[Asterisk-Users] dialing out to PSTN from SIP phones

[Tom Scott]

I installed Asterisk and a digium wildcard (X100P). Using
the extensions.conf with a few changes and a sip.conf file
that includes two extensions, I can place calls between the
SIP phones. I also can call in to the SIP phones from the
PSTN using the X100P. On incoming calls I can hear the
default demo announcement and call the digium IAX line.

The main problem i'm having is calling out to the PSTN from
the SIP phones. We have a 10-digit dialing pattern for local
calls, which matches _9NXXXXXXXXX in the extensions.conf
I also strip the 9 with the StripMSD command. But I still
can't get the SIP phones to dial out. I get the error 404
(Not Found) indication on the Grandstream display

Does anyone know if there is there a way that I can display
on the console the lines that are being executed in the .conf
files so I can maybe find where my mistake is? Or does anyone
know of a common mistake that I could look?

-- TIA, TT