Trouble to Integrate Asterisk and Avaya Definity

Hi everyone,

We´re trying to integrate our avaya definity g3si and asterisk using an E1 card with r2 signalling. At this moment I can make calls from the asterisk to pbx with no problem. When i try to generate call from the pbx to asterisk, the call rings two times and the is disconnected.

The codec i´m using both in pbx and softfone is alaw. Does anyone could give me a hint ?



A strategy to connect an asterisk and a propriarity pbx is setting the Asteriskbox in front of the propriarity pbx .

I assume that you have a isdn30/24 connection (europian/rest of the world).

You need a Dual E1 card (Sangoma, Digium) in the asterisk server so you can move the E1 of the provider to your asterisk box. The other E1 connection plays provider for the propriarity PBX and with an isdn cross cable you connect the asterisk server with the propriarity pbx

isnd connection (inbound/outbound) <-> (et mode) asterisk box (nt mode) <-> isdn crosscable <-> (et mode) propriarity pbx

You need some smart things in your dial plan especially when you have internal phones on your asterisk box and internal phones on your propriarity pbx that must be able to call each other but basically this is it.

This way you don’t have to spend money on extra ports on your propriarity pbx wich is most of the times very expensive.

Hi lesouvage, thanks for your reply.

Our main goal is really to avoid spending lot of $$ with propriarity pbx. There are lots of facilities we can deploy using Asterisk paying no bucks.

We have already acquired that E1 Card (it´s a digivoice) and runs using R2_MFC signalling standard. ISDN is not so widely spread in Brazil as you can see ISDN in other countries.

As I wrote, I can make calls from asterisk >> pbx but not from pbx to asterisk. The call is being dropped after two rings. That´s why I´m asking for help here.

What card are you connecting the E1 card from Asterisk to on the Avaya?

A sample of your config in Asterisk can also help in helping you.


We used the Sangoma 2 poort E1 card with one port in NT mode and the other in TE mode for Euro isdn. This way you have 30 channels available (+ one signaling channel) between the Asterisk and the other pbx.

With two ports you can also test locally by connecting the two ports on the card with an isdn cross cable. This way you can test it like the TE (rminiation) port (the port that receives the calls from the telephony provider) is on another pbx and the nt port (acting like telefony provider) is on the Asterisk server.

Info about configuring the Sangoma card for Asterisk is on the Sangoma site ( You can also choose for Digium and one of there E1 cards

The question is: is it possible to set the E1 on the current PBX in EURO ISDN mode. If you can without spending a lot of money this might be an interesting option.


You need to supply some debug info. IE a cli trace of a call that cuys off and the pri debug so we can see the cause code.

Just saying it doenst work means nothing, we need to whats happening to be able to help.

If you can call from * to the G3 then the link is OK, If you can call from th eG3 to the * and they cut off, it could be as simple as a context issue but who knows without seeing and information


Hi everyone, tks for the replies.

I was able to solve the issue. The problem was related to PBX that was not accepting to establish the call using only the TAC (trunk access code). I need to add a route-pattern and forward all calls to Asterisk extension range to integration trunk.

Worked !