sysmocom
/
asterisk
11
0
Fork 0
Code Issues Pull Requests Packages Projects Releases Wiki Activity
24,081 Commits 141 Branches 1,481 Tags 406 MiB
Commit Graph

6 Commits

Author SHA1 Message Date
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
Joshua Colp 94ec267888 Migrate PeerStatus events to stasis, add stasis endpoints, and add chan_pjsip device state. 
(closes issue ASTERISK-21489)
(closes issue ASTERISK-21503)

Review: https://reviewboard.asterisk.org/r/2601/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@392538 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2013-06-22 12:40:16 +00:00
Kinsey Moore 1b5a3069f9 Add base XML documentation for res_sip 
Thanks to Brad Latus, this patch adds a significant amount much-needed
documentation to res_sip. It should cover all existing configuration
options currently in Asterisk trunk.

Patch-by: Brad Latus (snuffy)
Review: https://reviewboard.asterisk.org/r/2471/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389148 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2013-05-19 17:45:42 +00:00
Jonathan Rose b90bba7a30 Stasis: Update security events to use Stasis 
Also moves ACL messages to the security topic and gets rid of the
ACL topic

(closes issue ASTERISK-21103)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2496/



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388975 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2013-05-17 17:36:10 +00:00
Joshua Colp 3aac3d2ee5 Tweak res_sip priority so it gets loaded first before all other SIP stuff. 
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386760 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2013-04-27 19:03:39 +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