When a peer registers using WebSocket do not resolve the Contact provided.

(closes issue ASTERISK-20238)
Reported by: james.mortensen
........

Merged revisions 371482 from http://svn.asterisk.org/svn/asterisk/branches/11

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371483 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index b42d42e..1c8abd1 100644 (file)
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -15305,8 +15305,9 @@ static enum parse_register_result parse_register_contact(struct sip_pvt *pvt, st
                ao2_t_unlink(peers_by_ip, peer, "ao2_unlink of peer from peers_by_ip table");
        }
 
-       if (!ast_test_flag(&peer->flags[0], SIP_NAT_FORCE_RPORT) && !ast_test_flag(&peer->flags[0], SIP_NAT_RPORT_PRESENT)) {
-                /* use the data provided in the Contact header for call routing */
+       if ((transport_type != SIP_TRANSPORT_WS) && (transport_type != SIP_TRANSPORT_WSS) &&
+           (!ast_test_flag(&peer->flags[0], SIP_NAT_FORCE_RPORT) && !ast_test_flag(&peer->flags[0], SIP_NAT_RPORT_PRESENT))) {
+               /* use the data provided in the Contact header for call routing */
                ast_debug(1, "Store REGISTER's Contact header for call routing.\n");
                /* XXX This could block for a long time XXX */
                /*! \todo Check NAPTR/SRV if we have not got a port in the URI */