]> git.ipfire.org Git - thirdparty/asterisk.git/commitdiff
Re-invite RTP during a masquerade so that, for instance, an AMI
authorMark Michelson <mmichelson@digium.com>
Thu, 24 Apr 2008 21:35:08 +0000 (21:35 +0000)
committerMark Michelson <mmichelson@digium.com>
Thu, 24 Apr 2008 21:35:08 +0000 (21:35 +0000)
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)

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@114632 65c4cc65-6c06-0410-ace0-fbb531ad65f3

channels/chan_sip.c

index d6d5411c30cb0fbdf3e6e8fdc4f2c5ac5abfa60f..e3fab92323c4c21d83f03ea99677028e2979f076 100644 (file)
@@ -3804,6 +3804,13 @@ 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;
        }
        if (option_debug > 2)