[asterisk-users] In the beginning-The first question.

Steve Langstaff steve.langstaff at citel.com
Wed Nov 15 02:52:36 MST 2006


> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com 
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of 
> James R. Stevens
> Sent: 14 November 2006 20:36
> To: asterisk-users at lists.digium.com
> Subject: [asterisk-users] In the beginning-The first question.
> 
> List,
> Im a Cisco certified Network guy with little telecom 
> experience (BRI/PRI at the time) so please forgive my 
> terminology. I am showing interest after the Network World 
> SHSU October 4 article. We have 3 offices (Hub-Spoke T1 Frame 
> relay to the remote offices(Data & voice on separate T)). 
> Each office currently does their own thing for telecom. Our
> Main(HUB) office currently has 14 channels of T1 into an ADIT 
> 600 punched down to the DEMARC. Our Panasonic (72 port) 
> VB-43050 DBS picks up from the DEMARC and spits out 4 lines 
> for our VM server. My goal is described below, the question 
> is how to make Asterisk do it.
> 
> Consolidate telecom services of the other two offices into 
> our HUB office. Try (Hard) to keep some of the current phones 
> (Panasonic-Digital_ Not a high priority). 

You could use a Citel SIP Handset Gateway (http://www.citel.com) to keep
the Panasonic DBS phones. This unit converts their proprietary
signalling to SIP.

Disclaimer: I work for Citel.




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