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Merged revisions 50006 via svnmerge from 

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

........
r50006 | oej | 2007-01-08 15:26:14 +0100 (Mon, 08 Jan 2007) | 11 lines

Issue #8677 - Handle failure of T.38 re-invite

This is not a fix, but adding an error message to tell the admin that
we have a bad configuration. We should not send T.38 re-invites to devices
that can't handle it (with the current architecture where you have to
hard-code t.38 support per device).

To really fix this, we need to figure out a way to tell the incoming
call that the re-invite failed, so we can signal failure on that
end and go back to the original call.

........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@50007 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Olle Johansson 2007-01-08 14:31:16 +00:00
parent 835e2ed684
commit 5d4f272a90
1 changed files with 31 additions and 1 deletions
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32
channels/chan_sip.c
View File

@ -11966,6 +11966,33 @@ static void handle_response_invite(struct sip_pvt *p, int resp, char *rest, stru
else if (!ast_test_flag(req, SIP_PKT_IGNORE))
update_call_counter(p, DEC_CALL_LIMIT);
break;
case 488: /* Not acceptable here */
transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
if (reinvite && p->udptl) {
/* If this is a T.38 call, we should go back to
audio. If this is an audio call - something went
terribly wrong since we don't renegotiate codecs,
only IP/port .
*/
p->t38.state = T38_DISABLED;
/* Try to reset RTP timers */
ast_rtp_set_rtptimers_onhold(p->rtp);
ast_log(LOG_ERROR, "Got error on T.38 re-invite. Bad configuration. Peer needs to have T.38 disabled.\n");
/*! \bug Is there any way we can go back to the audio call on both
sides here?
*/
/* While figuring that out, hangup the call */
if (p->owner && !ast_test_flag(req, SIP_PKT_IGNORE))
ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);
} else {
/* We can't set up this call, so give up */
if (p->owner && !ast_test_flag(req, SIP_PKT_IGNORE))
ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);
}
break;
case 491: /* Pending */
/* we really should have to wait a while, then retransmit */
/* We should support the retry-after at some point */
@ -12404,6 +12431,10 @@ static void handle_response(struct sip_pvt *p, int resp, char *rest, struct sip_
if (sipmethod == SIP_INVITE)
handle_response_invite(p, resp, rest, req, seqno);
break;
case 488: /* Not acceptable here - codec error */
if (sipmethod == SIP_INVITE)
handle_response_invite(p, resp, rest, req, seqno);
break;
case 491: /* Pending */
if (sipmethod == SIP_INVITE)
handle_response_invite(p, resp, rest, req, seqno);
@ -12460,7 +12491,6 @@ static void handle_response(struct sip_pvt *p, int resp, char *rest, struct sip_
ast_string_field_build(p->owner, call_forward,
"Local/%s@%s", p->username, p->context);
/* Fall through */
case 488: /* Not acceptable here - codec error */
case 480: /* Temporarily Unavailable */
case 404: /* Not Found */
case 410: /* Gone */