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

[Tom Scott]

thanks for responding.

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 stack

show 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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