[OpenSIPS-Users] In dialog requests misrouted

[Bruce Borrett]

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:xxxxxxx7239 at xx.xxx.0.42>;tag=as5e3b3ce0
To: <sip:xxxxxx0114 at xx.xxx.1.13>;tag=as33b0f85f
Call-ID: 4b3419c86b29e5ce5a387a1b74f7effc at xx.xxx.0.42
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:xxxxxxx0114 at xx.xxx.236.105>
Content-Type: application/sdp
Content-Length: 290


ACK sip:xxxxxx0114 at xx.xxx.1.13:5060 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:xxxxxxx7239 at xx.xxx.0.42>;tag=as5e3b3ce0
To: <sip:xxxxxx0114 at xx.xxx.1.13>;tag=as33b0f85f
Contact: <sip:xxxxxx7239 at xx.xxx.0.42>
Call-ID: 4b3419c86b29e5ce5a387a1b74f7effc at xx.xxx.0.42
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 69
Content-Length: 0

Thank you very much in advance..
Bruce


      
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