asterisk users - Feb 2008 - Telephony Migration Hardware

We are maxed out on our legacy PBX, and the question is the process of
migrating to a new * system from the legacy.  We current have 36 FXO
lines coming into our site, and the usage on these lines indicates we can
spare a few of them to launch the * server, and then move additional lines
over as the * server/network/cable plant is built out and traffic moves to
*.   Eventually, we will want to merge all oustide traffic to SIP WAN
circuit terminated on *.

The question is what * telephony interface hardware to use for the
migration.  At the beginning we need a couple of FXO ports, and at the end
we will want to have some kind digital trunking.

Any recommendation for hardware to use throughout  this migration process?

Thanks a bunch!
.
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> We are maxed out on our legacy PBX, and the question is the process of
> migrating to a new * system from the legacy.  We current have 36 FXO lines
> coming into our site, and the usage on these lines indicates we can spare a
few
> of them to launch the * server, and then move additional lines over as the
*
> server/network/cable plant is built out and traffic moves to *.  
Eventually, we
> will want to merge all oustide traffic to SIP WAN circuit terminated on *.
> The question is what * telephony interface hardware to use for the
migration.
> At the beginning we need a couple of FXO ports, and at the end we will want
to
> have some kind digital trunking.
Bit difficult to offer you useful advice without knowing which country
you're in (and hence what the local telco will do for you free of charge),
but I'll try and answer as if you were based in the UK.

If your eventual target is to have all calls coming in via IP, I'd recommend
one of the low-end Digium FXO cards (TDM400 with a couple of FXO modules). This
will give you a couple of analogue channels for things like emergency services
access etc. and avoid the need for you to register and (potentially) pay for
PATS (aka E911 in the US).

You'll then want to sort out whoever you're using for IP call
termination and create them as a  peer within asterisk. I'll assume you
already know how to do that. Hopefully, whichever company you're using for
inbound calls will provide you with a temporary number at this stage, which you
can use to test call quality on inbound calls.

Once you're satisfied the new server is behaving as it should, you can
contact your analogue line provider and ask them to forward calls from your
existing lines over to your temporary number from your IP provider. Certainly in
the UK, although you'll be charged divert fees for calls to the number,
there's no monthly charge for doing this.

Give it a week or so like that to make sure everything's fine, during which
time if any problems come to light, you can simply phone the telco and ask them
to cancel the divert. After that, you should be able to port your number(s) on
your analogue lines over to your IP trunk provider.

Regards,

Chris
-- 
C.M. Bagnall, Director, Minotaur I.T. Limited
For full contact details visit http://www.minotaur.it
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