asterisk.git - Unnamed repository; edit this file 'description' to name the repository.

Age	Commit message (Expand)	Author
2010-05-20	Add support for direct media ACLs	Terry Wilson
2010-05-17	Enhancements to connected line and redirecting work.	Mark Michelson
2010-05-06	Permit more lines within a SIP body to be parsed.	Tilghman Lesher
2010-05-05	Prevent unnecessary warnings when getting rtpsource or rtpdest.	Mark Michelson
2010-04-28	Don't override peer context with domain context.	Mark Michelson
2010-04-27	Be more explicit about field naming in a test.	Jason Parker
2010-04-09	Add routines for parsing SIP URIs consistently.	Mark Michelson
2010-04-09	Merge Call completion support into trunk.	Mark Michelson
2010-03-25	Add new rtpsource options to the CHANNEL function.	Mark Michelson
2010-03-25	Improve handling of T.38 re-INVITEs that arrive before a T.38-capable	Kevin P. Fleming
2010-03-12	Only change the RTP ssrc when we see that it has changed	Terry Wilson
2010-02-17	Make all of the various rtpqos parameters in this branch available from the C...	Tilghman Lesher
2010-02-15	chan_sip parse code refactoring plus two new unit tests	David Vossel
2010-02-11	fixes some test description formatting inconsistencies so log file looks nice	David Vossel
2010-02-10	additional parse_uri test and documentation	David Vossel
2010-02-09	Various updates to the unit test API.	Russell Bryant
2010-02-07	Tweak formatting and add minor updates to some comments.	Russell Bryant
2010-02-05	fixes issue with sip registry not having correct default expiry	David Vossel
2010-02-03	-----Changes -----	David Vossel