Greetings,<br>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):<br>
<br>INVITE <a href="http://sip:13@67.207.130.146:5060">sip:13@67.207.130.146:5060</a> SIP/2.0<br>Via: SIP/2.0/UDP 68.64.220.108:5060;branch=z9hG4bK78dd33c6;rport<br>From: "WIRELESS CALLER" <<a href="mailto:sip%3A9546496707@dev-asterisk.mydomain.com">sip:9546496707@dev-asterisk.mydomain.com</a>>;tag=as1cad6370<br>
To: <<a href="http://sip:13@67.207.130.146:5060">sip:13@67.207.130.146:5060</a>><br>Contact: <<a href="mailto:sip%3A9546496707@68.64.220.108">sip:9546496707@68.64.220.108</a>><br>Call-ID: <a href="mailto:43134ece101abfca6ecab20212295909@dev-asterisk.mydomain.com">43134ece101abfca6ecab20212295909@dev-asterisk.mydomain.com</a><br>
CSeq: 102 INVITE<br>User-Agent: G-Tel v1.0<br>Max-Forwards: 70<br>Remote-Party-ID: "WIRELESS CALLER" <<a href="mailto:sip%3A9546496707@dev-asterisk.mydomain.com">sip:9546496707@dev-asterisk.mydomain.com</a>>;privacy=off;screen=no<br>
Date: Tue, 01 May 2012 18:17:50 GMT<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO<br>Supported: replaces<br>Route: <<a href="mailto:sip%3A13@boulder-voip.mydomain.com">sip:13@boulder-voip.mydomain.com</a>><br>
P-Account-ID: 99990023<br>P-Proxy-Route: Yes<br>Content-Type: application/sdp<br>Content-Length: 240<br><br>The basics of what happen next are:<br><br>t_check_trans();<br>record_route();<br>remove_hf("P-Proxy-Route");<br>
if(loose_route()){<br> route(3);<br>}<br><br><br>route[3]{<br> t_on_reply("1");<br> if(!t_relay()){<br> sl_reply_error();<br> }<br>}<br><br>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). <br>
<br>INVITE <a href="mailto:sip%3A13@boulder-voip.mydomain.com">sip:13@boulder-voip.mydomain.com</a> SIP/2.0<br>Record-Route: <sip:67.207.130.146;lr;ftag=as1cad6370><br>Via: SIP/2.0/UDP 67.207.130.146;branch=z9hG4bKf183.456d51e1.0<br>
Via: SIP/2.0/UDP 68.64.220.108:5060;received=68.64.220.108;branch=z9hG4bK78dd33c6;rport=5060<br>From: "WIRELESS CALLER" <<a href="mailto:sip%3A9546496707@dev-asterisk.mydomain.com">sip:9546496707@dev-asterisk.mydomain.com</a>>;tag=as1cad6370<br>
To: <<a href="http://sip:13@67.207.130.146:5060">sip:13@67.207.130.146:5060</a>><br>Contact: <<a href="mailto:sip%3A9546496707@68.64.220.108">sip:9546496707@68.64.220.108</a>><br>Call-ID: <a href="mailto:43134ece101abfca6ecab20212295909@dev-asterisk.mydomain.com">43134ece101abfca6ecab20212295909@dev-asterisk.mydomain.com</a><br>
CSeq: 102 INVITE<br>User-Agent: G-Tel v1.0<br>Max-Forwards: 69<br>Remote-Party-ID: "WIRELESS CALLER" <<a href="mailto:sip%3A9546496707@dev-asterisk.mydomain.com">sip:9546496707@dev-asterisk.mydomain.com</a>>;privacy=off;screen=no<br>
Date: Tue, 01 May 2012 18:17:50 GMT<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO<br>Supported: replaces<br>P-Account-ID: 99990023<br>Content-Type: application/sdp<br>Content-Length: 240<br>Route: <<a href="http://sip:13@67.207.130.146:5060">sip:13@67.207.130.146:5060</a>><br>
<br><br>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.<br>
<br>Thanks,<br>Geoff<br><br><br>