[Asterisk-Users] Routing SIP calls via URI

[Shad Mortazavi]

Dear All,

I have the following setup;

SER/External Asterisk <--> Firwall <--> Internal Asterisk <-VPN-> Users

At the moment; 

Anybody can register with our SER proxy and call each other using VoIP.

Anybody can call one of our internal users via our SER/Asterisk gateway.
The INVITE is sent to our external Asterisk Server, this act as a UA and
uses IAX2 to send the call to our internal Asterisk server.

Our internal users use a VPN to connect to our corporate HQ. They
register with our Internal Asterisk server and can make internal and
PSTN calls. 

What I would like to do is to redirect external SIP calls to our
external Asterisk server. e.g if I call sip:shad at voipdomain.org I would
like the call to  be routed from our Internal Asterisk server to our
External Asterisk server via IAX2 and for the external asterisk server
to act as a UA and make the call.

I have tried the following syntax on our internal server;

exten => _sip.,1,Dial(IAX2/bxxxxxx:yyyyyy at 192.X.y.x/${EXTEN}) 

However this does not seem to work?

How do I change my dial plan so that SIP calls are routed from my
internal Asterisk box to my external Asterisk box over IAX2?

Warm Regards and Thanks

Shad Mortazavi
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Nexus Management Inc