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
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