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authorJoshua Colp <jcolp@digium.com>2012-08-17 19:50:58 +0000
committerJoshua Colp <jcolp@digium.com>2012-08-17 19:50:58 +0000
commit1a95c9a906e9548a04d91fe77d77216a93e30df9 (patch)
tree14096ed0814564fbd571cdb400b9aaeda82a8e72 /channels/chan_sip.c
parent064c7bd45676aa4c6cba4722a24831815523906b (diff)
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
Diffstat (limited to 'channels/chan_sip.c')
-rw-r--r--channels/chan_sip.c5
1 files changed, 3 insertions, 2 deletions
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index b42d42eff..1c8abd1e1 100644
--- 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 */