[asterisk-users] T1 problem (call using a .call file)

[Danny Nicholas]

Try this call file - replace XXX with your number and YYY with a valid SIP
exten on your system

 

Channel: DAHDI/g1/1XXXXXXXXXX
Callerid:  SIP/YYY

MaxRetries: 1
RetryTime: 5
WaitTime: 60



 

  _____  

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Pascal Bruno
Sent: Thursday, March 19, 2009 9:22 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] T1 problem (call using a .call file)

 

Here is what my extensions.conf file has:

 

exten => _NXXNXXXXXX,1,Dial(DAHDI/g1/${EXTEN})
exten => _NXXNXXXXXX,n,Hangup()

 

exten => _1NXXNXXXXXX,1,Dial(DAHDI/g1/${EXTEN})
exten => _1NXXNXXXXXX,n,Hangup()

 

Using the phone, I can dial any numbers succesfully.

 

And here is my call file:

 

Channel: DAHDI/g1/1XXXXXXXXXX
Callerid: XXXXXXXXXX
MaxRetries: 1
RetryTime: 5
WaitTime: 60
Context: test
Extension: s
Priority: 1

 

with the call file I can dial my cellphone which begin with 754XXXXXXX

but when I call my friend's cellphone from new york which is 201XXXXXXX i
get progress code 127 as follows

 

-- Attempting call on DAHDI/g1/1201XXXXXXX for s at test:1 (Retry 1)
    -- Requested transfer capability: 0x00 - SPEECH
    -- PROGRESS with cause code 127 received

 

I tried with the prefix 1 and without the prefix 1 it is always the same
thing, but with the handset I dial my phone and my friend's phone
succesfully with and without the 1

 



 

On Thu, Mar 19, 2009 at 9:21 AM, Danny Nicholas <danny at debsinc.com> wrote:

Please paste the call file content (with the number XXXX'ed of course) and
the Dial section from extensions.conf.

 

  _____  

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Pascal Bruno
Sent: Wednesday, March 18, 2009 6:24 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] T1 problem (call using a .call file)

 

This has to be a bug, because I dont know what else to try here

 

 

On Wed, Mar 18, 2009 at 11:16 AM, Pascal Bruno <tipascal at gmail.com> wrote:

Nope, I always dial 1 + 10 digits for all my numbers.  It works on all
numbers when I am using my phone (Analogue or IP) but when I do it using a
.call file it does not work on some numbers mostly.  That is the weirdest
thing I have ever seen.  I tried different codecs in the call file, I still
get the PROGRESS with cause code 127

 

 



 

On Wed, Mar 18, 2009 at 10:47 AM, David Backeberg <dbackeberg at gmail.com>
wrote:

On Mon, Mar 16, 2009 at 6:42 PM, Pascal Bruno <tipascal at gmail.com> wrote:
> I have a weird problem with call using my T1 card.  I can make calls fine
> using my analog and IP phones, but when I try to initiate a call using a
> .call file, I get the following error
>  -- Attempting call on DAHDI/g1/1XXXXXXXXXX for s at test:1 (Retry 1)
>     -- Requested transfer capability: 0x00 - SPEECH
>     -- PROGRESS with cause code 127 received
> it happens on certain numbers I dial, but if I dial that same number with
an
> ip or analog phone that use the T1 channel, the call is going through
> normally.
> Anybody knows why?

Are you doing anything silly with prefixing or short-circuit dialing?

in other words..

You dial 8 for an outside line, then 1+10 digits
and you're forgetting to do that for some numbers?

_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com
<http://www.api-digital.com/>  --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

 

 


_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com
<http://www.api-digital.com/>  --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090319/869bbd70/attachment.htm