[asterisk-commits] oej: branch 1.4 r48326 - in /branches/1.4: ./ channels/chan_sip.c

[asterisk-commits at lists.digium.com]

Author: oej
Date: Wed Dec  6 10:25:06 2006
New Revision: 48326

URL: http://svn.digium.com/view/asterisk?view=rev&rev=48326
Log:
Issue #8258 - fix handling of 487 being retransmitted to Asterisk

Modified:
    branches/1.4/   (props changed)
    branches/1.4/channels/chan_sip.c

Propchange: branches/1.4/
------------------------------------------------------------------------------
Binary property 'branch-1.2-merged' - no diff available.

Modified: branches/1.4/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/branches/1.4/channels/chan_sip.c?view=diff&rev=48326&r1=48325&r2=48326
==============================================================================
--- branches/1.4/channels/chan_sip.c (original)
+++ branches/1.4/channels/chan_sip.c Wed Dec  6 10:25:06 2006
@@ -11748,7 +11748,16 @@
 			ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
 		sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
 		break;
-
+	case 487: /* Cancelled transaction */
+		/* We have sent CANCEL on an outbound INVITE 
+			This transaction is already scheduled to be killed by sip_hangup().
+		*/
+		transmit_request(p, SIP_ACK, seqno, 0, 0);
+		if (p->owner && !ast_test_flag(req, SIP_PKT_IGNORE))
+			ast_queue_hangup(p->owner);
+		else if (!ast_test_flag(req, SIP_PKT_IGNORE))
+			update_call_counter(p, DEC_CALL_LIMIT);
+		break;
 	case 491: /* Pending */
 		/* we really should have to wait a while, then retransmit */
 			/* We should support the retry-after at some point */
@@ -11760,6 +11769,7 @@
 		break;
 
 	case 501: /* Not implemented */
+		transmit_request(p, SIP_ACK, seqno, 0, 0);
 		if (p->owner)
 			ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
 		break;
@@ -12166,6 +12176,10 @@
 				/* Guessing that this is not an important request */
 			}
 			break;
+		case 487:
+			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);
@@ -12210,12 +12224,6 @@
 				case 603: /* Decline */
 					if (p->owner)
 						ast_queue_control(p->owner, AST_CONTROL_BUSY);
-					break;
-				case 487:	/* Response on INVITE that has been CANCELled */
-					/* channel now destroyed - dec the inUse counter */
-					if (owner)
-						ast_queue_hangup(p->owner);
-					update_call_counter(p, DEC_CALL_LIMIT);
 					break;
 				case 482: /*
 					\note SIP is incapable of performing a hairpin call, which