How to configure Asterisk to answer SIP calls

I have an Asterisk 11.6.0 server configured, currently, with 16 analog lines (using Digium TDM800 cards). The 16 analog lines are all served as extensions from an Avaya analog telephone switch. This server is serving as an interactive voice response system, by answering the calls on the lines, taking touch-tone input from the caller, and using that information to look up database information and read scripted output back to the caller - very simple.

NOW, I am getting ready to replace the Avaya analog telephone switch, with an Avaya VoIP switch. And I need to configure this same Asterisk server application to perform the same function, but rather than answering analog lines, I would prefer it answer SIP calls, also served from the Avaya VoIP switch.

I can’t seem to find information that is pertinent to this by searching (for a couple of weeks now), or maybe I just don’t understand what I’m finding! At any rate, I need some help in configuring this scenario.

Can anyone help me?


Use the Biz and Jobs forum for paid consultancy, Most users work out how to use Asterisk with SIP very quickly.

Was hoping to not NEED paid consultancy! That’s why I was asking in the forum. Thanks.

If you don’t want to pay a consultant, you need to READ and LEARN how to use SIP on Asterisk. I suggest you start working. :wink:

Forum is not here to give free support. It is here to help out people who have done the reading and got stuck somewhere on the way.