chan_sip: Isolate code that manages struct sip_route.
[asterisk/asterisk.git] / channels / sip / include / sip.h
2014-02-10 Corey Farrellchan_sip: Isolate code that manages struct sip_route.
2013-12-19 Richard MudgettVoicemail: Remove mailbox identifier format (box@contex...
2013-11-08 Scott Griepentrogchan_sip: keep same local (from) tag for outgoing regis...
2013-09-27 Jonathan Rosechan_sip: Reject calls on 200 OKs if no SDP has been...
2013-09-12 Jonathan Rosechan_sip: Revert r398835 due to failing tests involving...
2013-09-11 Jonathan Rosechan_sip: Reject calls without prior SDP on 200 OK
2013-08-30 David M. Leeoptional_api: Fix linking problems between modules...
2013-08-19 Matthew JordanWhitespace cleanup
2013-07-01 Kinsey MooreRefactor extraneous channel events
2013-06-22 Joshua ColpMerge in current pimp_my_sip work, including:
2013-06-06 Mark MichelsonRefactor the features configuration scheme.
2013-05-28 Mark MichelsonAdd attended transfer support for chan_sip.c
2013-05-17 Jonathan RoseStasis: Update security events to use Stasis
2013-05-08 David M. LeeInitial support for endpoints.
2013-05-02 Alec L Davischan_sip: Session-Expires: Set timer to correctly expir...
2013-03-27 Matthew JordanAST-2013-003: Prevent username disclosure in SIP channe...
2013-03-16 Kinsey MooreTransition MWI to Stasis-core
2013-03-11 Kevin HarwellAdded an option to disallow music on hold
2013-03-05 Matthew JordanAdd RFC 3327 Path header support to chan_sip
2013-02-12 Kinsey MooreFix some more REF_DEBUG-related build errors
2013-01-02 Automerge scriptMerged revisions 378288 via svnmerge from
2013-01-02 Matthew JordanResolve crashes due to large stack allocations when...
2012-12-13 Mark MichelsonResolve conflict and reset automerge.
2012-12-13 Brent EaglesThis change adds a SIP peer configuration feature to...
2012-11-20 Automerge scriptMerged revisions 376551 via svnmerge from
2012-11-20 Mark MichelsonAdd "Require: timer" to 200 OK responses when appropriate.
2012-09-25 Terry WilsonProperly handle UAC/UAS roles for SIP session timers
2012-09-20 Joshua ColpAdd support for DTLS-SRTP to res_rtp_asterisk and chan_sip.
2012-08-09 Mark MichelsonExtend extension state callbacks to have more information.
2012-08-07 Matthew JordanAdd named callgroups/pickupgroups
2012-08-03 Mark MichelsonMultiple revisions 370769-370771
2012-07-31 Mark MichelsonAdd headers from SIPAddHeader to outbound REFER requests.
2012-07-31 Kinsey MooreClean up chan_sip
2012-07-22 Joshua ColpPrevent multiple local candidates from being added...
2012-07-16 Joshua ColpAdd support for SIP over WebSocket.
2012-07-11 Jonathan RoseNamed ACLs: Introduces a system for creating and sharin...
2012-07-03 Terry WilsonBetter handle re-INVITEs with provisional but no final...
2012-06-27 Terry WilsonAST-2012-010: Clean up after a reinvite that never...
2012-06-25 Mark MichelsonRe-fix how local tag is generated when sending a 481...
2012-06-15 Kinsey MooreAllow chan_sip to decline unwanted media streams
2012-06-04 Mark MichelsonMerge changes dealing with support for Digium phones.
2012-06-01 Mark MichelsonHelp mitigate potential reinvite glare scenarios.
2012-05-23 Matthew JordanRe-add LastMsgsSent value for SIP peers
2012-05-17 Jonathan Roselogger: Adds additional support for call id logging...
2012-04-28 Joshua ColpAdd support for lightweight NAT keepalive.
2012-03-16 Alec L DavisMissed lastinvite CSeq int to uint32_t change
2012-02-27 Jonathan RoseAdds an option to sip.conf that prevents diversion...
2012-02-24 Richard MudgettFix worker thread resource leak in SIP TCP/TLS.
2012-02-09 Terry WilsonAdd auto_force_rport and auto_comedia NAT options
2012-02-08 Terry WilsonAdd callbackextension matching & realtime callbackexten...
2012-01-30 Alec L DavisMerged revisions 353321 via svnmerge from
2012-01-27 Alec L DavisMerged revisions 352863 via svnmerge from
2012-01-25 Richard MudgettFixes for sending SIP MESSAGE outside of calls.
2011-12-23 Jonathan RoseINFO/Record request configurable to use dynamic features
2011-12-23 Jonathan Rosechan_sip autocreatepeer=persist option for auto-created...
2011-11-14 Richard MudgettRestore SIP DTMF overlap dialing method.
2011-11-10 Kinsey MooreFix several bugs with SDP parsing and well-formedness...
2011-11-09 Terry WilsonDon't treat a host:port string as a domain
2011-11-03 Terry WilsonMake room for the fax detect flags
2011-11-02 Terry WilsonRemove registertrying option in chan_sip
2011-10-18 Richard MudgettMore parking issues.
2011-09-29 Gregory NietskyMerged revisions 338417 via svnmerge from
2011-09-22 Jonathan RoseMerged revisions 337595,337597 via svnmerge from
2011-09-20 Gregory NietskyMerged revisions 336936 via svnmerge from
2011-09-12 Olle JohanssonNew sip.conf option for setting default tonezone for...
2011-08-16 Matthew NicholsonMerged revisions 332042 via svnmerge from
2011-08-16 Matthew NicholsonMerged revisions 332027 via svnmerge from
2011-07-25 Gregory NietskyRemove lastmsgssent from sip it has not been working...
2011-07-05 Richard MudgettMerged revisions 326291 via svnmerge from
2011-06-29 Kinsey MooreMerged revisions 325740 via svnmerge from
2011-06-13 David VosselAddition of "outofcall_message_context" sip.conf option.
2011-06-01 Russell BryantSupport routing text messages outside of a call.
2011-05-20 Jonathan RoseMerged revisions 319938 via svnmerge from
2011-05-16 Gregory Nietsky When a error in T.38 negotiation happens or its reject...
2011-04-21 Matthew NicholsonMerged revisions 314628 via svnmerge from
2011-04-13 Leif MadsenAdd 'description' field for CLI and Manager output
2011-02-03 David VosselAsterisk media architecture conversion - no more format...
2011-01-26 Matthew NicholsonMerged revisions 304245 via svnmerge from
2010-09-15 Jeff PeelerMerged revisions 286931 via svnmerge from
2010-09-03 David VosselMerged revisions 285006 via svnmerge from
2010-09-03 David VosselMerged revisions 284950 via svnmerge from
2010-08-25 David VosselMerged revisions 283559 via svnmerge from
2010-08-24 David VosselMerged revisions 283493 via svnmerge from
2010-08-10 Russell BryantMerged revisions 281650 via svnmerge from
2010-07-26 David VosselMerged revisions 279568 via svnmerge from
2010-07-23 Mark MichelsonAllow IPv6 addresses for UDPTL streams.
2010-07-19 Mark MichelsonFix port setting of external address in SIP.
2010-07-16 Olle JohanssonAdd ability to configure the Max-Forwards header in...
2010-07-13 David Vosselchan_sip: RFC compliant retransmission timeout
2010-07-08 Mark MichelsonAdd IPv6 to Asterisk.
2010-07-01 David Vosselcorrect handling of get_destination return values
2010-06-28 David Vosselrfc compliant sip option parsing + new unit test
2010-06-22 Matthew NicholsonMerged revisions 271689 via svnmerge from
2010-06-17 David Vosselretransmit response to BYE requests until timer J expires
2010-06-08 Terry WilsonAdd SRTP support for Asterisk
2010-06-07 Tilghman LesherMailbox list would previously grow at each reload,...
2010-06-02 Richard MudgettGeneric Advice of Charge.
2010-05-26 David Vosseldo all sip registry parsing before transmit_register
2010-05-20 Terry WilsonAdd support for direct media ACLs
2010-05-17 Mark MichelsonEnhancements to connected line and redirecting work.