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SIP trunks, SIParator connecting to off-system device

Dec 16, 2011 03:36 AM PST
Nolan Tininenko
NVOQ Incorporated
Was wondering if anyone had advice/experience in setting up the following:

We have a homegrown digital receptionist (we call it a Voice Activated Dialer, or VAD) that

1) prompts the caller to say the name of the party they're trying to reach

2) performs voice recognition

3) transfers the call to the user's extension matched by the voice reco.
This digital receptionist works on SIP and was set up on our previous analoq system like an auto-attendant to work with a Cisco device which sent the SIP message to the VAD whenever a particular number was dialed. Currently the VAD does not work like a full SIP phone (it doesn't register with a SIP server), it just receives the SIP message and then transfers the call. We're looking at implementing this, but not sure if/when it'll happen.

Does anyone have any ideas on ways to tie this digital receptionist into our new ShoreTel system? Here's the thoughts I've had so far:

1) If we can get the VAD to register as a SIP endpoint, we may be able to tie it in to the ShoreTel system just like any other SIP phone.

2) Somehow use a SIP trunk, so that calls to a particular extension would use this trunk and therefore be directed to the VAD.

3) Use a Ingate SIParator to assist in the setup of a SIP trunk for communication to the VAD.

Is #2 possible? If so, how would I set it up so that all calls to extension 4000 (for example) would be directed to route over a SIP trunk, and therefore ending up at the VAD?

Thank you for any ideas you can provide.
Jan 19, 2012 11:52 PM PST
Chris Herring
Computer Instruments, Inc
Did you ever get any information for this post? We have been searching on how to set SIP End Points with Dialogic HMP. Having issues with registration. Do you have any information that might help?
Mar 04, 2012 07:10 PM PST
Nolan Tininenko
NVOQ Incorporated
Sorry, never got a notification that anyone posted here. Unfortunately I never did any information for this. I thought I had figured out how to create the SIP trunk and direct it to the correct IP but I never got calls to route over it.

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