Hi, Alex:

After taking your suggestion change from e&m to fxoks, it still did not work, and this time even calling to normal PSTN number also failed?

Any more suggestion?

Charles

On Wed, 30 Nov 2005, Charles Huang wrote:

> span=1,0,0,esf,b8zs e&m=1-24
>
> and in my /etc/asterisk/zapata.conf file, I set the following
>
> signalling=em group=1 channel=>1-24
>
> Can anyone give me any suggestion, this is really strange, how come some
> will succeed and some will fail.
a) this is not a -dev question!

b) have you asked whether the signaling is really e&m, or, more likely, it
is fxo?

-alex

Message: 1
Date: Wed, 30 Nov 2005 21:19:21 -0800
From: Charles Huang <[EMAIL PROTECTED]>
Subject: [Asterisk-Dev] New Jersey ATT Vocie T1 Asterisk Toll free
       number  not working
To: [email protected]
Message-ID:
       <[EMAIL PROTECTED]>
Content-Type: text/plain; charset="iso-8859-1"

Hi, All:

I set up an Asterisk machine in our New Jersey Office which is using ATT
Voice T1, I am facing one very strange problems.

I can dial to any of the Normal PSTN numbers (including Cell phone numbers),
but for Toll Free numbers, some of them are working, and some of them are
not working.

For example, the following Toll Free numbers are working and can dial from
my SIP phone:

1-800-315-9339
1-866-562-6506
1-877-568-8746

but these numbers when I dial, I will only hear the rings when not set
progess=yes option; if set the progess on zapata.conf, then I can hear from
CO that this was fail.

1-888-848-4792

1-866-280-6429

1-866-825-5460

1-800-361-5659


 In my /etc/zaptel.conf, I set the following

span=1,0,0,esf,b8zs
e&m=1-24

and in my /etc/asterisk/zapata.conf file, I set the following

signalling=em
group=1
channel=>1-24


Can anyone give me any suggestion, this is really strange, how come some
will succeed and some will fail.


Thanks a lot!

Charles
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --

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

Reply via email to