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<font face="monospace">Hi Nitesh,<br>
<br>
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).<br>
<br>
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.<br>
<br>
Best regards,<br>
</font>
<pre class="moz-signature" cols="72">Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
<a class="moz-txt-link-freetext" href="https://www.opensips-solutions.com">https://www.opensips-solutions.com</a>
OpenSIPS Summit 27-30 Sept 2022, Athens
<a class="moz-txt-link-freetext" href="https://www.opensips.org/events/Summit-2022Athens/">https://www.opensips.org/events/Summit-2022Athens/</a></pre>
<div class="moz-cite-prefix">On 9/27/22 6:09 PM, Nitesh Divecha
wrote:<br>
</div>
<blockquote type="cite"
cite="mid:CAJbqcDOhdLLYGNZZsXdMxAYVEQf-iizf=HEGBVrjYMYjAUfO7A@mail.gmail.com">
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<div dir="ltr">Hello All,
<div><br>
</div>
<div>I'm a newbie with Opensips! Got good knowledge with
Asterisk and SIP in general. </div>
<div><br>
</div>
<div>Trying to figure out how to route calls out on the SIP
trunk. </div>
<div><br>
</div>
<div>Running following: </div>
<div>"Server": "OpenSIPS (3.3.1 (x86_64/linux))"</div>
<div>OpenSIPS Control Panel 9.3.2</div>
<div>Debian 11</div>
<div><br>
</div>
<div>Opensips is configured with residential configuration and I
can make the following: </div>
<div>1) local SIP to SIP calls (registered SIP endpoints).</div>
<div>2) External DID to Opensips to local SIP endpoint. </div>
<div><br>
</div>
<div>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. </div>
<div><br>
</div>
<div>I did follow all the instructions regarding Opensips-CP
from (<a
href="https://powerpbx.org/content/opensips-v30-debian-v10-mariadb-apache-v1"
moz-do-not-send="true">https://powerpbx.org/content/opensips-v30-debian-v10-mariadb-apache-v1</a>)
to setup SIP trunk, dial plan, dynamic routing and
edit "opensips_residential.cfg" but failing to send the call
out.</div>
<div><br>
</div>
<div>Any suggestions?</div>
<div><br>
</div>
<div>Thanking in advance. </div>
<div><br>
</div>
<div>Cheers, </div>
<div>Nite</div>
<div><br>
</div>
<div><br>
</div>
<div><br>
</div>
<div><br>
</div>
</div>
<br>
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</pre>
</blockquote>
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