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

Stuart Mackintosh sm at opusvl.com
Sat May 1 08:21:50 MST 2004


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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