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Merged revisions 336307 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r336307 | jrose | 2011-09-16 16:09:20 -0500 (Fri, 16 Sep 2011) | 20 lines Merged revisions 336294 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r336294 | jrose | 2011-09-16 14:53:40 -0500 (Fri, 16 Sep 2011) | 13 lines Fix bad RTP media bridges in directmedia calls on peers separated by multiple Asterisk nodes. In a situation involving devices on separate Asterisk trunks, the remote RTP bridge would break when starting a call with directmedia. This patch queues a new type of control frame so that our RTP bridge loop can properly detect when these situations occur and check to see if peers need to be updated in order to send their media to the proper location. (Closes issue ASTERISK-18340) Reported by: Thomas Arimont (Closes issue ASTERISK-17725) Reported by: kwk Tested by: twilson, jrose ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@336311 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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@@ -265,6 +265,7 @@ enum ast_control_frame_type {
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AST_CONTROL_END_OF_Q = 29, /*!< Indicate that this position was the end of the channel queue for a softhangup. */
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AST_CONTROL_INCOMPLETE = 30, /*!< Indication that the extension dialed is incomplete */
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AST_CONTROL_MCID = 31, /*!< Indicate that the caller is being malicious. */
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AST_CONTROL_UPDATE_RTP_PEER = 32, /*!< Interrupt the bridge and have it update the peer */
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};
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enum ast_frame_read_action {
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