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
Stuart Mackintosh
2004-May-01 08:21 UTC
[Asterisk-Users] dialing out to PSTN from SIP phones
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 > > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users