[Asterisk-Users] Routing SIP calls via URI

[Bobby Lee]

I believe that they covered this exact procedures at www.voip-info.org.  
Look for the topic on connecting two Asterisk servers.  They outline three 
different ways that you can do so.


>From: "Eric \"ManxPower\" Wieling" <eric at fnords.org>
>Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion 
><asterisk-users at lists.digium.com>
>To: Asterisk Users Mailing List - Non-Commercial Discussion 
><asterisk-users at lists.digium.com>
>Subject: Re: [Asterisk-Users] Routing SIP calls via URI
>Date: Wed, 29 Mar 2006 13:18:07 -0600
>
>Shad Mortazavi wrote:
>
>>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/${EXTEN})
>>
>>However this does not seem to work?
>
>Have you tried this?
>
>exten => shad,1,Dial(IAX2/bxxxxxx:yyyyyy at 192.X.y.x/${EXTEN})
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