Commit Graph

3 Commits

Author SHA1 Message Date
Mark Michelson 7fdeb52910 Fix some broken logic in sending outbound caller ID.
* trust_id_outbound was required even when the caller ID was not marked
private. This is against intentions and documentation.
* We now check both name and number privacy instead of checking name privacy
twice.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393793 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-07-08 13:57:28 +00:00
Joshua Colp 77002bc377 Merge in current pimp_my_sip work, including:
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
2013-06-22 14:03:22 +00:00
Mark Michelson 74f2318051 Merge the pimp_my_sip branch into trunk.
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
2013-04-25 18:25:31 +00:00