.9........INVITE sip:44800800150@pstn-out.netfuse.net SIP/2.0
Record-Route: <sip:85.13.242.55;lr=on>
Via: SIP/2.0/UDP 85.13.242.55;branch=z9hG4bK388f.04bc8632.1
Via: SIP/2.0/UDP 81.88.163.210:5060;rport=5060;branch=z9hG4bK82ae6ced
Via: SIP/2.0/UDP mvno-edge;branch=z9hG4bK388f.04bc8632.1
Via: SIP/2.0/UDP mvno-carrier:5060;rport=5060;branch=z9hG4bK82ae6ced
I have replaced the relevant IP addresses in the example with mvno-edge, mvno-carrier, and outbound-carrier. So the route got "recorded" but the Contact: still referenced my mvno-carrier when inviting my outbound-carrier.
Accordingly, I do not get the BYE message from my originating mvno-carrier, after I send them 200 OK they try to talk to my outbound-carrier.
Note this is how I am routing the call to my gateway:
# Change destination URI to our carrier
$ru = "sip:" + $rU + "@" + $sel(cfg_get.gateways.outbound_carrier_1);
Any other ideas on how the Contact header should be modified?
Cheers
Leo
if you want to ensure that your kamailio stays on the path of the dialog for
following requests you probably want to use record-route headers for this.
This is usally done with the rr module, record_route() function.