Oh, so you have traffic both from public and private network, right ? If
so, you have 2 options:
1) use a single socket, without advertise and use the script advertise
function depending on the source of the call - see
set_advertised_address() [1]
2) use 2 sockets, one for public traffic, with advertise and one for the
internal traffic, without advertise. For each call, control the outbound
interface via the $socket_out variable [2]
My 2 cents, (1) may be too complicated as the set_advertised_address()
function as a message level scope, so you need to take care and do it
for each and every single request of the call. With (2) you need to do
it only for the initial request (INVITE) as the sockets are remembered
for the whole duration of the call by the Record Route mechanism.
[1]
https://www.opensips.org/Documentation/Script-CoreFunctions-3-2#set_advertised_address
[2] https://opensips.org/Documentation/Script-CoreVar-3-2#socket_out
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 7:27 PM, Karsten Wemheuer wrote:
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
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