[Asterisk-Users] No dialtone in a E1

Marco Castillo mabcastillo at vanderkaaden.net
Fri Feb 11 08:58:39 MST 2005


Thank you Peter, how can I add the options to Dial to generate ringback???
do you have an example???
By the way, it is a PRI E1, with 30 bchannels and 1 dchannel. For a little
background, I'm intending to replace my actual PBX with Asterisk, and
everything is just working fine, until yesterday when I realized that when a
call was made from some external lines, this lines didn't receive a
dialtone. For this reason, I began to make some exhaustive test cases, and
began to make calls from distinct providers to my E1. In all this testing I
received a dialtone, except for a GSM cellular phone from a specific Telco.
I tested some others GSM cellulars from the same Telco, and got always the
same functionality, they didn't receive a dialtone. I think that if Asterisk
can generate a ringback, this is going to solve all my problems with this
little issue.
Thank you in advance Peter for your help.

Marco

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com]On Behalf Of Peter
Svensson
Sent: Thursday, February 10, 2005 6:10 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] No dialtone in a E1


On Thu, 10 Feb 2005, Marco Castillo wrote:

> Hi, I'm having a little problem when trying to make a call from asterisk.
I
> connect a SIP phone to asterisk, and in the asterisk box I have a TE110P
> card connected to a E1. When a SIP client makes a call through the E1, I
> received no dialtone in the SIP client.
> In the same manner, when somebody from the POTS network makes a call to a
> SIP client (through * and the E1) he doesn't receive the apropiate tone of
> call progress. Does anyone has some ideas about this?

Are you talking about an ISDN E1 or another form of E1?

On isdn dialtone is an optional feature of the specification and there are
many implementations of isdn. I think it is mandatory on EuroISDN. Since
asterisk normally generates the dialtone itself there should be little
nead for the dialtone from the pstn. We use the dialtone from the network
ourselves, but asterisk could provide it as well.

In band call progress is also a feature of the net on isdn. If the net
does not provide it you will have to do so yourself. Just add the proper
options to Dial to generate ringback and if the call fails you generate
the matching sound (Busy etc).

Peter

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