[OpenSIPS-Users] In dialog requests misrouted

[Bogdan-Andrei Iancu]

Hi Bruce,

The ACK for a 200OK is routed based on the route set - this the RR set + 
the contact of the other party.    So, the ACK will have in RURI the 
contact of the other party (from 200 OK) and the RR set as Route hdrs.

Regards,
Bogdan

Bruce Borrett wrote:
> 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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-- 
Bogdan-Andrei Iancu
www.voice-system.ro