<div dir="ltr"><div><div><div><div><div><div>Hello Guys, <br><br><br></div>Im getting a strange situation here that i dont know how to deal<br><br></div>i have an enviroment where i have freeswitch receiving a call to billing and opensips doing the load_balance to the gateways.<br>
<br></div>When i send the call to the gateway i receive the reply without the record-route header, i try to put an asterisk server as gateway and this not happen in this scenario .<br><br></div><div>Below the invite that i send to the gateway<br>
<br>U <a href="http://10.1.69.1:5079">10.1.69.1:5079</a> -> <a href="http://10.255.2.31:5031">10.255.2.31:5031</a><br>INVITE <a href="http://sip:255755813256@10.1.69.1:5079">sip:255755813256@10.1.69.1:5079</a> SIP/2.0.<br>
Record-Route: <sip:10.1.69.1:5079;lr;ftag=HgcSt10Xa854e;did=9d2.723c6252>.<br>Via: SIP/2.0/UDP 10.1.69.1:5079;branch=z9hG4bKe98.72455346.0.<br>Via: SIP/2.0/UDP 10.1.69.1:5069;received=10.1.69.1;rport=5069;branch=z9hG4bKK5N8yU10cgage.<br>
Max-Forwards: 68.<br>From: "200214" <<a href="mailto:sip%3A200214@10.1.69.1">sip:200214@10.1.69.1</a>>;tag=HgcSt10Xa854e.<br>To: <<a href="http://sip:255755813256@10.1.69.1:5079">sip:255755813256@10.1.69.1:5079</a>>.<br>
Call-ID: 4c6591da-e483-1231-6cb4-001a4bd5a0b4.<br>CSeq: 53458861 INVITE.<br>Contact: <sip:gw+os@10.1.69.1:5069;transport=udp;gw=os>.<br>User-Agent: vBilling.<br>Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, NOTIFY.<br>
Supported: precondition, path, replaces.<br>Allow-Events: talk, hold, conference, refer.<br>Content-Type: application/sdp.<br>Content-Disposition: session.<br>Content-Length: 195.<br>X-FS-Support: update_display,send_info.<br>
Remote-Party-ID: "200214" <<a href="mailto:sip%3A200214@10.1.69.1">sip:200214@10.1.69.1</a>>;party=calling;screen=yes;privacy=off.<br><br></div><div><br>and below the 200 ok that i receive<br><br>U <a href="http://10.255.2.31:5031">10.255.2.31:5031</a> -> <a href="http://10.1.69.1:5079">10.1.69.1:5079</a><br>
SIP/2.0 200 OK.<br>Via: SIP/2.0/UDP 10.1.69.1:5079;branch=z9hG4bKe98.72455346.0;received=10.1.69.1.<br>Via: SIP/2.0/UDP 10.1.69.1:5069;received=10.1.69.1;rport=5069;branch=z9hG4bKK5N8yU10cgage.<br>To: <<a href="http://sip:255755813256@10.1.69.1:5079">sip:255755813256@10.1.69.1:5079</a>>;tag=12ab34cd.<br>
From: "200214" <<a href="mailto:sip%3A200214@10.1.69.1">sip:200214@10.1.69.1</a>>;tag=HgcSt10Xa854e.<br>CSeq: 53458861 INVITE.<br>Call-ID: 4c6591da-e483-1231-6cb4-001a4bd5a0b4.<br>Allow: INVITE, BYE, CANCEL, ACK, INFO, REGISTER.<br>
Supported:.<br>Allow-Events: telephone-event.<br>Contact: <sip:255755813256@10.1.69.1:5031;transport=udp>.<br>Content-Type: application/sdp.<br>Content-Length: 196.<br><br></div>when i send the call to this gateway the loose route did not execute, and i get error's on dialog because the dialog is not matched<br>
<br><br></div>how should i deal with a situation like this ?<br><br></div><br><div><div><div><div><br><br><br></div></div></div></div></div>