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

Bogdan-Andrei Iancu bogdan at opensips.org
Thu Mar 31 14:44:20 UTC 2022

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
OpenSIPS eBootcamp 23rd May - 3rd June 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

More information about the Users mailing list