[asterisk-users] Relaying calls to another SIP extension

[Paul Campbell]

Hi,

 

I am learning Asterisk for a small project.  

 

At this stage I have an AsteriskNOW system running locally. I can call
SIP phone to SIP phone fine, the operator and voice mail work fine,
except some stuttering (probably caused by it running in MS-VPC)

 

What I need to figure out is...

 

I have an automated response telephony server (Voice Media Server)
available via a SIP URI like:

 

test at foo.bar.com

 

If I place this into the SIP phone, X-Lite the call goes through fine...
but not through my Asterisk.

 

What I need to do is to add this service 'through' my Asterisk.  So when
a user calls a certain extension through Asterisk it 'rings' the
extension test at foo.bar.com as if it were a standard SIP phone.

 

That is stage 1.

 

Stage 2 is for the media server handling test at foo.bar.com to be able to
forward the call onto another SIP phone, allowing it to drop the call
completely and the call tromboning or bridging to happen in the Asterisk
PBX and not take up 'lines' on the media server.  (Eventually both SIP
ends may become PSTN's).

 

I'm having trouble deducing how to do stage 1.  In asteriskNow I don't
see a way to add a calling rule for this.  I tried adding a service
provider "foo.bar.com" and a calling rule to send all calls for
extension 6002 to that provider, but all I get is "Service unavailable".

 

With the Asterisk-docs site down I'm finding it tough going.

 

Thanks for any pointers you can give me.

Paul Campbell  

 



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