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

[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
>
>
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