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path: root/include/asterisk/res_pjsip_session.h
AgeCommit message (Expand)Author
2018-01-24Remove redundant module checks and references.Corey Farrell
2017-12-22Remove as much trailing whitespace as possible.Sean Bright
2017-12-12chan_pjsip/res_pjsip: Add CHANNEL(pjsip,request_uri)Richard Mudgett
2017-10-30core / pjsip: Add support for grouping streams together.Joshua Colp
2017-10-04res_pjsip: Add REF_DEBUG info to module references.Corey Farrell
2017-10-04res_pjsip: Fix issues that prevented shutdown of modules.Corey Farrell
2017-09-21bridge: Change participant SFU streams when source streams change.Joshua Colp
2017-08-25res/res_pjsip_session: allow SDP answer to be regeneratedTorrey Searle
2017-08-04res_pjsip_session/_sdp_rtp: Handling of 'msid' is incorrectKevin Harwell
2017-08-01chan_pjsip: add a new function PJSIP_DTMF_MODETorrey Searle
2017-07-13res_pjsip: Add "webrtc" configuration optionKevin Harwell
2017-07-13res_rtp_asterisk / res_pjsip: Add support for BUNDLE.Joshua Colp
2017-06-28chan_pjsip: Add support for multiple streams of the same type.Mark Michelson
2017-06-13res_pjsip_refer/session: Calls dropped during transferKevin Harwell
2017-04-27res_pjsip_session: Add cleanup to ast_sip_session_terminateGeorge Joseph
2017-03-15Add rtcp-mux supportMark Michelson
2016-07-19res_pjsip: Add fax_detect_timeout endpoint option.Richard Mudgett
2016-06-09res_pjsip_session: Use distributor serializer for incoming calls.Richard Mudgett
2016-03-03res_pjsip_caller_id: Anonymize 'From' when caller id presentation is prohibitedGeorge Joseph
2015-08-28res_pjsip_sdp_rtp: Fix multiple keepalive scheduled items.Joshua Colp
2015-07-24pjsip: Add rtp_timeout and rtp_timeout_hold endpoint options.Joshua Colp
2015-07-20res_pjsip: Add rtp_keepalive endpoint option.Mark Michelson
2015-04-02pjsip: resolve compatibility problem with ast_sip_sessionScott Griepentrog
2015-03-10res_pjsip_refer: Fix occasional unexpected BYE sent after receiving a REFER.Richard Mudgett
2015-01-05pjsip: Add 'PJSIP_AOR' and 'PJSIP_CONTACT' dialplan functions.Joshua Colp
2014-12-12res_pjsip_session: Delay sending BYE if a re-INVITE transaction is in progress.Joshua Colp
2014-11-19res_pjsip_sdp_rtp: Add support for optimistic SRTP.Joshua Colp
2014-11-03chan_pjsip: Add support for passing hold and unhold requests through.Joshua Colp
2014-10-16PJSIP: Enforce module load dependenciesKinsey Moore
2014-10-03chan_pjsip: Fix deadlock when masquerading PJSIP channels.Richard Mudgett
2014-09-02Resolve race condition where channels enter dialplan application before media...Mark Michelson
2014-07-20media formats: re-architect handling of media for performance improvementsMatthew Jordan
2014-06-30Recorded merge of revisions 417677 from http://svn.asterisk.org/svn/asterisk/...Joshua Colp
2014-06-24Move eid functions to utils.c, mark netsock.h deprecatedCorey Farrell
2014-05-28res_pjsip_session: Fix leaked video RTP ports.Richard Mudgett
2014-02-26PJSIP: Fix some bad spacingKinsey Moore
2014-01-15PJSIP: Add Path header supportKinsey Moore
2013-12-11func_channel, chan_pjsip: Add CHANNEL read function support for chan_pjsipMatthew Jordan
2013-10-26chan_pjsip: Fix a crash when direct media is enabled and an ACK is received a...Joshua Colp
2013-07-31Fix remnants of the pjsip renamingKinsey Moore
2013-07-30The large GULP->PJSIP renaming effort.Mark Michelson
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