Hi We are trying to migrate from an SBC to Opensips 1.6. When we are sending calls to another provider who are using Openser, they are not taking the contact address from our 200 replies, instead they are putting our Openser address in the RURI of all Acks, Byes and Cancels. Am I right in saying that this is incorrect? Im not sure where they are getting this address either, maybe from the To: field, or from the record route header?
Is there a way to match the message received to a transaction and route the message to the contact in the original invite stored by TM? Or perhaps some better way of solving this? Here are the messages: SIP/2.0 200 OK Via: SIP/2.0/UDP xx.xxx.0.33;rport=5060;received=41.221.0.33;branch=z9hG4bKb9fc.f1cf4e03.0 Via: SIP/2.0/UDP xx.xxx.0.42:5060;branch=z9hG4bK3d3a5800;rport=5060 Record-Route: <sip:xx.xxx.1.13;lr=on;ftag=as5e3b3ce0> Record-Route: <sip:xx.xxx.0.33;lr=on;ftag=as5e3b3ce0> From: "xxxxxxx7239" <sip:[email protected]>;tag=as5e3b3ce0 To: <sip:[email protected]>;tag=as33b0f85f Call-ID: [email protected] CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.0.9 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: <sip:[email protected]> Content-Type: application/sdp Content-Length: 290 ACK sip:[email protected]:5060 SIP/2.0 Record-Route: <sip:xx.xxx.0.33;lr=on;ftag=as5e3b3ce0> Via: SIP/2.0/UDP xx.xxx.0.33;branch=z9hG4bKb9fc.f1cf4e03.2 Via: SIP/2.0/UDP xx.xxx.0.42:5060;branch=z9hG4bK5e0350ba;rport=5060 Route: <sip:xx.xxx.1.13;lr=on;ftag=as5e3b3ce0> From: "xxxxxxx7239" <sip:[email protected]>;tag=as5e3b3ce0 To: <sip:[email protected]>;tag=as33b0f85f Contact: <sip:[email protected]> Call-ID: [email protected] CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 69 Content-Length: 0 Thank you very much in advance.. Bruce
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