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2012-07-24chan_oss: fix "sample rate" error messageTzafrir Cohen
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
2012-07-24Remove code, that operate with cdr in attempt_transfer(). That was removed ↵Igor Goncharovskiy
somewhere between 1.2 and 1.4 and acidentaly put back in chan_unistim. (closes issue ASTERISK-19628) Reported by: Igor Olhovskiy git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370413 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-23Add separate configuration options for subscription and registration ↵Mark Michelson
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
2012-07-22Prevent multiple local candidates from being added with the same information ↵Joshua Colp
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
2012-07-20chan_iax2: Fix a segfault introduced by call ID loggingJonathan Rose
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
2012-07-20Fix build error in chan_misdn from commit 370316Kinsey Moore
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
2012-07-20Add hangupcause translation supportKinsey Moore
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
2012-07-19Add the ability to specify technology specific documentationMatthew Jordan
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
2012-07-18Ensure that all ast_datastore_info structures are 'const'.Kevin P. Fleming
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
2012-07-18Fix a crash occurring as a result of excess stack usage.Joshua Colp
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
2012-07-18Added option 'interdigit_timer' to unistim.conf to make able controll ↵Igor Goncharovskiy
hardcoded dial timeout constant. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370165 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-16Code cleanup and bugfix in chan_sip outboundproxy parsing.Walter Doekes
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
2012-07-16Fix a bug exposed by the testsuite where text streams would no longer be ↵Joshua Colp
parsed correctly. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370111 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-16Add support for SIP over WebSocket.Joshua Colp
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
2012-07-16Deactivate timer for dialing entered number on hook switch hang up.Igor Goncharovskiy
(closes issue ASTERISK-19554) Reported by: Stefano Villani git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370067 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-16Add French translation for chan_unistim phones on-screen menus. Igor Goncharovskiy
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370066 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-13Add support for parsing SDP attributes, generating SDP attributes, and ↵Joshua Colp
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
2012-07-12Add missing ast_hangup() calls on some analog exception paths.Richard Mudgett
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
2012-07-12Include Expires header for SIP PUBLISH requestsKinsey Moore
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
2012-07-12Prevent double uri_escaping in chan_sip when pedantic is enabledKinsey Moore
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
2012-07-11Named ACLs: Introduces a system for creating and sharing ACLsJonathan Rose
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
2012-07-11Do not consider failure to read the configuration file in chan_motif to be a ↵Joshua Colp
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
2012-07-11Fix validation errors when producing documentation using default build scriptMatthew Jordan
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
2012-07-10Add some additional documentation for core AMI eventsMatthew Jordan
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
2012-07-10Fix failing SDP_offer_answer testKinsey Moore
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
2012-07-10Add additional description stanza names from the old Google Talk protocol ↵Joshua Colp
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
2012-07-10Respect codec preference order when adding codecs to a media description.Joshua Colp
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
2012-07-10Add required items for Google video support.Joshua Colp
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
2012-07-09Add support for exposing the received contact URI and also for setting the ↵Joshua Colp
request URI in messages. (closes issue AST-911) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369847 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-09Force the clock rate of G.722 to be 16000 when using the Google transports ↵Joshua Colp
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
2012-07-09Fix dependency to be on res_xmpp. Long ago in a galaxy far far away it used ↵Joshua Colp
to use res_jabber. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369811 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-09chan_sip: Fix small behavioral change accidentally introduced in r369750Jonathan Rose
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
2012-07-07Add a new unified Jingle, Google Jingle, and Google Talk channel driver ↵Joshua Colp
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
2012-07-06Remove unnecessary generation of informational cause framesKinsey Moore
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
2012-07-06chan_sip: Add case for FLASH control frames so that we don't display a warning.Jonathan Rose
chan_sip channels can receive flash control frames when connected to analog phones and possibly for other reasons. There really isn't a reason to warn when these frames are received, we can safely ignore them. Patches: dahdi_sip_flash.diff uploaded by Jonathan Rose (license 6182) ........ Merged revisions 369750 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 369751 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369764 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-05Do not send a BYE when a provisional response arrives during a re-INVITEMatthew Jordan
Commits r369557 and r369579 were done to improve handling of re-INVITEs when the UA that was supposed to receive the re-INVITE fails to respond. A limitation of those patches occurred when a UA sent a provisional response to the re-INVITE. This triggered a sending of a BYE in check_pending. This patch tweaks the handling of the re-INVITE such that a BYE is not sent in response to those messages. (issue ASTERISK-19992) Reported by: Steve Davies Tested by: Steve Davies patches: (reinvite_tweak.diff license #5012 by Steve Davies) ........ Merged revisions 369626 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 369627 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369628 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-03More improvements to re-INVITEs timing out after a provisional responseTerry Wilson
There is no need to call check_pendings() on a final response to an INVITE when destroying the scheduler entry as it will be done later during normal processing. (issue ASTERISK-19992) ........ Merged revisions 369579 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 369580 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369581 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-03Better handle re-INVITEs with provisional but no final repsonsesTerry Wilson
A previous attempt at fixing this issue had negative side effects related to attended transfers which this patch should resolve. Many thanks to Steve Davies for all of the good suggestions and testing. (closes issue ASTERISK-19992) Reported by: Steve Davies Tested by: Steve Davies, Terry Wilson Review: https://reviewboard.asterisk.org/r/2009/ ........ Merged revisions 369557 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 369558 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369559 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-01Add support for ICE/STUN/TURN in res_rtp_asterisk and chan_sip.Joshua Colp
Review: https://reviewboard.asterisk.org/r/1891/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369517 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-29Hangup handlers - Dialplan subroutines that run when the channel hangs up.Richard Mudgett
Hangup handlers are an alternative to the h extension. They can be used in addition to the h extension. The idea is to attach a Gosub routine to a channel that will execute when the call hangs up. Whereas which h extension gets executed depends on the location of dialplan execution when the call hangs up, hangup handlers are attached to the call channel. You can attach multiple handlers that will execute in the order of most recently added first. (closes issue ASTERISK-19549) Reported by: Mark Murawski Tested by: rmudgett Review: https://reviewboard.asterisk.org/r/2002/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369493 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-29With some configurations a transport is not actually specified so assume UDP ↵Joshua Colp
in these cases. ........ Merged revisions 369490 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 369491 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369492 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-29Make the address family filter specific to the transport.Joshua Colp
(closes issue ASTERISK-16618) Reported by: Leif Madsen Review: https://reviewboard.asterisk.org/r/1667/ ........ Merged revisions 369471 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 369472 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369473 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-27AST-2012-010: Clean up after a reinvite that never gets a final responseTerry Wilson
The basic problem is that if a re-INVITE is sent by Asterisk and it receives a provisional response, but no final response, then the dialog is never torn down. In addition to leaking memory, this also leaks file descriptors and will eventually lead to Asterisk no longer being able to process calls. This patch just keeps track of whether there is an outstanding re-INVITE, and if there is goes ahead and cleans up everything as though there was no outstanding reinvite. Review: https://reviewboard.asterisk.org/r/2009/ (closes issue ASTERISK-19992) Reported by: Steve Davies Tested by: Steve Davies, Terry Wilson ........ Merged revisions 369436 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 369437 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369449 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-26Unique Call ID logging Phases III and IVJonathan Rose
Adds call ID logging changes to specific channel drivers that weren't handled handled in phase II of Call ID Logging. Also covers logging for threads for threads created by systems that may be involved with many different calls. Extra special thanks to Richard for rigorous review of chan_dahdi and its various signalling modules. review: https://reviewboard.asterisk.org/r/1927/ review: https://reviewboard.asterisk.org/r/1950/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369414 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-25Re-fix how local tag is generated when sending a 481 to an INVITE.Mark Michelson
Match our local tag to whatever to-tag was sent in the initial INVITE. Because the size of the to-tag may not fit in the buffer in the sip_pvt, it has been changed to a string field. (closes issue ASTERISK-19892) reported by Walter Doekes Review: https://reviewboard.asterisk.org/r/1977 ........ Merged revisions 369352 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 369353 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369367 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-25Be more consistent with the return code for requests received from invalid ↵Mark Michelson
domain. When Asterisk receives an INVITE from an external domain when allowexternaldomains=no send a 403 instead of a 404. This is consistent with Asterisk's behavior when receiving a REGISTER in this situation. (Closes issue ASTERISK-19601) Reported by Matthew Jordan Patches: ASTERISK-19601-no401.patch uploaded by Mark Michelson (License #5049) ........ Merged revisions 369302 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 369303 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369304 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-22Change incorrect chan_sip zombie hangup debug message. They are all zombies ↵Richard Mudgett
now. ........ Merged revisions 369235 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 369236 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369237 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-22Don't crash on a guest directmedia callTerry Wilson
A sip_pvt may not have relatedpeer set if a call doesn't match up with a peer. If there is no relatedpeer, there is no direct media ACL to apply, so just return that it is allowed. (closes issue ASTERISK-20040) Reported by: Terry Wilson ........ Merged revisions 369214 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 369215 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369217 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-22Fix wrong variable name in the R2 disconnect callbackKinsey Moore
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369216 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-22Don't parse media stream state for SIP video streamsKinsey Moore
The sendonly/recvonly/sendrecv/inactive media stream attributes were parsed for video, but nothing was ever done with them. With this code removed, an UNSUPPORTED message is produced when these attributes are used in conjunction with a video stream which is the better behavior since they were never really supported in the first place. ........ Merged revisions 369195 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 369206 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369207 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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