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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