Only wipe the redirected audio & video IP/port if it's specified, and trigger a reinvite.

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@41768 65c4cc65-6c06-0410-ace0-fbb531ad65f3

channels/chan_sip.c

@@ -12937,14 +12937,18 @@ static int sip_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, struc
 	if (!p) 
 		return -1;
 	ast_mutex_lock(&p->lock);
-	if (rtp)
+	if (rtp) {
 		changed |= ast_rtp_get_peer(rtp, &p->redirip);
-	else
+	} else if (p->redirip.sin_addr.s_addr || ntohs(p->redirip.sin_port) != 0) {
 		memset(&p->redirip, 0, sizeof(p->redirip));
-	if (vrtp)
+		changed = 1;
+	}
+	if (vrtp) {
 		changed |= ast_rtp_get_peer(vrtp, &p->vredirip);
-	else
+	} else if (p->vredirip.sin_addr.s_addr || ntohs(p->vredirip.sin_port) != 0) {
 		memset(&p->vredirip, 0, sizeof(p->vredirip));
+		changed = 1;
+	}
 	if (codecs && (p->redircodecs != codecs)) {
 		p->redircodecs = codecs;
 		changed = 1;