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

[brian k. west]

Just FYI stop using BYEXTENSION because it will be going away soon.

use ${EXTEN} or ${EXTEN:x}

bkw

----- Original Message ----- 
From: "Tom Scott" <telecomtom at vedatel.com>
To: <asterisk-users at lists.digium.com>
Sent: Saturday, May 01, 2004 12:29 PM
Subject: Re: [Asterisk-Users] dialing out to PSTN from SIP phones


> 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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