Commit Graph

9 Commits

Author SHA1 Message Date
Mark Michelson 895c8e0d2c Reorganize the ast_sip_endpoint structure into substructures.
(closes issue ASTERISK-22135)
reported by Matt Jordan

Review: https://reviewboard.asterisk.org/r/2707



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395748 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-07-30 15:17:56 +00:00
Kinsey Moore 98504fec8e Add DTLS-SRTP support to chan_pjsip
This patch introduces DTLS-SRTP support to chan_pjsip and the options
necessary to configure it including an option to allow choosing between
32 and 80 byte SRTP tag lengths.

During the implementation and testing of this patch, three other bugs
were found and their fixes are included with this patch. The two in
chan_sip were a segfault relating to DTLS setup and mistaken call
rejection. The third bug fix prevents chan_pjsip from attempting to
perform bridge optimization between two endpoints if either of them is
running any form of SRTP.

Review: https://reviewboard.asterisk.org/r/2683/
(closes issue ASTERISK-21419)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395121 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-07-23 13:52:06 +00:00
Mark Michelson c47787feab Add a bunch of options from sip.conf to res_sip.conf
For a complete list of the options added, see the review linked
at the bottom of this commit message.

(closes issue ASTERISK-21506)
reported by Matt Jordan

Review: https://reviewboard.asterisk.org/r/2671



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394759 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-07-18 19:25:51 +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
Jason Parker 154fbf8cae Split Hold event into Hold/Unhold, and move it into core.
(closes issue ASTERISK-21487)
Review: https://reviewboard.asterisk.org/r/2565/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389746 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-24 21:21:25 +00:00
Joshua Colp 61927f7c82 Fix a bug where the DTMF mode was not set on newly created RTP instances in the res_sip_sdp_rtp module.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389568 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-23 18:39:05 +00:00
Joshua Colp 7f43acc49c Fix a bug with applying the end result of the codec negotiation to the Asterisk channel.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389567 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-23 18:19:27 +00:00
Joshua Colp e236c2ea36 Use the configured formats for Gulp sessions if there are no joint formats between requested formats and configured formats.
(closes issue ASTERISK-21756)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@387545 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-03 11:35:14 +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