[SR-Users] Rewriting header of BYE message to carrier

[Daniel-Constantin Mierla]

Hello,

changing the R-URI (sip address in the first line of request) can be
done with varables:

  - $ru - the entire r-uri
  - $rd - only the domain part of r-uri

Cheers,
Daniel

On 14/01/16 23:25, Ryan Mottley wrote:
> Hi,
>
> We're running a system with Kamailio running in front of Asterisk just
> handling registrations and forwarding everything else to Asterisk. But
> we're having an issue during hangup on incoming calls. If the
> initiator hangs up, the call completes successfully. But if one of our
> phones hangs up, the BYE message comes back with a 404 "Not Found" and
> the call doesn't hang up on the carrier side.
>
> According to the carrier, it's because the IP in the contact on our
> ACK message goes to their audio IP while the header of our BYE points
> to their signaling IP. 
>
> ACK sip:[Kamailio Pub
> IP]:5060;line=sr--rkpsDAp6YAp6DIpZDZmZeI2ZYI26YIRVDcpsDIpsem* SIP/2.0
> Via: SIP/2.0/UDP *[Carrier Signaling IP]*;branch=z9hG4bK2236.1402e7b4.2
> Via: SIP/2.0/UDP *[Carrier Audio IP]*;received=*[Carrier Audio
> IP]*;branch=z9hG4bK07a8bccb;rport=5060
> Route: <sip:[Kamailio Pub
> IP];r2=on;lr=on;ftag=as67cef00d;nat=yes>,<sip:10.120.0.1;line=sr--rkpsDthVDIhVDNh6Ogo6eKh6eAQs4LRflC2srQRflC2srGqAl-CAP6rZrZkGDmpGed2APtCvlx1>
> From: "+16014477389" <sip:6014477389@*[Carrier Audio IP]*>;tag=as67cef00d
> To: <sip:6016025063@*[Carrier Signaling IP]*>;tag=as643b40ca
> Contact: <sip:6014477389@*[Carrier Audio IP]*>
> Call-ID: 4aaefec90826a2a221f0af9500ad211b@*[Carrier Audio IP]*
> CSeq: 102 ACK
> User-Agent: packetrino
> Max-Forwards: 69
> Content-Length: 0
>
> BYE sip:6014477389@*[Carrier Signaling IP] *SIP/2.0
> Via: SIP/2.0/UDP [Kamailio Pub
> IP];branch=z9hG4bK2236.fca983a45913fb510f97e781a85c7392.0
> Via: SIP/2.0/UDP
> 10.120.0.1;branch=z9hG4bKsr-IqktV1L26BCx0jmwZeI2ZYI26YIRVDcpsDIpsem.-EF8-EtCZYmg6edIMehhA4A.AEzyuiZKfPKo7N-qAcWq6D-rsYc4Zp**
> Route: <sip:*[Carrier Signaling IP]*;lr=on>
> Max-Forwards: 69
> From: <sip:6016025063@[Kamailio Pub IP]>;tag=as643b40ca
> To: "+16014477389" <sip:6014477389@*[Carrier Audio IP]*>;tag=as67cef00d
> Call-ID: 4aaefec90826a2a221f0af9500ad211b@*[Carrier Audio IP]*
> CSeq: 102 BYE
> User-Agent: Asterisk PBX 13.6.0
> X-Asterisk-HangupCause: Normal Clearing
> X-Asterisk-HangupCauseCode: 16
> Content-Length: 0
>
> I'm thinking it's happening because their side isn't configured
> correctly to handle traffic coming back from a proxy, but in the
> meantime is there a way to rewrite the top of the BYE header to match
> the "audio IP" they're requesting it be sent to?
>
> Thanks!
>
> -- 
> Ryan Mottley, Developer
> VOXO, LLC
> voxo.co <http://voxo.co> - (601)602-5063
>
>
> _______________________________________________
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-- 
Daniel-Constantin Mierla
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Book: SIP Routing With Kamailio - http://www.asipto.com
http://miconda.eu

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