[OpenSIPS-Users] Dialplan/Routing

[Bogdan-Andrei Iancu]

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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