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

Age	Commit message (Collapse)	Author
2013-06-22	Merge in current pimp_my_sip work, including:	Joshua Colp
	1. Security events 2. Websocket support 3. Diversion header + redirecting support 4. An anonymous endpoint identifier 5. Inbound extension state subscription support 6. PIDF notify generation 7. One touch recording support (special thanks Sean Bright!) 8. Blind and attended transfer support 9. Automatic inbound registration expiration 10. SRTP support 11. Media offer control dialplan function 12. Connected line support 13. SendText() support 14. Qualify support 15. Inband DTMF detection 16. Call and pickup groups 17. Messaging support Thanks everyone! Side note: I'm reminded of the song "How Far We've Come" by Matchbox Twenty. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@392565 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-04-25	Don't bind to anything in the sample configuration so we don't clash with ↵	Joshua Colp
	chan_sip on a "make samples" right now. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386577 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-04-25	Merge the pimp_my_sip branch into trunk.	Mark Michelson
	The pimp_my_sip branch is being merged at this point because it offers basic functionality, and from an API standpoint, things are complete. SIP work is not feature-complete; however, with the completion of the SUBSCRIBE/NOTIFY API, all APIs (except a PUBLISH API) have been created, and thus it is possible for developers to attempt to create new SIP work. API documentation can be found in the doxygen in the code, but usability documentation is still lacking. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386540 65c4cc65-6c06-0410-ace0-fbb531ad65f3