diff options
| author | mmichelson <mmichelson@f38db490-d61c-443f-a65b-d21fe96a405b> | 2008-04-24 21:36:05 +0000 |
|---|---|---|
| committer | mmichelson <mmichelson@f38db490-d61c-443f-a65b-d21fe96a405b> | 2008-04-24 21:36:05 +0000 |
| commit | be28be088f986dd533024dd27229f0d394398678 (patch) | |
| tree | 07ebf053576391cc2a9f85b646614e6b53f21517 /channels | |
| parent | 1e8141a6de9d0eb72f9ddda7844a1a1d7c305ed5 (diff) | |
Merged revisions 114633 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk
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r114633 | mmichelson | 2008-04-24 16:35:39 -0500 (Thu, 24 Apr 2008) | 19 lines
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/branches/1.6.0@114634 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'channels')
| -rw-r--r-- | channels/chan_sip.c | 7 |
1 files changed, 7 insertions, 0 deletions
diff --git a/channels/chan_sip.c b/channels/chan_sip.c index 3cd353b30..559ba767c 100644 --- a/channels/chan_sip.c +++ b/channels/chan_sip.c @@ -4991,6 +4991,13 @@ 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); |