[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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