[asterisk-commits] oej: branch 1.4 r66503 - /branches/1.4/channels/chan_sip.c

asterisk-commits at lists.digium.com asterisk-commits at lists.digium.com
Tue May 29 12:32:58 MST 2007


Author: oej
Date: Tue May 29 14:32:57 2007
New Revision: 66503

URL: http://svn.digium.com/view/asterisk?view=rev&rev=66503
Log:
Properly handle 408 request timeout - according to the RFC, the dialog dies if a request in a dialog gets this response.

Modified:
    branches/1.4/channels/chan_sip.c

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=66503&r1=66502&r2=66503
==============================================================================
--- branches/1.4/channels/chan_sip.c (original)
+++ branches/1.4/channels/chan_sip.c Tue May 29 14:32:57 2007
@@ -11926,6 +11926,7 @@
 		sip_alreadygone(p);
 		break;
 
+	case 408: /* Request timeout */
 	case 481: /* Call leg does not exist */
 		/* Could be REFER caused INVITE with replaces */
 		ast_log(LOG_WARNING, "Re-invite to non-existing call leg on other UA. SIP dialog '%s'. Giving up.\n", p->callid);
@@ -12099,6 +12100,14 @@
 			ast_log(LOG_NOTICE, "Failed to authenticate on REGISTER to '%s' (tries '%d')\n", get_header(&p->initreq, "From"), p->authtries);
 			ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);	
 		}
+		break;
+	case 408:	/* Request timeout */
+		if (global_regattempts_max)
+			p->registry->regattempts = global_regattempts_max+1;
+		ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);	
+		r->call = NULL;
+		ast_sched_del(sched, r->timeout);
+		r->timeout = -1;
 		break;
 	case 479:	/* SER: Not able to process the URI - address is wrong in register*/
 		ast_log(LOG_WARNING, "Got error 479 on register to %s@%s, giving up (check config)\n", p->registry->username,p->registry->hostname);
@@ -12389,6 +12398,21 @@
 			} else	/* We can't handle this, giving up in a bad way */
 				ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);	
 
+			break;
+		case 408: /* Request timeout - terminate dialog */
+			if (sipmethod == SIP_INVITE)
+				handle_response_invite(p, resp, rest, req, seqno);
+			else if (sipmethod == SIP_REGISTER) 
+				res = handle_response_register(p, resp, rest, req, ignore, seqno);
+			else if (sipmethod == SIP_BYE) {
+				ast_set_flag(&p->flags[0], SIP_NEEDDESTROY); 
+				if (option_debug)
+					ast_log(LOG_DEBUG, "Got timeout on bye. Thanks for the answer. Now, kill this call\n");
+			} else {
+				if (owner)
+					ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
+				ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);	
+			}
 			break;
 		case 481: /* Call leg does not exist */
 			if (sipmethod == SIP_INVITE) {



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