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path: root/channels/chan_sip.c
AgeCommit message (Expand)Author
2012-07-23Add separate configuration options for subscription and registration minexpir...Mark Michelson
2012-07-22Prevent multiple local candidates from being added with the same information ...Joshua Colp
2012-07-20Add hangupcause translation supportKinsey Moore
2012-07-19Add the ability to specify technology specific documentationMatthew Jordan
2012-07-18Fix a crash occurring as a result of excess stack usage.Joshua Colp
2012-07-16Code cleanup and bugfix in chan_sip outboundproxy parsing.Walter Doekes
2012-07-16Fix a bug exposed by the testsuite where text streams would no longer be pars...Joshua Colp
2012-07-16Add support for SIP over WebSocket.Joshua Colp
2012-07-13Add support for parsing SDP attributes, generating SDP attributes, and passin...Joshua Colp
2012-07-12Include Expires header for SIP PUBLISH requestsKinsey Moore
2012-07-12Prevent double uri_escaping in chan_sip when pedantic is enabledKinsey Moore
2012-07-11Named ACLs: Introduces a system for creating and sharing ACLsJonathan Rose
2012-07-10Fix failing SDP_offer_answer testKinsey Moore
2012-07-09Add support for exposing the received contact URI and also for setting the re...Joshua Colp
2012-07-09chan_sip: Fix small behavioral change accidentally introduced in r369750Jonathan Rose
2012-07-06chan_sip: Add case for FLASH control frames so that we don't display a warning.Jonathan Rose
2012-07-05Do not send a BYE when a provisional response arrives during a re-INVITEMatthew Jordan
2012-07-03More improvements to re-INVITEs timing out after a provisional responseTerry Wilson
2012-07-03Better handle re-INVITEs with provisional but no final repsonsesTerry Wilson
2012-07-01Add support for ICE/STUN/TURN in res_rtp_asterisk and chan_sip.Joshua Colp
2012-06-29With some configurations a transport is not actually specified so assume UDP ...Joshua Colp
2012-06-29Make the address family filter specific to the transport.Joshua Colp
2012-06-27AST-2012-010: Clean up after a reinvite that never gets a final responseTerry Wilson
2012-06-25Re-fix how local tag is generated when sending a 481 to an INVITE.Mark Michelson
2012-06-25Be more consistent with the return code for requests received from invalid do...Mark Michelson
2012-06-22Change incorrect chan_sip zombie hangup debug message. They are all zombies ...Richard Mudgett
2012-06-22Don't crash on a guest directmedia callTerry Wilson
2012-06-22Don't parse media stream state for SIP video streamsKinsey Moore
2012-06-19Fix request routing issue when outboundproxy is used.Mark Michelson
2012-06-15Allow chan_sip to decline unwanted media streamsKinsey Moore
2012-06-12Set the Caller ID "tag" on peers even if remote party information is present.Mark Michelson
2012-06-12Fix deadlock in SIP transfers that involve a REFER requestMatthew Jordan
2012-06-12Parse ANI2 information from SIP From header parametersKinsey Moore
2012-06-11Fix deadlock potential with ast_set_hangupsource() calls.Richard Mudgett
2012-06-11Fix coverity UNUSED_VALUE findings in core support level filesKinsey Moore
2012-06-06Fix a specific scenario where ACKs are not matched.Mark Michelson
2012-06-06Ensure overlapping hold flags do not conflictKinsey Moore
2012-06-05Convert AST_FLAG_ANSWERED_ELSEWHERE usage to AST_CAUSE_ANSWERED_ELSEWHEREKinsey Moore
2012-06-04Relay proper SIP responses on calling side.Mark Michelson
2012-06-04Merge changes dealing with support for Digium phones.Mark Michelson
2012-06-01Improve SDP offer/answer RFC complianceKevin P. Fleming
2012-06-01Improve SDP parsing warning messagesKevin P. Fleming
2012-06-01Help mitigate potential reinvite glare scenarios.Mark Michelson
2012-05-31Coverity Report: Fix issues for error type REVERSE_INULL (core modules)Richard Mudgett
2012-05-25Fix pvt_sip for inbound call to use peer's allowtransfer settingMichael L. Young
2012-05-24chan_sip: fix problem directmediapermit/deny uses the wrong addressJonathan Rose
2012-05-23Re-add LastMsgsSent value for SIP peersMatthew Jordan
2012-05-22Resolve crash in subscribing for MWI notificationsTerry Wilson
2012-05-21Revert revision 367163.Mark Michelson
2012-05-21Add "send to voicemail" Digium phone functionality to Asterisk.Mark Michelson
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