[asterisk-commits] rizzo: branch rizzo/astobj2 r51279 - /team/rizzo/astobj2/channels/chan_sip.c

[asterisk-commits at lists.digium.com]

Author: rizzo
Date: Fri Jan 19 02:49:42 2007
New Revision: 51279

URL: http://svn.digium.com/view/asterisk?view=rev&rev=51279
Log:
merge from trunk 50007 (all versions before already done):

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


Modified:
    team/rizzo/astobj2/channels/chan_sip.c

Modified: team/rizzo/astobj2/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/team/rizzo/astobj2/channels/chan_sip.c?view=diff&rev=51279&r1=51278&r2=51279
==============================================================================
--- team/rizzo/astobj2/channels/chan_sip.c (original)
+++ team/rizzo/astobj2/channels/chan_sip.c Fri Jan 19 02:49:42 2007
@@ -12428,6 +12428,34 @@
 		 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 */
@@ -12931,6 +12959,12 @@
 			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);
@@ -12987,7 +13021,6 @@
 						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 */