[SR-Users] Problem with Route header

[Geoffrey Mina]

Greetings,
I am confused at some functionality I am seeing with Kamailio 1.5.4.  I
know this is an old version, but I don't have the time to go through a
lengthy upgrade process right now.  The issue I am seeing is that the
server is inserting a Route header with it's own IP address for an unknown
reason.  Here is the initial invite (removed SDP for simplicity):

INVITE sip:13 at 67.207.130.146:5060 SIP/2.0
Via: SIP/2.0/UDP 68.64.220.108:5060;branch=z9hG4bK78dd33c6;rport
From: "WIRELESS CALLER" <sip:9546496707 at dev-asterisk.mydomain.com
>;tag=as1cad6370
To: <sip:13 at 67.207.130.146:5060>
Contact: <sip:9546496707 at 68.64.220.108>
Call-ID: 43134ece101abfca6ecab20212295909 at dev-asterisk.mydomain.com
CSeq: 102 INVITE
User-Agent: G-Tel v1.0
Max-Forwards: 70
Remote-Party-ID: "WIRELESS CALLER" <sip:9546496707 at dev-asterisk.mydomain.com
>;privacy=off;screen=no
Date: Tue, 01 May 2012 18:17:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Route: <sip:13 at boulder-voip.mydomain.com>
P-Account-ID: 99990023
P-Proxy-Route: Yes
Content-Type: application/sdp
Content-Length: 240

The basics of what happen next are:

t_check_trans();
record_route();
remove_hf("P-Proxy-Route");
if(loose_route()){
   route(3);
}


route[3]{
        t_on_reply("1");
        if(!t_relay()){
                sl_reply_error();
        }
}

The INVITE that goes out has the funky Route: header with the Kamailio IP
in there.  This is causing problems for some of the upstream proxy servers
(obviously).

INVITE sip:13 at boulder-voip.mydomain.com SIP/2.0
Record-Route: <sip:67.207.130.146;lr;ftag=as1cad6370>
Via: SIP/2.0/UDP 67.207.130.146;branch=z9hG4bKf183.456d51e1.0
Via: SIP/2.0/UDP 68.64.220.108:5060
;received=68.64.220.108;branch=z9hG4bK78dd33c6;rport=5060
From: "WIRELESS CALLER" <sip:9546496707 at dev-asterisk.mydomain.com
>;tag=as1cad6370
To: <sip:13 at 67.207.130.146:5060>
Contact: <sip:9546496707 at 68.64.220.108>
Call-ID: 43134ece101abfca6ecab20212295909 at dev-asterisk.mydomain.com
CSeq: 102 INVITE
User-Agent: G-Tel v1.0
Max-Forwards: 69
Remote-Party-ID: "WIRELESS CALLER" <sip:9546496707 at dev-asterisk.mydomain.com
>;privacy=off;screen=no
Date: Tue, 01 May 2012 18:17:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
P-Account-ID: 99990023
Content-Type: application/sdp
Content-Length: 240
Route: <sip:13 at 67.207.130.146:5060>


Any idea what may be causing this to happen and how I could prevent it?  I
have tried removing the Route header using the remove_hf("Route") before
doing the t_relay, but that doesn't seem to help.

Thanks,
Geoff
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sip-router.org/pipermail/sr-users/attachments/20120501/ee33d9b0/attachment.htm>