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2011-03-04Add setvar option to calendaringTerry Wilson
Adding the setvar option with variable substitution on the value allows things like setting the outbound caller id name to the summary of a calendar event, etc. Values could be chained together as they are appended in order to do some scripting if necessary. Review: https://reviewboard.asterisk.org/r/1134/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@309640 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-22Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd ↵David Vossel
audio ConfBridge, and other stuff -Functional changes 1. Dynamic global format list build by codecs defined in codecs.conf 2. SILK 8khz, 12khz, 16khz, and 24khz with custom attributes defined in codecs.conf 3. Negotiation of SILK attributes in chan_sip. 4. SPEEX 32khz with translation 5. SLINEAR 8khz, 12khz, 24khz, 32khz, 44.1khz, 48khz, 96khz, 192khz with translation using codec_resample.c 6. Various changes to RTP code required to properly handle the dynamic format list and formats with attributes. 7. ConfBridge now dynamically jumps to the best possible sample rate. This allows for conferences to take advantage of HD audio (Which sounds awesome) 8. Audiohooks are no longer limited to 8khz audio, and most effects have been updated to take advantage of this such as Volume, DENOISE, PITCH_SHIFT. 9. codec_resample now uses its own code rather than depending on libresample. -Organizational changes Global format list is moved from frame.c to format.c Various format specific functions moved from frame.c to format.c Review: https://reviewboard.asterisk.org/r/1104/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308582 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-09Add new manager action MeetmeListRooms.Jeff Peeler
From the submitter: I've added a new manager action to list only the active conferences on an Asterisk system. It shows the same data displayed when you run a 'meetme list' on the Asterisk CLI. (closes issue #17905) Reported by: rcasas Patches: app_meetme.c.patch uploaded by rcasas (license 641) Review: https://reviewboard.asterisk.org/r/874/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307359 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-09Allow parkedmusicclass to be settable for non-default parking lots.Jeff Peeler
(closes issue #17946) Reported by: bluecrow76 Patches: asterisk-1.8.0-beta4-multipark-fixes-2010SEP02.diff git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307231 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-07Pass a MCID request to the bridged channel.Richard Mudgett
Pass a MCID request to the bridged channel so the bridged channel can send it to the network. The ability to send the MCID request on an ISDN span is enabled with the new chan_dahdi.conf mcid_send option. JIRA SWP-2845 JIRA ABE-2736 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306755 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-04Add ISDN display ie text handling options to chan_dahdi.conf.Richard Mudgett
The display ie handling can be controlled independently in the send and receive directions with the following options: * Block display text data. * Use display text in SETUP/CONNECT messages for name. * Use display text for COLP name updates (FACILITY/NOTIFY as appropriate). * Pass arbitrary display text during a call. Sent in INFORMATION messages. Received from any message that the display text was not used as a name. If the display options are not set then the options default to legacy behavior. The arbitrary display text is exchanged between bridged channels using the AST_FRAME_TEXT frame type. To send display text from the dialplan use the SendText() application when the arbitrary display text option is enabled. JIRA SWP-2688 JIRA ABE-2693 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306396 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-21Add DB_KEYS.Tilghman Lesher
Discussion on #asterisk on 2011-01-19: (02:07:03 PM) boch: i wonder how to cycle all entries in a tree (02:07:11 PM) leifmadsen: use While() (02:07:17 PM) leifmadsen: you need to know the tree structure already though (02:07:36 PM) boch: what you mean? (02:09:02 PM) leifmadsen: you need to know the structure prior to looping, because you can't just return the structure from the dialplan (02:09:43 PM) leifmadsen: the only way I can think of doing that is via something like writing the output of: asterisk -rx "database show" to a file, then looping through that to know the structure of the database and check everything (02:09:59 PM) leifmadsen: but at that point you're better off just using either a relational database or an external script (02:10:13 PM) boch: for example i need to know all entries in the tree (02:10:15 PM) boch: got it (02:10:20 PM) leifmadsen: exactly (02:10:22 PM) leifmadsen: that's the problem (02:10:22 PM) boch: thank you (02:13:09 PM) mateu: yeah, i'm surprised there isn't something from the dialplan like 'database show family' so one can get all keys in a family to loop over. (02:15:35 PM) leifmadsen: database shows everything (02:16:22 PM) mateu: i mean something from the dial plan that mimics 'database show <family>' (02:16:41 PM) leifmadsen: guess no one has found that important enough to program :) (02:16:52 PM) leifmadsen: at that point you should probably just use a relational database... (02:17:10 PM) mateu: i dunno (02:17:16 PM) mateu: seems pretty basic to me. (02:17:16 PM) leifmadsen: me either (02:17:19 PM) leifmadsen: sure does (02:17:24 PM) leifmadsen: no one has programmed it though (02:17:28 PM) ***leifmadsen shrugs (02:17:43 PM) mateu: ok, well at least we know how it currently stands. thanks leifmadsen (02:28:52 PM) Corydon76-home: leifmadsen: something like HASHKEYS() ? (02:30:11 PM) leifmadsen: Corydon76-home: ummm, I was thinking more like DUNDI_QUERY() and DUNDI_RESULT() (02:30:31 PM) leifmadsen: although HASHKEYS() might work (02:30:58 PM) leifmadsen: actually ya, looking at it, similar to HASHKEYS() (02:31:01 PM) leifmadsen: DBKEYS() I guess? (02:31:45 PM) Corydon76-home: So with no argument, retrieves families, with an argument, retrieves keys of that family? (02:34:02 PM) leifmadsen: ya (02:34:16 PM) leifmadsen: how would you iterate through layers of them? (02:34:30 PM) leifmadsen: i.e. family/key/key/key ? (02:34:43 PM) Corydon76-home: Essentially, yes git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@303198 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-13Add dialplan variables for asterisk.conf directoriesPaul Belanger
Review: https://reviewboard.asterisk.org/r/1075/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@301729 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-04Optional HOLD/RETRIEVE signaling for PTMP TE when the bridge goes on and off ↵Richard Mudgett
hold. Added the moh_signaling option to specify what to do when the channel's bridged peer puts the ISDN channel on and off of hold. Implemented as a FSM to control libpri ISDN signaling when the bridged peer places the channel on and off of hold with the AST_CONTROL_HOLD and AST_CONTROL_UNHOLD control frames. JIRA SWP-2687 JIRA ABE-2691 Review: https://reviewboard.asterisk.org/r/1063/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@300212 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-12-31Support negative filters.Tilghman Lesher
(closes issue #17979) Reported by: tilghman Patches: 20100911__for_blitzrage.diff.txt uploaded by tilghman (license 14) Tested by: lmadsen git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@300045 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-12-31Support an alternate configuration file for the 'logger reload' command.Tilghman Lesher
(closes issue #17668) Reported by: tilghman Patches: 20100718__logger_reload_altconf__2.diff.txt uploaded by tilghman (license 14) Review: (by lmadsen, russell within comments on issue tracker) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@300044 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-24Meetme use voicemail greet for join/leave announceAndrew Parisio
Added option v(mailbox@[context]) which tells MeetMe where to look for a users greet file. If one does not exist it clears the v option and defers to the functionality of i/I as/if set by the MeetMe() command. Review: https://reviewboard.asterisk.org/r/1009/ (closes issue #18297) Reported by: parisioa Patches: meetme_final_patch_v.diff uploaded by parisioa (license 1153) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@296249 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-02New CLI command 'gtalk show settings'.Paul Belanger
Review: https://reviewboard.asterisk.org/r/984/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@293578 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-02Add to the CHANGES file that the HTTP server supports IPv6 addressing.Mark Michelson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@293577 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-10-11Merged revisions 291194 via svnmerge from David Vossel
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r291194 | dvossel | 2010-10-11 16:44:04 -0500 (Mon, 11 Oct 2010) | 2 lines Update CHANGES to reflect new gtalk.conf options. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@291195 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-09-23Merged revisions 288606 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r288606 | tilghman | 2010-09-23 13:44:44 -0500 (Thu, 23 Sep 2010) | 2 lines Add note about the checkhangup option of ${CHANNEL()} ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@288607 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-09-20Merged revisions 287647 via svnmerge from David Vossel
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r287647 | dvossel | 2010-09-20 17:09:16 -0500 (Mon, 20 Sep 2010) | 21 lines Addition of the FrameHook API (AKA AwesomeHooks) So far all our tools for viewing and manipulating media streams within Asterisk have been entirely focused on audio. That made sense then, but is not scalable now. The FrameHook API lets us tap into and manipulate _ANY_ type of media or signaling passed on a channel present today or in the future. This tool is a step in the direction of expanding Asterisk's boundaries and will help generate some rather interesting applications in the future. In addition to the FrameHook API, a simple dialplan function exercising the api has been included as well. This function is called FRAME_TRACE(). FRAME_TRACE() allows for the internal ast_frames read and written to a channel to be output. Filters can be placed on this function to debug only certain types of frames. This function could be thought of as an internal way of doing ast_frame packet captures. Review: https://reviewboard.asterisk.org/r/925/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@287648 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-09-15Merged revisions 286931 via svnmerge from Jeff Peeler
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r286931 | jpeeler | 2010-09-15 14:22:15 -0500 (Wed, 15 Sep 2010) | 16 lines Add parking extension for non-default parking lots. This is a new feature that allows for parking to custom parking lots to be accessed directly, rather than with channel variables or by changing the default parking lot. The extension is set with the parkext option just as the default parking lot is done. Also, the manager action has been updated to optionally allow a specified parking lot. (closes issue #14882) Reported by: vmikhnevych Patches: patch_14882.txt uploaded by mnick (license 874) modified by me Review: https://reviewboard.asterisk.org/r/884/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@286939 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-09-10Merged revisions 285992 via svnmerge from David Ruggles
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r285992 | diruggles | 2010-09-10 09:13:16 -0400 (Fri, 10 Sep 2010) | 1 line Added missing documentation for ExternalIVR feature added in January 2010 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@285993 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-09-03Merged revisions 284950 via svnmerge from David Vossel
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r284950 | dvossel | 2010-09-03 12:29:02 -0500 (Fri, 03 Sep 2010) | 14 lines authenticate OPTIONS requests just like we would an INVITE OPTIONS requests should be treated the same as an INVITE This includes authentication. This patch adds the ability for incoming out of dialog OPTION requests to be authenticated before providing a response indicating whether an extension is available or not. The authentication routine works the exact same way as it does for incoming INVITEs. This means that if a peer has 'insecure=invite' in their peer definition, the same will be true for the processing of the OPTIONS request. Review: https://reviewboard.asterisk.org/r/881/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@284951 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-08-13Merged revisions 282302 via svnmerge from David Vossel
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r282302 | dvossel | 2010-08-13 17:23:38 -0500 (Fri, 13 Aug 2010) | 10 lines remove current STUN support from chan_sip.c This patch removes the current broken/useless stun support from chan_sip. (closes issue #17622) Reported by: philipp2 Review: https://reviewboard.asterisk.org/r/855/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@282304 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-08-13Merged revisions 282271 via svnmerge from David Vossel
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r282271 | dvossel | 2010-08-13 15:11:58 -0500 (Fri, 13 Aug 2010) | 2 lines res_stun_monitor and corresponding options CHANGES documentation ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@282272 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-08-12Merged revisions 282066 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r282066 | russell | 2010-08-12 15:41:17 -0500 (Thu, 12 Aug 2010) | 4 lines Add a "core reload" CLI command. Review: https://reviewboard.asterisk.org/r/859/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@282067 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-08-12Merged revisions 282047 via svnmerge from David Vossel
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r282047 | dvossel | 2010-08-12 15:15:41 -0500 (Thu, 12 Aug 2010) | 35 lines improved translation paths for wideband codecs The problem I'm addressing is that Asterisk's current method of building the least cost translation paths between codecs does not take into account sample rate. For instance, it was possible for siren14 (a 32khz codec), to contain the a translation path to siren7 (a 16khz audio codec) that goes through slin at 8khz. In this case Asterisk takes a 32khz codec, down samples it to 8khz and then up samples it to 16khz which is terrible regardless if it is computationally less expensive. This patch now builds translation paths that give priority to maintaining the best possible sample rate before taking into consideration computational cost. This patch also adds cli commands to expose what translation paths are actually being used. Changes: 1. Translation paths will never contain a step that changes the sample rate unless absolutely necessary. 2. When choosing the best codec to make two channels compatible. Shared codecs with the highest sample rate are given priority. 3. A new cli command to show all translation paths available for a specific codec 'core show translation paths [codec name]' has been added. 4. 'core show translation' which displays the translation matrix now includes the new higher bit audio codecs in the table. 5. 'core show channel [channel name]' now displays the translation paths if translation is used. (closes issue #16841) Reported by: dvossel Review: https://reviewboard.asterisk.org/r/842/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@282048 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-08-03Merged revisions 280809 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r280809 | tilghman | 2010-08-03 15:25:10 -0500 (Tue, 03 Aug 2010) | 12 lines Sneak FIELDNUM() into 1.8. Returns a 1-based index into a list of a specified item. Matches up with FIELDQTY() and CUT(). (closes issue #17713) Reported by: gareth Patches: svn-279754.diff uploaded by gareth (license 208) Tested by: gareth, tilghman Review: https://reviewboard.asterisk.org/r/810/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@280810 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-29PeerStatus now includes Address and PortPaul Belanger
(closes issue #17730) Reported by: jkroon Patches: iax2-peerstate-address.patch uploaded by jkroon (license 714) Tested by: lmadsen git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@280555 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-27Make a formatting change. (Demonstrating the commit IRC bot to pabelanger)Russell Bryant
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@279725 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-26Merged revisions 279689 via svnmerge from Paul Belanger
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r279689 | pabelanger | 2010-07-26 19:29:34 -0400 (Mon, 26 Jul 2010) | 2 lines Updated documentation for FAX logger level. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@279692 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-26Merged revisions 279566 via svnmerge from Paul Belanger
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r279566 | pabelanger | 2010-07-26 15:51:39 -0400 (Mon, 26 Jul 2010) | 8 lines Add documentation for FAX logger level. (closes issue #17715) Reported by: vrban Patches: 17715.patch uploaded by pabelanger (license 224) Tested by: vrban ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@279567 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-23Start a new section in CHANGES for 1.10.Russell Bryant
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@279116 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-23Merge the realtime failover branchTilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278957 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-20Separate queue_log arguments into separate fields, and allow the text file ↵Tilghman Lesher
to be used, even when realtime is used. (closes issue #17082) Reported by: coolmig Patches: 20100720__issue17082.diff.txt uploaded by tilghman (license 14) Tested by: coolmig git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278307 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-16Add ability to configure the Max-Forwards header in the dialplan, as well as inOlle Johansson
sip.conf configuration for the channel and for devices. The Max-Forwards header is used to prevent loops in a SIP network. Each intermediary, like SIP proxys and SBCs, decrement this counter and detects when it reaches zero, at which point the SIP request is nicely killed in a SIP-friendly way. Review: https://reviewboard.asterisk.org/r/778/ Thanks to dvossel for the review and good advice. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276951 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-16Add a dialplan function to check if a queue exists: QUEUE_EXISTSOlle Johansson
Review: https://reviewboard.asterisk.org/r/777/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276950 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-13FILE() now supports line-mode and writing (altering) files.Tilghman Lesher
(closes issue #16461) Reported by: skyman Patches: 20100622__issue16461.diff.txt uploaded by tilghman (license 14) Tested by: tilghman Review: https://reviewboard.asterisk.org/r/737/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276114 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-10Make indentation consistent, move some queue features to the queue section.Russell Bryant
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275467 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-10Add support for devices with less than 3 lines on the LCD.Russell Bryant
(closes issue #17600) Reported by: minaguib Patches: ast_unistim_height_v2.patch uploaded by minaguib (license 1078) Tested by: minaguib git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275466 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-09Include rdnis in msgXXXX.txt file.Paul Belanger
(closes issue #17566) Reported by: outcast Patches: voicemail-rdnis.patch uploaded by outcast (license 1071) Tested by: outcast git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275307 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-08Add IPv6 to Asterisk.Mark Michelson
This adds a generic API for accommodating IPv6 and IPv4 addresses within Asterisk. While many files have been updated to make use of the API, chan_sip and the RTP code are the files which actually support IPv6 addresses at the time of this commit. The way has been paved for easier upgrading for other files in the near future, though. Big thanks go to Simon Perrault, Marc Blanchet, and Jean-Philippe Dionne for their hard work on this. (closes issue #17565) Reported by: russell Patches: asteriskv6-test-report.pdf uploaded by russell (license 2) Review: https://reviewboard.asterisk.org/r/743 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274783 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-07Also run the externnotify script when the pollmailboxes thread notices a change.Tilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274491 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-22Add regular expression filtering for manager events.Jeff Peeler
This patch as documented in the sample config allows one to optionally apply white, black, or both types of filtering to manager events. The new 'eventfilter' option is set per user. (closes issue #14861) Reported by: fnordian Patches: eventfilter3.patch uploaded by fnordian (license 110), modified by me Review: https://reviewboard.asterisk.org/r/673/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271868 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-22Updated the CHANGES file documenting the addition of a configurable port in ↵Matthew Nicholson
the dundi config file. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271764 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-21Add new application for declining counting words in multiple languages.Tilghman Lesher
(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
2010-06-17adds support for slin16 in sipDavid Vossel
(closes issue #16153) Reported by: kfister Patches: 16153-1.6.2.0-rc5.patch uploaded by kfister (license 912) slin16.sip.patch.1 uploaded by malcolmd (license 924) Tested by: kfister, malcolmd git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271261 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-17adds speex 16khz audio supportDavid Vossel
(closes issue #17501) Reported by: fabled Patches: asterisk-trunk-speex-wideband-v2.patch uploaded by fabled (license 448) Tested by: malcolmd, fabled, dvossel git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271231 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-16addition of G.719 pass-through supportDavid Vossel
(closes issue #16293) Reported by: malcolmd Patches: g719.passthrough.patch.7 uploaded by malcolmd (license 924) format_g719.c uploaded by malcolmd (license 924) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270940 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-16MSG_OOB flag on HANGUP packet removed.Paul Belanger
Per Tilghman's request on IRC (#asterisk-bugs). (closes issue #17506) Reported by: brycebaril Tested by: pabelanger, tilghman git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270936 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-15Add distributed devicestate via the XMPP protocol.Tilghman Lesher
(closes issue #15757) Reported by: Marquis Patches: distributed_devstate-XMPP.txt uploaded by lmadsen (license 10) Tested by: Marquis, lmadsen, marcelloceschia Review: https://reviewboard.asterisk.org/r/351/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270519 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-11Add DBGetComplete event after a DBGetResponse.Tilghman Lesher
(closes issue #16965) Reported by: rrb3942 Patches: DBGetComplete.patch uploaded by rrb3942 (license 1003) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@269938 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-09dial by name in chan_dahdiTzafrir Cohen
* chan_dahdi supports dialing configuring and dialing by device file name. DAHDI/span-name!local!1 will use /dev/dahdi/span-name/local/1 . Likewise it may appear in chan_dahdi.conf as 'channel => span-name!local!1'. * A new options for chan_dahdi.conf: 'ignore_failed_channels'. Boolean. False by default. If set, chan_dahdi will ignore failed 'channel' entries. Handy for the above name-based syntax as it does not depend on initialization order. * have my_pri_make_cc_dialstring() only manupulate dial-strings of group (gGrR) dialing, which make it lsightly more complicated. https://reviewboard.asterisk.org/r/535/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@269238 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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