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

[Stuart Mackintosh]

show dialplan will show the asterisk view of the dialplan. 
show channels will display channels in use and 
sip debug will show what the sip phones are doing.

Also, have a console open as this often provides clues, especially if
started with some verboseness -vvvvvv

You may try making a more generic match for pstn, like _. to see if it
is your match that is causing the problem.

sm

On Sat, 2004-05-01 at 13:59, Tom Scott wrote:
> 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
> 
> 
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