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chan_sip: Isolate code that manages struct sip_route.
[asterisk/asterisk.git]
/
channels
/
sip
/
include
/
sip.h
2014-02-10
Corey Farrell
chan_sip: Isolate code that manages struct sip_route.
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commitdiff
2013-12-19
Richard Mudgett
Voicemail: Remove mailbox identifier format (box@contex...
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2013-11-08
Scott Griepentrog
chan_sip: keep same local (from) tag for outgoing regis...
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2013-09-27
Jonathan Rose
chan_sip: Reject calls on 200 OKs if no SDP has been...
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2013-09-12
Jonathan Rose
chan_sip: Revert r398835 due to failing tests involving...
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diff to current
2013-09-11
Jonathan Rose
chan_sip: Reject calls without prior SDP on 200 OK
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2013-08-30
David M. Lee
optional_api: Fix linking problems between modules...
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diff to current
2013-08-19
Matthew Jordan
Whitespace cleanup
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diff to current
2013-07-01
Kinsey Moore
Refactor extraneous channel events
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2013-06-22
Joshua Colp
Merge in current pimp_my_sip work, including:
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2013-06-06
Mark Michelson
Refactor the features configuration scheme.
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2013-05-28
Mark Michelson
Add attended transfer support for chan_sip.c
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2013-05-17
Jonathan Rose
Stasis: Update security events to use Stasis
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2013-05-08
David M. Lee
Initial support for endpoints.
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2013-05-02
Alec L Davis
chan_sip: Session-Expires: Set timer to correctly expir...
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2013-03-27
Matthew Jordan
AST-2013-003: Prevent username disclosure in SIP channe...
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2013-03-16
Kinsey Moore
Transition MWI to Stasis-core
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2013-03-11
Kevin Harwell
Added an option to disallow music on hold
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2013-03-05
Matthew Jordan
Add RFC 3327 Path header support to chan_sip
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2013-02-12
Kinsey Moore
Fix some more REF_DEBUG-related build errors
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2013-01-02
Automerge script
Merged revisions 378288 via svnmerge from
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2013-01-02
Matthew Jordan
Resolve crashes due to large stack allocations when...
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2012-12-13
Mark Michelson
Resolve conflict and reset automerge.
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2012-12-13
Brent Eagles
This change adds a SIP peer configuration feature to...
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2012-11-20
Automerge script
Merged revisions 376551 via svnmerge from
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2012-11-20
Mark Michelson
Add "Require: timer" to 200 OK responses when appropriate.
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2012-09-25
Terry Wilson
Properly handle UAC/UAS roles for SIP session timers
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2012-09-20
Joshua Colp
Add support for DTLS-SRTP to res_rtp_asterisk and chan_sip.
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2012-08-09
Mark Michelson
Extend extension state callbacks to have more information.
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2012-08-07
Matthew Jordan
Add named callgroups/pickupgroups
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2012-08-03
Mark Michelson
Multiple revisions 370769-370771
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2012-07-31
Mark Michelson
Add headers from SIPAddHeader to outbound REFER requests.
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2012-07-31
Kinsey Moore
Clean up chan_sip
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2012-07-22
Joshua Colp
Prevent multiple local candidates from being added...
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2012-07-16
Joshua Colp
Add support for SIP over WebSocket.
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2012-07-11
Jonathan Rose
Named ACLs: Introduces a system for creating and sharin...
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2012-07-03
Terry Wilson
Better handle re-INVITEs with provisional but no final...
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2012-06-27
Terry Wilson
AST-2012-010: Clean up after a reinvite that never...
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2012-06-25
Mark Michelson
Re-fix how local tag is generated when sending a 481...
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2012-06-15
Kinsey Moore
Allow chan_sip to decline unwanted media streams
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2012-06-04
Mark Michelson
Merge changes dealing with support for Digium phones.
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2012-06-01
Mark Michelson
Help mitigate potential reinvite glare scenarios.
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2012-05-23
Matthew Jordan
Re-add LastMsgsSent value for SIP peers
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2012-05-17
Jonathan Rose
logger: Adds additional support for call id logging...
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2012-04-28
Joshua Colp
Add support for lightweight NAT keepalive.
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2012-03-16
Alec L Davis
Missed lastinvite CSeq int to uint32_t change
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2012-02-27
Jonathan Rose
Adds an option to sip.conf that prevents diversion...
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2012-02-24
Richard Mudgett
Fix worker thread resource leak in SIP TCP/TLS.
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diff to current
2012-02-09
Terry Wilson
Add auto_force_rport and auto_comedia NAT options
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2012-02-08
Terry Wilson
Add callbackextension matching & realtime callbackexten...
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2012-01-30
Alec L Davis
Merged revisions 353321 via svnmerge from
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2012-01-27
Alec L Davis
Merged revisions 352863 via svnmerge from
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2012-01-25
Richard Mudgett
Fixes for sending SIP MESSAGE outside of calls.
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2011-12-23
Jonathan Rose
INFO/Record request configurable to use dynamic features
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2011-12-23
Jonathan Rose
chan_sip autocreatepeer=persist option for auto-created...
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2011-11-14
Richard Mudgett
Restore SIP DTMF overlap dialing method.
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2011-11-10
Kinsey Moore
Fix several bugs with SDP parsing and well-formedness...
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2011-11-09
Terry Wilson
Don't treat a host:port string as a domain
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2011-11-03
Terry Wilson
Make room for the fax detect flags
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2011-11-02
Terry Wilson
Remove registertrying option in chan_sip
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2011-10-18
Richard Mudgett
More parking issues.
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2011-09-29
Gregory Nietsky
Merged revisions 338417 via svnmerge from
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2011-09-22
Jonathan Rose
Merged revisions 337595,337597 via svnmerge from
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2011-09-20
Gregory Nietsky
Merged revisions 336936 via svnmerge from
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2011-09-12
Olle Johansson
New sip.conf option for setting default tonezone for...
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2011-08-16
Matthew Nicholson
Merged revisions 332042 via svnmerge from
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diff to current
2011-08-16
Matthew Nicholson
Merged revisions 332027 via svnmerge from
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2011-07-25
Gregory Nietsky
Remove lastmsgssent from sip it has not been working...
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2011-07-05
Richard Mudgett
Merged revisions 326291 via svnmerge from
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2011-06-29
Kinsey Moore
Merged revisions 325740 via svnmerge from
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2011-06-13
David Vossel
Addition of "outofcall_message_context" sip.conf option.
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2011-06-01
Russell Bryant
Support routing text messages outside of a call.
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2011-05-20
Jonathan Rose
Merged revisions 319938 via svnmerge from
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2011-05-16
Gregory Nietsky
When a error in T.38 negotiation happens or its reject...
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2011-04-21
Matthew Nicholson
Merged revisions 314628 via svnmerge from
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2011-04-13
Leif Madsen
Add 'description' field for CLI and Manager output
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2011-02-03
David Vossel
Asterisk media architecture conversion - no more format...
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2011-01-26
Matthew Nicholson
Merged revisions 304245 via svnmerge from
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2010-09-15
Jeff Peeler
Merged revisions 286931 via svnmerge from
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2010-09-03
David Vossel
Merged revisions 285006 via svnmerge from
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2010-09-03
David Vossel
Merged revisions 284950 via svnmerge from
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2010-08-25
David Vossel
Merged revisions 283559 via svnmerge from
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2010-08-24
David Vossel
Merged revisions 283493 via svnmerge from
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2010-08-10
Russell Bryant
Merged revisions 281650 via svnmerge from
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2010-07-26
David Vossel
Merged revisions 279568 via svnmerge from
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2010-07-23
Mark Michelson
Allow IPv6 addresses for UDPTL streams.
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2010-07-19
Mark Michelson
Fix port setting of external address in SIP.
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2010-07-16
Olle Johansson
Add ability to configure the Max-Forwards header in...
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2010-07-13
David Vossel
chan_sip: RFC compliant retransmission timeout
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2010-07-08
Mark Michelson
Add IPv6 to Asterisk.
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2010-07-01
David Vossel
correct handling of get_destination return values
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2010-06-28
David Vossel
rfc compliant sip option parsing + new unit test
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diff to current
2010-06-22
Matthew Nicholson
Merged revisions 271689 via svnmerge from
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2010-06-17
David Vossel
retransmit response to BYE requests until timer J expires
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2010-06-08
Terry Wilson
Add SRTP support for Asterisk
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2010-06-07
Tilghman Lesher
Mailbox list would previously grow at each reload,...
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2010-06-02
Richard Mudgett
Generic Advice of Charge.
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2010-05-26
David Vossel
do all sip registry parsing before transmit_register
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2010-05-20
Terry Wilson
Add support for direct media ACLs
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2010-05-17
Mark Michelson
Enhancements to connected line and redirecting work.
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