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)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114633 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index dc43decc937ce088339be89e56f75914a5ec6b03..fb1f1967d1cebad8d35208f49f35f3802792f381 100644 (file)
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -5266,6 +5266,13 @@ static int sip_fixup(struct ast_channel *oldchan, struct ast_channel *newchan)
                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);