7483 Commits (b03e7cc4c723e417a926ec946d1a9b9f098b30d4)

Author	SHA1	Message	Date
Richard Mudgett	062becab80	Convert sig_analog to use a global callback table. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370926 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	f1dce57742	Fix the analog dial *0 flash-hook of bridged peer feature. ... The flash-hook the bridged peer feature now correctly determines if the bridged peer is another chan_dahdi channel, that it is an analog channel, and that it has the correct signaling for an FXO port. It now also flash-hooks the correct channel. ........ Merged revisions 370900 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 370901 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370902 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	35bf5efeaf	Convert sig_pri to use a global callback table. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370893 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	f24be2740b	Convert sig_ss7 to use a global callback table. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370887 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Damien Wedhorn	f4d1b7ab12	Rewrite of skinny debugging. ... Debugging messages and associated controls only compiled in if configured with --enable-dev-mode. Debug messages provide more detail (including thread id) and are grouped so the user/dev can limit the type of messages displayed. Functionally no real change to chan_skinny. Review: https://reviewboard.asterisk.org/r/2040/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370881 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	15e41c7542	Reduce memory consumption significantly for users of the RTP engine API by storing only the payloads present and in use instead of every possible one. ... Review: https://reviewboard.asterisk.org/r/2052/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370832 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	5c4578f4ad	Add named callgroups/pickupgroups ... This patch adds named calledgroups/pickupgroups to Asterisk. Named groups are implemented in parallel to the existing numbered callgroup/pickupgroup implementation. However, unlike the existing implementation, which is limited to a maximum of 64 defined groups, the number of defined groups allowed for named callgroups/pickupgroups is effectively unlimited. Named groups are configured with the keywords "namedcallgroup" and "namedpickupgroup". This corresponds to the numbered group definitions of "callgroup" and "pickupgroup". Note that as the implementation of named groups coexists with the existing numbered implementation, a defined named group of "4" does not equate to numbered group 4. Support for the named groups has been added to the SIP, DAHDI, and mISDN channel drivers. Review: https://reviewboard.asterisk.org/r/2043 Uploaded by: Guenther Kelleter(license #6372) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370831 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	e46db5d943	Improve debug message for temporary outbound proxies. ... Thanks to Paul Belanger for pointing this out. ........ Merged revisions 370797 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 370798 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370801 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	9f0127f087	Multiple revisions 370769-370771 ... ........ r370769 | mmichelson | 2012-08-03 16:35:00 -0500 (Fri, 03 Aug 2012) | 24 lines Fix error in the "IPorHost" section of a SIP dialstring. This is based on the review request posted by Walter Doekes (referenced lower in the commit message) The main fix here is to treat the IPorHost portion of the dial string as a temporary outbound proxy. This ensures requests get sent to the proper location. Due to the age of the request, some parts were no longer relevant. For instance, the request moved outbound proxy parsing code into a single method. This is done in a previous commit, so it was not necessary to do again. Also, the review request fixed some errors with regards to request routing for CANCEL and ACK requests. This has also been fixed in more recent commits. (closes issue ASTERISK-19677) reported by Walter Doekes Review https://reviewboard.asterisk.org/r/1859 ........ r370770 | mmichelson | 2012-08-03 16:39:35 -0500 (Fri, 03 Aug 2012) | 3 lines Remove unused variable. ........ r370771 | mmichelson | 2012-08-03 16:43:52 -0500 (Fri, 03 Aug 2012) | 5 lines Seriously? Another compilation error fixed. Somebody beat me. ........ Merged revisions 370769-370771 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 370772 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370773 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	e108a5777a	Fix regression from r370636 ... When the chan_sip cleanup went in, a typo was included that caused some subscriptions of non-Polycom phones to be limited to the same capabilities as Polycom phones. This resolves the failures in the test suite resulting from this regression. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370740 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	4377d511ae	Add headers from SIPAddHeader to outbound REFER requests. ... This is a patch from kkm from review board. This is useful for adding headers to REFER requests that emanate from a Transfer() dialplan application call. This also fixes some uses of the Referred-by header, removing an extra set of angle brackets. I've modified the reporter's original patch to not require any additions to the sip_refer header and to just remove the referred_by_name from sip_refer since it is no longer needed or used. (closes Issue ASTERISK-17639) reported by Kirill Katsnelson Patches: 019059-sip-refer-addheaders-trunk-353549.diff uploaded by Kirill Katsnelson (license #5845) Review: https://reviewboard.asterisk.org/r/1159 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370691 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	d5d41741cc	Schedule pokes of registered SIP peers within a given timespan after SIP reload ... With a large number of SIP peers registered, performing a SIP reload causes a flood of SIP OPTIONS request packets. These are immediately sent out, and, as responses come back, can cause peers to be flagged as 'lagged' due to handling of the many response messages. This fix prevents this "packet storm" and schedules the pokes for a random time. That time varies between 1 ms and the peer's qualify time, or, if the qualify time is unknown, the global qualifyfreq setting. The committed patch has some very small modifications to the patch schmidts wrote for the review. (closes issue ASTERISK-19154) Reported by: Nicolo Mazzon patches: issue19154.patch license #6034 uploaded by schmidts Review: https://reviewboard.asterisk.org/r/1652 ........ Merged revisions 370666 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 370672 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370677 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	9b16c8b0f6	Clean up and ensure proper usage of alloca() ... This replaces all calls to alloca() with ast_alloca() which calls gcc's __builtin_alloca() to avoid BSD semantics and removes all NULL checks on memory allocated via ast_alloca() and ast_strdupa(). (closes issue ASTERISK-20125) Review: https://reviewboard.asterisk.org/r/2032/ Patch-by: Walter Doekes (wdoekes) ........ Merged revisions 370642 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 370643 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370655 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	e5210366e4	Clean up chan_sip ... This clean up was broken out from https://reviewboard.asterisk.org/r/1976/ and addresses the following: - struct sip_refer converted to use the stringfields API. - sip_{refer|notify}_allocate -> sip_{notify|refer}_alloc to match other *alloc functions. - Replace get_msg_text, get_msg_text2 and get_pidf_body -> No, not get_pidf_msg_text_body3 but get_content, to match add_content. - get_body doesn't get the request body, renamed to get_content_line. - get_body_by_line doesn't get the body line, and is just a simple if test. Moved code inline and removed function. - Remove camelCase in struct sip_peer peer state variables, onHold -> onhold, inUse -> inuse, inRinging -> ringing. - Remove camelCase in struct sip_request rlPart1 -> rlpart1, rlPart2 -> rlpart2. - Rename instances of pvt->randdata to pvt->nonce because that is what it is, no need to update struct sip_pvt because _it already has a nonce field_. - Removed struct sip_pvt randdata stringfield. - Remove useless (and inconsistent) 'header' suffix on variables in handle_request_subscribe. - Use ast_strdupa on Event header in handle_request_subscribe to avoid overly complicated strncmp calls to find the event package. - Move get_destination check in handle_request_subscribe to avoid duplicate checking for packages that don't need it. - Move extension state callback management in handle_request_subscribe to avoid duplicate checking for packages that don't need it. - Remove duplicate append_date prototype. - Rename append_date -> add_date to match other add_xxx functions. - Added add_expires helper function, removed code that manually added expires header. - Remove _header suffix on add_diversion_header (no other header adding functions have this). - Don't pass req->debug to request handle_request_XXXXX handlers if req is also being passed. - Don't pass req->ignore to check_auth as req is already being passed. - Don't create a subscription in handle_request_subscribe if p->expiry == 0. - Don't walk of the back of referred_by_name when splitting string in get_refer_info - Remove duplicate check for no dialog in handle_incoming when sipmethod == SIP_REFER, handle_request_refer checks for that. Review: https://reviewboard.asterisk.org/r/1993/ Patch-by: gareth git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370636 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	00d8fae66b	Release B channel allocation on error path in chan_misdn. ... ........ Merged revisions 370563 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 370564 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370565 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	3da07b3ec0	chan_sip: Add SIPpeerstatus command to AMI ... This patch was submitted by mnicholson a while back. It adds a new AMI action which allows users to request SIP peer status on demand similar to existing PeerStatus events and to the output you would see from CLI with sip show peer Review: https://reviewboard.asterisk.org/r/1098/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370518 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Tzafrir Cohen	6f8bb47833	chan_oss: fix "sample rate" error message ... Merged revisions 370428 from http://svn.asterisk.org/svn/asterisk/branches/1.8 Merged revisions 370432 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370433 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Igor Goncharovskiy	8eaba809ab	Remove code, that operate with cdr in attempt_transfer(). That was removed somewhere between 1.2 and 1.4 and acidentaly put back in chan_unistim. ...	13 years ago
Mark Michelson	a28e6fc7bd	Add separate configuration options for subscription and registration minexpiry and maxexpiry. ... This offers more fine-grained control over how long subscriptions last without negatively affecting the expiration range for registrations. Uploaded by: Guenther Kelleter(license #6372) Review: https://reviewboard.asterisk.org/r/2051 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370386 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	4d6b524b61	Prevent multiple local candidates from being added with the same information and add support for disabling ICE on a per-peer basis. ... (closes issue ASTERISK-20088) Reported by: wimpy Review: https://reviewboard.asterisk.org/r/2044/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370347 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	a5e10001b2	chan_iax2: Fix a segfault introduced by call ID logging ... Didn't previously check that a non NULL IAX channel was stored in the array at the requested position before attempting iax_pvt_callid_get (closes issue ASTERISK-20145) Reported by: Birger "WIMPy" Harzenetter git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370335 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	c2d9192660	Fix build error in chan_misdn from commit 370316 ... chan_misdn was not updated properly to account for a change in parameters for HANGUPCAUSE functionality. It now builds properly. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370328 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	cb9756daa2	Add hangupcause translation support ... The HANGUPCAUSE hash (trunk only) meant to replace SIP_CAUSE has now been replaced with the HANGUPCAUSE and HANGUPCAUSE_KEYS dialplan functions to better facilitate access to the AST_CAUSE translations for technology-specific cause codes. The HangupCauseClear application has also been added to remove this data from the channel. (closes issue SWP-4738) Review: https://reviewboard.asterisk.org/r/2025/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370316 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	86ff5585fd	Add the ability to specify technology specific documentation ... A number of applications/AMI commands in Asterisk have specific behavioral differences depending on the resource or channel technology those applications are executed on. For example, the MessageSend application/ command is technology agnostic, but how the channel drivers that support that functionality behave is dependant on the protocols and channel driver implementation. Prior to this patch, those details were either documented in the application/command documentation itself, or were left undocumented. This patch adds a new element to the documentation schema, <info/>. An info node is essentially a piece of technology specific reference information that can be included by any top level XML documentation node. For example, the MessageSend application can now include XMPP/SIP specific information, where that technology specific information can be defined in chan_motif/res_xmpp/ chan_sip. Likewise, that information can also be included in the MessageSend AMI command. Review: https://reviewboard.asterisk.org/r/2049 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370278 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kevin P. Fleming	79087cbbd5	Ensure that all ast_datastore_info structures are 'const'. ... While addressing a bug, I came across a instance of 'struct ast_datastore_info' that was not declared 'const'. Since the API already expects them to be 'const', this patch changes the declarations of all existing instances that were not already declared that way. ........ Merged revisions 370183 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 370184 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370187 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	cbdb2dbb0e	Fix a crash occurring as a result of excess stack usage. ... This fix involves moving the allocation of some temporary codec structures to the heap and also reduces the number of maximum payloads to something more sane for both regular and low memory builds. (closes issue ASTERISK-20140) Reported by: jonnt git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370171 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Igor Goncharovskiy	9278b5e51e	Added option 'interdigit_timer' to unistim.conf to make able controll hardcoded dial timeout constant. ...	13 years ago
Walter Doekes	6027b26fa7	Code cleanup and bugfix in chan_sip outboundproxy parsing. ... The bug was clearing the global outboundproxy when a peer-specific outboundproxy was bad. The cleanup reduces duplicate code. Review: https://reviewboard.asterisk.org/r/2034/ Reviewed by: Mark Michelson ........ Merged revisions 370131 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 370132 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370133 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	f234eae9ee	Fix a bug exposed by the testsuite where text streams would no longer be parsed correctly. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370111 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	e938737570	Add support for SIP over WebSocket. ... This allows SIP traffic to be exchanged over a WebSocket connection which is useful for rtcweb. Review: https://reviewboard.asterisk.org/r/2008 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370072 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Igor Goncharovskiy	f9c3585d73	Deactivate timer for dialing entered number on hook switch hang up. ...	13 years ago
Igor Goncharovskiy	95ac8f4743	Add French translation for chan_unistim phones on-screen menus. ...	13 years ago
Joshua Colp	a693fd1d87	Add support for parsing SDP attributes, generating SDP attributes, and passing it through. ... This support includes codecs such as H.263, H.264, SILK, and CELT. You are able to set up a call and have attribute information pass. This should help considerably with video calls. Review: https://reviewboard.asterisk.org/r/2005/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370055 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	9773d2351b	Add missing ast_hangup() calls on some analog exception paths. ... Make starting analog_ss_thread() or __analog_ss_thread() failure paths hangup the channel. ........ Merged revisions 370017 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 370025 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370037 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	c1354af599	Include Expires header for SIP PUBLISH requests ... RFC3903 requres SIP PUBLISH requests to have Expires headers, so add them. Review: https://reviewboard.asterisk.org/r/2003/ Patch-by: gareth ........ Merged revisions 370014 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 370015 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370016 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	65fe6976ae	Prevent double uri_escaping in chan_sip when pedantic is enabled ... If pedantic mode is enabled, outbound invites will have double-escaped contacts. This avoids setting an already-escaped string into a field where it is expected to be unescaped. (closes issue ASTERISK-20023) Reported by: Walter Doekes ........ Merged revisions 369993 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 369994 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369995 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	10afdf3a2a	Named ACLs: Introduces a system for creating and sharing ACLs ... This patch adds Named ACL functionality to Asterisk. This allows system administrators to define an ACL and refer to it by a unique name. Configurable items can then refer to that name when specifying access control lists. It also includes updates to all core supported consumers of ACLs. That includes manager, chan_sip, and chan_iax2. This feature is based on the deluxepine-trunk by Olle E. Johansson and provides a subset of the Named ACL functionality implemented in that branch. For more information on this feature, see acl.conf and/or the Asterisk wiki. Review: https://reviewboard.asterisk.org/r/1978/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369959 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	a25b4b7457	Do not consider failure to read the configuration file in chan_motif to be a show stopper for loading Asterisk by returning decline instead of failure. ... (closes issue ASTERISK-20103) Reported by: Terry Wilson git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369936 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	9bc2127d7b	Fix validation errors when producing documentation using default build script ... The awk script parses out the first instance of the DOCUMENTATION tag that it finds within a file. If a file did not previously have a DOCUMENTATION tag but received one due to it having an AMI event, then the XML fragment associated with the AMI event was erroneously placed in the resulting XML file. Without the python scripts, these XML fragments will not validate. This patch adds DOCUMENTATION tags at the top of those files that did not previously have them to prevent the awk script from pulling AMI event documentation. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369910 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	2ffae5745d	Add some additional documentation for core AMI events ... This patch adds some basic documentation for a number of modules. This includes core source files in Asterisk (those in main), as well as chan_agent, chan_dahdi, chan_local, sig_analog, and sig_pri. The DTD has also been updated to allow referencing of AMI commands. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369905 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	3805e2ae4d	Fix failing SDP_offer_answer test ... Asterisk now generates image stream declinations with the same transport case that it used to before the stream declination improvements. (udptl vs UDPTL) (closes issue SWP-4736) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369900 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	55871d3a67	Add additional description stanza names from the old Google Talk protocol which is used with Google Voice. ... (closes issue ASTERISK-20114) Reported by: Malcolm Davenport git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369898 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	74ebe6d5ab	Respect codec preference order when adding codecs to a media description. ... This change allows an endpoint in motif.conf to be configured with a preference of G.722 and fallback of ulaw. With Google this allows communication with Google Talk clients to use G.722 while when using Google Voice ulaw will be used. (closes issue ASTERISK-20114) Reported by: Malcolm Davenport git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369873 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	7296b670d4	Add required items for Google video support. ... This adds legacy STUN support for RTCP sockets, adds RTCP candidates to the Google transport information, and adds required codec parameters. (closes issue ASTERISK-20106) Reported by: Malcolm Davenport git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369864 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	7baa8bf43d	Add support for exposing the received contact URI and also for setting the request URI in messages. ... (closes issue AST-911) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369847 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	b46e1b45e4	Force the clock rate of G.722 to be 16000 when using the Google transports as it is 8000 elsewhere. ... (closes issue ASTERISK-20105) Reported by: Malcolm Davenport git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369838 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	fa0bcb6c70	Fix dependency to be on res_xmpp. Long ago in a galaxy far far away it used to use res_jabber. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369811 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	60bc927579	chan_sip: Fix small behavioral change accidentally introduced in r369750 ... When removing the warning for AST_CONTROL_FLASH from sip_indicate, I also inadvertently changed the return value, which would likely make the indication not be sent in audio. This fixes that while still removing the warning message. ........ Merged revisions 369792 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 369793 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369794 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	a3fa37b8cf	Add a new unified Jingle, Google Jingle, and Google Talk channel driver written from scratch called chan_motif. ... This channel driver is a replacement for both chan_gtalk and chan_jingle but adds additional features not found in either. These features include full configuration reload, video, full codec support, bidirectional cause code mapping, hold, unhold, and ringing indication. It is also compliant with the current published Jingle and Google Jingle specifications. The original Google Talk protocol is also supported for Google Voice interoperability. You may ask yourself though where the name motif comes from... and I would say to you... music! motif: a perceivable or salient recurring fragment or succession of notes Sorta like a jingle! Review: https://reviewboard.asterisk.org/r/1917/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369769 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	db59a3f123	Remove unnecessary generation of informational cause frames ... It is not necessary to generate information cause code frames on every protocol event that occurs. This removes all the instances where the frame was not conveying a cause code and was instead just conveying a protocol-specific message. This also corrects the generation of the message associated with disconnects for MFC/R2 to use the MFC/R2 specific text for the disconnect cause. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369765 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago