How to set up a remote agent?

[pspenning]

Good Day All,
I am not even sure I can do this but here is what I want to do just in case someone has any ideas...

My goal is to keep calls in one single T-Server but pass them between agents on two different switch types.

Switch one is an Asterisk SIP Server
Switch two is an Avaya S8710

Calls are coming in via SIP T-Server from the Asterisk.  The calls get routed to a SIP Agent using Genesys Agent Desktop.  I would like to have other agents with Avaya extensions log in to the SIP T-Server via GAD but have the call sent to an Avaya extension via h.323 either through DMX or back through the Asterisk and then to the Avaya.

Ideally, I was thinking that when the agent on the Avaya logs in to GAD they would establish an audio patch via their extension and keep it nailed up all day so they can take advantage of features like "Auto-Answer".

In theory, this could be done from a home type phone also.  I just don't know how to get the system to call the "Non-SIP" agents phone to establish the audio path.

We have an Avaya SES server but it is version 3.1.1 and I have pegged some one way audio problems that have been resolved with later releases.  Rather than spend $$$ upgrading the SES, I figured I would use the reliable h.323 which is built in.

So in short, I am looking to have the "Remote" agent log in to the SIP Server via GAD and have audio calls transcoded and sent to an extension.  In this case the extension sits on an Avaya.

Any thoughts on if I can even do this?

Thanks for all ideas!
Perry

[René]

Hi Perry,

I think that both ways you described are doable but of course, I could be wrong

Using Asterisk as gateway between Avaya and SIP Server - Asterisk should behave as H.323/SIP proxy. Unfortunately, I don't know how to configure Asterisk to acts as proxy. But DN has to be provisioned on Asterisk and monitored by SIP Server using SUBSCRIBE/NOTIFY mechanism. Please check "Framework 7.6 SIP Server Integration Reference Manual" for details how to configure SIP Server with Asterisk and for supported call flows. However, most of work has to be done on Asterisk side.

Using DMX - That configuration could be a bit tricky but it should be possible either. I've never done something like that but (I think) you would need to configure VoIP trunk with some prefix (e.g.89) that will point to DMX. Then create DN (number matches extension of Avaya) and specify "contact" attribute (..AvayaExtension..@AvayaH323gateway) on Annex tab. Incoming call from Asterisk will be forwarded by SIP Server to Avaya via DMX. There is detailed description about DMX behaviour in SIP Server Deployment Guide.

Hope it helps you a bit. Would be great if you can post it results of your testing...

R.

[kowari]

You can do it the other way around.... have SIP/asterisk agents take advantage of the avaya remote thingy and dial into avaya and use the avaya TServer instead of the SIP one?  I have forgotten what the function is called on the avaya.