[OpenSIPS-Users] Call forking, branches, Record-routing

[Karsten Wemheuer]

Hi Bogdan-Andrei,

many thanks for Your help.

I tried with record_route. It doesn't work for me, as I set
"advertised_address" and "advertised_port" to the natted address of the
(only) interface. I wasn't able to avoid this. It seemed to be required
to be able to reflect the path "phone -> proxy -> pbx".

I removed the "advertised"-stuff and checked again the call with
record_route. Now this seems to work.

I think, I have to fix the other call flow to avoid the global setting
of the advertised address and port.

Best regards,

Karsten

Am Donnerstag, dem 31.03.2022 um 17:44 +0300 schrieb Bogdan-Andrei
Iancu:
> Hi Karsten,
>
> See my prev email, just to record_route() before the t_relay() for
> the
> initial INVITE. And the loose_route() stuff for whatever
> sequential/in-dialog requests.
>
> Best regards,
>
> Bogdan-Andrei Iancu
>
> OpenSIPS Founder and Developer
>    https://www.opensips-solutions.com
> OpenSIPS eBootcamp 23rd May - 3rd June 2022
>    https://opensips.org/training/OpenSIPS_eBootcamp_2022/
>
> On 3/31/22 2:50 PM, Karsten Wemheuer wrote:
> > Hi*,
> >
> > I have a understanding problem regarding branches and call forking.
> > A call from a PBX is to be routed to phone(s) via OpenSIPS. The
> > phones
> > are registered to OpenSIPs.
> >
> > INVITE --> lookup ----> 1. Destination
> >                     |
> >                     \--> 2. Destination
> >
> > When the call is terminated by the caller, the BYE request shall
> > take
> > the same path. Currently, the BYE is sent from the PBX directly to
> > the
> > Contact URI (which is not reachable by the PBX).
> >
> > Is it possible to use record_route in the branch_route so that
> > different record route headers are used? Or is there another way?
> >
> > Thanks in advance,
> >
> > Karsten
> >
> >
> > _______________________________________________
> > Users mailing list
> > Users at lists.opensips.org
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users