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