automerge commit

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2-netsec@18297 65c4cc65-6c06-0410-ace0-fbb531ad65f3

channels/chan_sip.c

@@ -11658,12 +11658,13 @@ static struct ast_channel *sip_request_call(const char *type, int format, void *
 	format &= ((AST_FORMAT_MAX_AUDIO << 1) - 1);
 	if (!format) {
 		ast_log(LOG_NOTICE, "Asked to get a channel of unsupported format %s while capability is %s\n", ast_getformatname(oldformat), ast_getformatname(global_capability));
+		*cause = AST_CAUSE_BEARERCAPABILITY_NOTAVAIL;	/* Can't find codec to connect to host */
 		return NULL;
 	}
 	p = sip_alloc(NULL, NULL, 0, SIP_INVITE);
 	if (!p) {
 		ast_log(LOG_ERROR, "Unable to build sip pvt data for '%s' (Out of memory)\n", (char *)data);
-		*cause = AST_CAUSE_CONGESTION;
+		*cause = AST_CAUSE_SWITCH_CONGESTION;
 		return NULL;
 	}
 
@@ -11671,7 +11672,7 @@ static struct ast_channel *sip_request_call(const char *type, int format, void *
 	if (!p->options) {
 		sip_destroy(p);
 		ast_log(LOG_ERROR, "Unable to build option SIP data structure - Out of memory\n");
-		*cause = AST_CAUSE_CONGESTION;
+		*cause = AST_CAUSE_SWITCH_CONGESTION;
 		return NULL;
 	}