Hi Nitesh,
The "420 Bad Extension" is generated by the residential cfg when the
lookup on the caller fails (the caller party is not found as registered
in OpenSIPS).
Now, I assume you are dialing kind of DID (to be routed to PSTN), so it
should NOT hit the lookup (which is when calling to local subscribers).
So you may dial something wrong. As per default residential cfg, the
dialed number must start with `+` in order to be considered a PSTN
destination.
Best regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
https://www.opensips-solutions.com
OpenSIPS Summit 27-30 Sept 2022, Athens
https://www.opensips.org/events/Summit-2022Athens/
On 9/27/22 6:09 PM, Nitesh Divecha wrote:
Hello All,
I'm a newbie with Opensips! Got good knowledge with Asterisk and SIP
in general.
Trying to figure out how to route calls out on the SIP trunk.
Running following:
"Server": "OpenSIPS (3.3.1 (x86_64/linux))"
OpenSIPS Control Panel 9.3.2
Debian 11
Opensips is configured with residential configuration and I can make
the following:
1) local SIP to SIP calls (registered SIP endpoints).
2) External DID to Opensips to local SIP endpoint.
But failing to call out from the local SIP endpoint to SIP trunk
(external). Every time I make a call I get SIP 420 Bad Extension.
I did follow all the instructions regarding Opensips-CP from
(https://powerpbx.org/content/opensips-v30-debian-v10-mariadb-apache-v1
<https://powerpbx.org/content/opensips-v30-debian-v10-mariadb-apache-v1>)
to setup SIP trunk, dial plan, dynamic routing and
edit "opensips_residential.cfg" but failing to send the call out.
Any suggestions?
Thanking in advance.
Cheers,
Nite
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