h323 and SIP trunk configurations for a SIP<>H323 gateway


I am using FreePBX that came with AsteriskNOW 1.5. I am trying build a gateway between an Avaya system running CM 3.1.2 running h323 and a video conference controller (MCU) that runs SIP.

I have searched both forums as well as Google and found Cyril’s and Gollum’s posts. They both have a lot of info but they didn’t seem to understand when they wrote them that we are not supposed to edit the extensions.conf and sip.conf files. Yet some respondants seem to be able to get it working. I cannot understand how.

For one thing, since the freepbx shell creates several layers of .conf files in each category I am not sure what context to identify in the custom files.

The h323 configs dont explain how to make *now aware of the existance of the h323.conf file and I don’t see any commands in the *now cli related to h323 like I do for SIP. I am not sure if this is a *now issue or freepbx but I thought I’d try posting that here. I dont see anywhere in the avaya interface for userid and password. The user id may be the far end node name but no password.

The SIP config examples do not use the FreePBX interface and do not appear to create SIP trunks, just SIP phones. When I use the FreePBX interface I think I am figuring it out but it is not clear how the routes relate to * configuration statements. I understand what routes are from my avaya background. They seem to translate to extensions that point to channel destinations but I am not sure of all of the implications, and again the contexts.

I have not found any doumentation on the freepbx interface yet so any pointers to that would be appreciated too.

Any help will be appreciated,

Thank you

Dallas TX, USA