diff options
author | Joshua Colp <jcolp@digium.com> | 2013-06-22 14:03:22 +0000 |
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committer | Joshua Colp <jcolp@digium.com> | 2013-06-22 14:03:22 +0000 |
commit | 77002bc377f19ea11e60732c486b6ef371688773 (patch) | |
tree | c19fd245c519c6d7905403849a7af9c7e4a4be3e /channels/sip/include/sip.h | |
parent | ea03516cb5426915d183526335d3a7d662ea29dc (diff) |
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
Diffstat (limited to 'channels/sip/include/sip.h')
-rw-r--r-- | channels/sip/include/sip.h | 6 |
1 files changed, 3 insertions, 3 deletions
diff --git a/channels/sip/include/sip.h b/channels/sip/include/sip.h index 0adde37f2..8b4672b25 100644 --- a/channels/sip/include/sip.h +++ b/channels/sip/include/sip.h @@ -1165,9 +1165,9 @@ struct sip_pvt { AST_LIST_HEAD_NOLOCK(request_queue, sip_request) request_queue; /*!< Requests that arrived but could not be processed immediately */ struct sip_invite_param *options; /*!< Options for INVITE */ struct sip_st_dlg *stimer; /*!< SIP Session-Timers */ - struct sip_srtp *srtp; /*!< Structure to hold Secure RTP session data for audio */ - struct sip_srtp *vsrtp; /*!< Structure to hold Secure RTP session data for video */ - struct sip_srtp *tsrtp; /*!< Structure to hold Secure RTP session data for text */ + struct ast_sdp_srtp *srtp; /*!< Structure to hold Secure RTP session data for audio */ + struct ast_sdp_srtp *vsrtp; /*!< Structure to hold Secure RTP session data for video */ + struct ast_sdp_srtp *tsrtp; /*!< Structure to hold Secure RTP session data for text */ int red; /*!< T.140 RTP Redundancy */ int hangupcause; /*!< Storage of hangupcause copied from our owner before we disconnect from the AST channel (only used at hangup) */ |