[Asterisk-Users] Newbie can't dial out to pstn

Wiley Siler wsiler at education2020.com
Fri Mar 18 07:42:31 MST 2005


What version of Asterisk?  If this is not Asterisk at home you may want to install it and start over.  It eases many of the problems experienced by newbs when learning *.  

Otherwise, make sure you use the ztcfg -vvvv so you can see some error verbosity.

You may need to recompile your zaptel stuff.  Just make sure you follow the instructions and recompile asterisk after.

Regards,
Wliey 

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Greg
Sent: Thursday, March 17, 2005 10:12 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: Shane Dalgleish
Subject: Re: [Asterisk-Users] Newbie can't dial out to pstn

I have just run ztcfg and got these errors:

# ztcfg
Notice: Configuration file is /etc/zaptel.conf line 209: Cannot get number of tones chanel 1 line 209: Cannot init tones chanel 1 line 209: Cannot get number of tones chanel 2 line 209: Cannot init tones chanel 2 line 209: Cannot get number of tones chanel 3 line 209: Cannot init tones chanel 3 line 209: Cannot get number of tones chanel 4 line 209: Cannot init tones chanel 4

What would these mean. I searched the archives and couldn't find these errors.

Greg

On 18/03/2005, at 1:24 PM, Greg wrote:

> I was just copy an example from somewhere. I made the change but the 
> mobile still doesn't ring. The line the card is attached to works 
> fine. here is the new output
>
> Executing Goto("SIP/2002-4385", "mobile|0400039953|1") in new stack
>     -- Goto (mobile,0400039953,1)
>     -- Executing Goto("SIP/2002-4385", "localcall|0400039953|1") in 
> new stack
>     -- Goto (localcall,0400039953,1)
>     -- Executing Dial("SIP/2002-4385", "ZAP/1/0400039953|60|r") in new 
> stack
>     -- Called 1/0400039953
>     -- Zap/1-1 answered SIP/2002-4385
>     -- Hungup 'Zap/1-1'
>   == Spawn extension (localcall, 0400039953, 1) exited non-zero on 
> 'SIP/2002-4385'
>
> is this line -- Zap/1-1 answered SIP/2002-4385 displayed when the card 
> tries to make the call or when the card thinks it has established the 
> call?
>
> Regards,
> Greg
>
> By the way, I'm on the Gold Coast.
>
> On 18/03/2005, at 12:32 PM, Shane Dalgleish wrote:
>
>> Greg,
>>
>> Any reason why you are putting the country code on the front for a 
>> mobile call through pstn?
>> (Unless you have something like an Ericsson F220M Fixed Cellular 
>> Terminal connected to it?)
>>
>> And you said the tdm400p never tries to pick up the phone?
>> Have you connected a normal phone on the line and had a listen?
>>
>>
>> Where is Aus are you? :o)
>>
>> Cheers
>> Shane
>>
>>> -----Original Message-----
>>> From: asterisk-users-bounces at lists.digium.com
>>> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Greg
>>> Sent: Friday, 18 March 2005 1:01 PM
>>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>>> Subject: [Asterisk-Users] Newbie can't dial out to pstn
>>>
>>> Hi,
>>> I have just put in a tdm400p with 4 fxo modules and am trying to 
>>> dial out from x-lite to dial my mobile phone just to test.
>>>
>>> The output in the asterisk console is like this
>>>
>>> Executing Goto("SIP/2002-239b", "mobile|61400039953|1") in new stack
>>>      -- Goto (mobile,61400039953,1)
>>>      -- Executing Goto("SIP/2002-239b",
>>> "localcall|61400039953|1") in new stack
>>>      -- Goto (localcall,61400039953,1)
>>>      -- Executing Dial("SIP/2002-239b",
>>> "ZAP/1/61400039953|60|r") in new stack
>>>      -- Called 1/61400039953
>>>      -- Zap/1-1 answered SIP/2002-239b
>>>      -- Hungup 'Zap/1-1'
>>>    == Spawn extension (localcall, 61400039953, 1) exited non-zero on 
>>> 'SIP/2002-239b'
>>>
>>> It never tries to pick up the phone and dial out. I'm not sure if 
>>> the config is correct, but I can easily dial between x-lite clients, 
>>> just not get the pstn.
>>>
>>> Can anyone see any glaring mistakes?
>>>
>>> Any help is grealty appreciated.
>>>
>>> Regards,
>>> Greg
>>>
>>> My extensions.conf part is this:
>>>
>>> exten => _04XXXXXXXX,1,GoTo(mobile,61${EXTEN:1},1)
>>>
>>> [localcall] ; local calls by PSTN ?is a fixed charge, voip is per 
>>> minute exten => _X.,1,Dial(ZAP/1/${EXTEN},60,r) exten => 
>>> _X.,2,Congestion exten => _X.,3,Hangup exten => _X.,103,Hangup exten 
>>> => _X.,104,Hangup exten => _X.,105,Hangup
>>>
>>> [mobile] ; Maybe be cheaper to route mobile calls differently to STD 
>>> in some cases exten => _X.,1,Goto(localcall,${EXTEN},1)
>>>
>>> zaptel.conf
>>> fxsks=1-4
>>> loadzone=au
>>> defaultzone=au
>>> channels=1-4
>>>
>>> zapata.conf
>>> [channels]
>>>  
>>> busydetect=1
>>> busycount=7
>>>  
>>> relaxdtmf=yes
>>> callwaiting=yes
>>> callwaitingcallerid=yes
>>> threewaycalling=yes
>>> transfer=yes
>>> cancallforward=yes
>>>  
>>> usecallerid=yes
>>>  
>>> echocancel=yes
>>> echocancelwhenbridged=yes
>>>  
>>> rxgain=0.0
>>> txgain=0.0
>>>  
>>> group=1
>>> pickupgroup=1-4
>>>  
>>> immediate=no
>>>  
>>> context=incomingcall
>>>  
>>> signalling=fxs_ks
>>> callerid=asreceived
>>> channel=1-4
>>>
>>> _______________________________________________
>>> Asterisk-Users mailing list
>>> Asterisk-Users at 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
>>>
>>
>>
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at 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
>

_______________________________________________
Asterisk-Users mailing list
Asterisk-Users at 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



More information about the asterisk-users mailing list