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Merged revisions 114632 via svnmerge from 

https://origsvn.digium.com/svn/asterisk/branches/1.4

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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: http://svn.digium.com/svn/asterisk/trunk@114633 f38db490-d61c-443f-a65b-d21fe96a405b
This commit is contained in:
mmichelson 2008-04-24 21:35:39 +00:00
parent a06caadecb
commit 5a42196d6b
1 changed files with 7 additions and 0 deletions
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7
channels/chan_sip.c
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@ -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);