[SR-Users] Question about relaying

[Henning Westerholt]

Am Freitag, 3. Mai 2013, 16:54:29 schrieb Leo Brown:
> I added record_route() and now I see an extra record-route and Via: header:
> 
> .9........INVITE sip:44800800150 at 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
>     INVITE sip:44800800150 at our-pstn-switch SIP/2.0
>     Record-Route: <sip:mvno-edge;lr=on>
>     Via: SIP/2.0/UDP mvno-edge;branch=z9hG4bK388f.04bc8632.1
>     Via: SIP/2.0/UDP mvno-carrier:5060;rport=5060;branch=z9hG4bK82ae6ced
>     Contact: <sip:441234567890 at mvno-carrier:5060>
> 
> 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?

Hi Leo,

maybe the originating gateway user agent is not configured correctly. Normally 
one does not need to modify the Contact header for this functionality.

The originating user agent needs to "learn" the route set that he should 
follow and specify then this the respective Route headers.

Take a look to this example on how it works in detail:

http://www.in2eps.com/fo-sip/tk-fo-sip-dialog.html

BTW, if your proxy on the path also needs to honour Route sets you need to use 
also the loose_route function in rr module.

Henning Westerholt