the changed the include commands and they are now at least causing the extension to match using one of the local 10-digit numbers. this is what shows up on the console:
Executing StripMSD("SIP/1008-32df", "1") in new stack
-- Executing Dial("SIP/1008-32df", "Zap/1|BYEXTENSION") in new stack
-- Called 1
-- Hungup 'Zap/1-1'
-- Executing Congestion("SIP/1008-32df", "") in new stackshow dialplan looks alright, but show channels indicates: 0 active channels.
i'm guessing that that's a problem (0 channels). so i look in the zapata.conf file, which is the only place i remember doing anything with channels. this is the content of that file:
[channels] language=en context=default signalling=fxs_ks ! with only the FXO card, no FXS usecallerid=yes hidecallerid=no relaxdtmf=yes callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callerid=asreceived echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 immediate=no channel=1
the commands that are getting executed by the match in the extensions.conf file are:
exten => _9NXXXXXXXXXX,1,StripMSD,1 ! using 10-digits for CO switch exten => _NXXXXXXXXXX,2,Dial(Zap/1,BYEXTENSION) exten => _NXXXXXXXXXX,3,Congestion
when we hit the Dial command it appears to fail. that's when we get the "Hungup 'Zap/1-1'" error message on the console.
the SIP phones are still working. we can call from one sip phone to the other on the LAN, and we can call into the asterisk system from a PSTN phone and connect to the sip phone.
any ideas? is there something we have to do to activate a channel? is Dial(Zap/1,BYEXTENSION) command correct?
-- TT
Stuart Mackintosh wrote:
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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