[asterisk-commits] russell: branch 1.6.0 r108354 - in /branches/1.6.0: ./ channels/chan_sip.c

[SVN commits to the Asterisk project]

Author: russell
Date: Wed Mar 12 17:52:58 2008
New Revision: 108354

URL: http://svn.digium.com/view/asterisk?view=rev&rev=108354
Log:
Merged revisions 107157 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/trunk

........
r107157 | file | 2008-03-10 15:00:21 -0500 (Mon, 10 Mar 2008) | 4 lines

If we receive a 488 on a T38 request reinvite back to audio. As well reinvite across a bridge back to audio if one side doesn't negotiate to T38.
(closes issue #8677)
Reported by: alex-911

........

Modified:
    branches/1.6.0/   (props changed)
    branches/1.6.0/channels/chan_sip.c

Propchange: branches/1.6.0/
------------------------------------------------------------------------------
Binary property 'trunk-merged' - no diff available.

Modified: branches/1.6.0/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/branches/1.6.0/channels/chan_sip.c?view=diff&rev=108354&r1=108353&r2=108354
==============================================================================
--- branches/1.6.0/channels/chan_sip.c (original)
+++ branches/1.6.0/channels/chan_sip.c Wed Mar 12 17:52:58 2008
@@ -5133,6 +5133,8 @@
 					transmit_reinvite_with_sdp(p, TRUE, FALSE);
 				}
 				break;
+			case AST_T38_TERMINATED:
+			case AST_T38_REFUSED:
 			case AST_T38_REQUEST_TERMINATE:		/* Shutdown T38 */
 				if (p->t38.state == T38_ENABLED)
 					transmit_reinvite_with_sdp(p, FALSE, FALSE);
@@ -14714,24 +14716,13 @@
 		break;
 	case 488: /* Not acceptable here */
 		xmitres = 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 .
-			*/
+		if (p->udptl && p->t38.state == T38_LOCAL_REINVITE) {
 			change_t38_state(p, 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 && !req->ignore)
-				ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
-			p->needdestroy = 1;
+
+			/* Trigger a reinvite back to audio */
+			transmit_reinvite_with_sdp(p, FALSE, FALSE);
 		} else if (p->udptl && p->t38.state == T38_LOCAL_DIRECT) {
 			/* We tried to send T.38 out in an initial INVITE and the remote side rejected it,
 			   right now we can't fall back to audio so totally abort.