21fe96a405b
2009-12-19app_dial optional parameter to option 'r' to allow play indication from ↵alecdavis1-0/+2
indications.conf (closes issue #14504) Reported by: alecdavis Tested by: alecdavis,jsmith Patch app_dial.play_ring_indications.diff7.txt uploaded by alecdavis (license 585) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@235740 f38db490-d61c-443f-a65b-d21fe96a405b
2009-12-16Add auth_policy option to jabber.conf for auto user registration.jpeeler1-0/+1
The option is global and currently the acceptable values as noted in the sample config are accept or deny. (closes issue #15228) Reported by: lp0 git-svn-id: http://svn.digium.com/svn/asterisk/trunk@235342 f38db490-d61c-443f-a65b-d21fe96a405b
2009-12-16Enhance AMI redirect to allow channels to be redirected to different places.jpeeler1-0/+3
New parameters ExtraContext, ExtraExtension, and ExtraPriority have been added to redirect the second channel to a different location. Previously, it was only possible to redirect both channels to the same place. (closes issue #15853) Reported by: haakon Patches: trunk-manager.c.patch uploaded by haakon (license 880) Tested by: jpeeler git-svn-id: http://svn.digium.com/svn/asterisk/trunk@235265 f38db490-d61c-443f-a65b-d21fe96a405b
2009-12-14Allow greetings-only mailboxes for Voicemail.tilghman1-0/+2
(closes issue #15132) Reported by: floletarmo Patches: voicemail_changes.patch uploaded by floletarmo (license 784) (with some additional changes by me) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@234820 f38db490-d61c-443f-a65b-d21fe96a405b
2009-12-10Add audio announcement option to app_pagejpeeler1-0/+4
As described in the CHANGES file: * MeetMe has a new option 'G' to play an announcement before joining a conference. * Page has a new option 'A(x)' which will playback an announcement simultaneously to all paged phones (and optionally excluding the caller's one using the new option 'n') before the call is bridged. To add the new option to meetme, the conference flag options had to be extended to 64 bits. (closes issue #14365) Reported by: dferrer Patches: page_announce.patch uploaded by dferrer (license 525) modified by me Review: https://reviewboard.asterisk.org/r/188/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@234173 f38db490-d61c-443f-a65b-d21fe96a405b
2009-12-09Move an entry from CHANGES to UPGRADE.txt.russell1-3/+0
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@234055 f38db490-d61c-443f-a65b-d21fe96a405b
2009-12-09Move an entry from CHANGES that should be in UPGRADE.txt.russell1-2/+0
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@234053 f38db490-d61c-443f-a65b-d21fe96a405b
2009-12-09Provide a real description of LOCAL_PEEK().russell1-3/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@234051 f38db490-d61c-443f-a65b-d21fe96a405b
2009-12-09Remove a feature from CHANGES that was listed twice for 1.6.2.russell1-2/+0
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@234028 f38db490-d61c-443f-a65b-d21fe96a405b
2009-12-09Fix up the faxdetect entry in CHANGES.russell1-3/+4
This feature was listed as a 1.6.2 feature, even though it's in all 1.6.X versions. The description of the feature was also no longer accurate. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@234008 f38db490-d61c-443f-a65b-d21fe96a405b
2009-12-09Remove an entry from CHANGES that is already in UPGRADE.txt (where it should ↵russell1-3/+0
be). git-svn-id: http://svn.digium.com/svn/asterisk/trunk@233967 f38db490-d61c-443f-a65b-d21fe96a405b
2009-12-07Add applications JabberJoin, JabberLeave, JabberSendGroup for XMPP groupchatjpeeler1-0/+2
(closes issue #14352) Reported by: fiddur Patches: trunk-14352-2.diff uploaded by phsultan (license 73) Tested by: fiddur git-svn-id: http://svn.digium.com/svn/asterisk/trunk@233468 f38db490-d61c-443f-a65b-d21fe96a405b
2009-12-04update CHANGES file for .m3u support in Mp3Player applicationdvossel1-0/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@233235 f38db490-d61c-443f-a65b-d21fe96a405b
2009-12-04update CHANGES for new queue option, penaltymemberslimit.dvossel1-0/+3
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@233198 f38db490-d61c-443f-a65b-d21fe96a405b
2009-12-03Add pagerdateformat, to allow shorter dates for SMS messages.tilghman1-0/+2
(closes issue #16263) Reported by: andrew Patches: pagerdate.patch uploaded by andrew (license 240) (with a slight modification by me) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@232916 f38db490-d61c-443f-a65b-d21fe96a405b
2009-12-03Extend voicemail to allow IMAP folders to be specified per mailbox.jpeeler1-0/+3
Previously only possible per context, new option called imapfolder. (closes issue #14298) Reported by: jablko Patches: patch-200906202 uploaded by jablko (license 675) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@232700 f38db490-d61c-443f-a65b-d21fe96a405b
2009-12-02update CHANGES and UPGRADE.txt for early media behavior change between 1.6.1 ↵dvossel1-0/+3
and 1.6.2 (closes issue #16212) Reported by: miki git-svn-id: http://svn.digium.com/svn/asterisk/trunk@232657 f38db490-d61c-443f-a65b-d21fe96a405b
2009-12-02Add an 'X' option to the asterisk application which enables #exec for ↵file1-1/+3
configuration files. This option can be used to enable #exec support in the asterisk.conf configuration file. (closes issue #16260) Reported by: atis Patches: exec_includes.patch uploaded by atis (license 242) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@232510 f38db490-d61c-443f-a65b-d21fe96a405b
2009-12-02Add an option to Record which enables a mode where any DTMF digit will ↵file1-0/+2
terminate recording. (closes issue #15436) Reported by: Vince Patches: app_record.diff uploaded by Vince (license 823) Tested by: dbrooks git-svn-id: http://svn.digium.com/svn/asterisk/trunk@232442 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-24Updated CHANGES file to describe the new 'd' option to app_followme added in ↵mnicholson1-0/+2
r230964 (related to issue #14155) Reported by: junky git-svn-id: http://svn.digium.com/svn/asterisk/trunk@231025 f38db490-d61c-443f-a65b-d21fe96a405b
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