https://origsvn.digium.com/svn/asterisk/branches/10
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r336792 | twilson | 2011-09-19 17:13:34 -0500 (Mon, 19 Sep 2011) | 9 lines
Merged revisions 336791 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r336791 | twilson | 2011-09-19 17:07:58 -0500 (Mon, 19 Sep 2011) | 2 lines
Don't interfere with T.38 reinvites
This is an update to the fix for ASTERISK-18340 and ASTERISK-17725
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@336837
65c4cc65-6c06-0410-ace0-
fbb531ad65f3
"Channel: %s\r\nChanneltype: %s\r\nUniqueid: %s\r\nSIPcallid: %s\r\nSIPfullcontact: %s\r\nPeername: %s\r\n",
p->owner->name, "SIP", p->owner->uniqueid, p->callid, p->fullcontact, p->peername);
} else { /* RE-invite */
- ast_queue_control(p->owner, AST_CONTROL_UPDATE_RTP_PEER);
+ if (p->t38.state == T38_DISABLED || p->t38.state == T38_REJECTED) {
+ ast_queue_control(p->owner, AST_CONTROL_UPDATE_RTP_PEER);
+ } else {
+ ast_queue_frame(p->owner, &ast_null_frame);
+ }
}
} else {
/* It's possible we're getting an 200 OK after we've tried to disconnect