[Asterisk-Users] YAAN (Yet Another Asterisk Newbie)

[Lyle Giese]

You need two 4chan TDM cards with six total FXO modules to drive the six
incoming lines on the Starplus.  I am assuming that you will be taking out
your POTS lines going into the StarPlus.  ATA's or a small channel bank will
also work for the * to analog conversion.

And please don't post html to the list.

Lyle

----- Original Message ----- 
From: Dave Covert (Sailtech)
To: asterisk-users at lists.digium.com
Sent: Wednesday, August 25, 2004 9:15 AM
Subject: [Asterisk-Users] YAAN (Yet Another Asterisk Newbie)



I plan to set up an Asterisk server later today or tomorrow to begin putzing
and learning about it. Learn by doing...

I would like to cut thru some of the confusion that such a flexible system
tends to breed by quickly describing my end goal and getting some input from
the 'group mind' as to the pieces I should concentrate my efforts on.

We are a 5-person operation with 6 VOIP numbers an old-style POTS PBX
(Vodavi Starplus 616EX) and a dozen 6-line desk phone stations. Rather than
using a small bank of ATAs, we would like to use an Asterisk server to
'terminate' the VOIP lines and route them to both the Starplus desk phones
and to softphones running on certain workstations. That is, a new incoming
call would ring both the first unused line hooked to the Starplus and the
first unused line on the softphones.

So... The question is... to get that to work, what sort of hardware do I
need in the Asterisk box to turn the incoming VOIP calls into a two-wire
POTS input for the Starplus PBX and what is a suggested softphone we can use
with Asterisk?

Thank you for your time,
Dave Covert, KB5GOG | Sailtech | Office 281-334-4690 | Fax 281-538-3270 |
Email dave at sailtechmarine.com | www.sailtechmarine.com
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