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 > > [email protected] > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list [email protected] http://lists.opensips.org/cgi-bin/mailman/listinfo/users
