[Asterisk-Users] No dialtone in a E1

[Peter Svensson]

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