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 > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users@lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- 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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