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

[Tom Scott]

okay, will use ${EXTEN}.

it all seems to be working now. I think my problem was
understanding the flow of control using contexts, but
i also needed to do some reading on syntax and variables
-- and more to come.

the working commands that we ended up using are:

[trunklocal]
exten => _9NXXXXXXXXX,1,StripMSD(1)
exten => _NXXXXXXXXX,2,Dial(${TRUNK}/${EXTEN})
exten => _NXXXXXXXXX,3,Congestion

thanks for the suggestions.

-- TT

brian k. west wrote:

>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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>>>      
>>>
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