[asterisk-commits] mmichelson: trunk r114633 - in /trunk: ./ channels/chan_sip.c

[SVN commits to the Asterisk project]

Author: mmichelson
Date: Thu Apr 24 16:35:39 2008
New Revision: 114633

URL: http://svn.digium.com/view/asterisk?view=rev&rev=114633
Log:
Merged revisions 114632 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r114632 | mmichelson | 2008-04-24 16:35:08 -0500 (Thu, 24 Apr 2008) | 11 lines

Re-invite RTP during a masquerade so that, for instance, an AMI
redirect of two channels which are natively bridged will preserve audio
on both channels. This prevents a problem with Asterisk not re-inviting
due to one of the channels having being a zombie.

(closes issue #12513)
Reported by: mneuhauser
Patches:
      asterisk-1.4-114602_restore-RTP-on-fixup.patch uploaded by mneuhauser (license 425)


........

Modified:
    trunk/   (props changed)
    trunk/channels/chan_sip.c

Propchange: trunk/
------------------------------------------------------------------------------
Binary property 'branch-1.4-merged' - no diff available.

Modified: trunk/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/trunk/channels/chan_sip.c?view=diff&rev=114633&r1=114632&r2=114633
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Thu Apr 24 16:35:39 2008
@@ -5266,6 +5266,13 @@
 		ast_log(LOG_WARNING, "old channel wasn't %p but was %p\n", oldchan, p->owner);
 	else {
 		p->owner = newchan;
+		/* Re-invite RTP back to Asterisk. Needed if channel is masqueraded out of a native
+		   RTP bridge (i.e., RTP not going through Asterisk): RTP bridge code might not be
+		   able to do this if the masquerade happens before the bridge breaks (e.g., AMI
+		   redirect of both channels). Note that a channel can not be masqueraded *into*
+		   a native bridge. So there is no danger that this breaks a native bridge that
+		   should stay up. */
+		sip_set_rtp_peer(newchan, NULL, NULL, 0, 0);
 		ret = 0;
 	}
 	ast_debug(3, "SIP Fixup: New owner for dialogue %s: %s (Old parent: %s)\n", p->callid, p->owner->name, oldchan->name);