Call Routing

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rgelfand

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I have successfully setup my sip provider. The log shows that registration took place.

I was wondering if I could get some help on the following:

Can all incoming phone calls be routed to sip uri address?

If yes, would [email protected]:5060;transport=tcp be a valid sip uri?

If not, how is routing to [email protected]:5060 setup? Is there a way to convert sip udp traffic to that of sip tcp?

Thanks in advance
 
You need to simply setup a DID to transfer calls from a particular INBOUND caller to a specific registered extension.

http://www.3cx.com/manual/3CXPhoneSystemManual7/phone-system49.html
 
In my case the client is microsoft speech server 2007. It doesn't have a facility to register with 3cx. All it is, is unauthenticated call from gateway using sip uri.

BTW... would you know of sip over tcp voip providers

Thanks for your help.
 
3CX supports registered SIP extensions and gateways to FXS and FXO....I dont think 3CX will interface with MS Speech Server... although this server is a component of Exchange 2007 UM Role..
 
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