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

[Karsten Wemheuer]

Hi Bogdan-Andrei,

in case of global advertising is active and set to the natted address
the advertised address is used, but this leads to problems using phones
in the LAN.

As written in my other post: Without setting the advertise address and
port, I have a problem with the phones behind NAT. Is it possible to
manipulate the route before in a branch or something like that?

Regards,

Karsten

Am Donnerstag, dem 31.03.2022 um 18:53 +0300 schrieb Bogdan-Andrei
Iancu:
> Hi Karsten,
>
> You say the record_route() does not take into consideration the
> global
> advertising ??
>
> 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 6:22 PM, Karsten Wemheuer wrote:
> > 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
> >
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