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Chris,
So you have an asterisk system for your calling/PBX, and asterisk uses
VoicePulse for connectivity. I assume VoicePulse is pure public internet
SIP traffic correct?
I'll be honest, everyone that I talk to has had issues using asterisk at
an office site with a remote public internet SIP provider. I mean it
works, but there are these intermittend outages issues.
It all comes down to price. The pure public internet SIP traffic is the
cheapest, but has the most stability issues.
The next level up is to get a DS1 data circuit from a provider that offers
SIP over that private line (i.e. MPLS). This will be much much better for
overall reliability. Price wont be too bad once you get past the $400 or
so for the data line. Some big providers (like XO) are starting to do this
more and more now.
Another option is to simply get voice lines right into the asterisk box.
Going that route, the cheapest option is to get a channelized voice T1
with 24 analog ports coming of a MUX. You can then feed those lines into
asterisk via a digium card, OR, use an IAD and convert it to SIP going
into asterisk via TCP/IP. The analog digium card is probably the best bet.
A channelized voice T1 is pretty competitive price wise, much less then a
data T1.
-John
----------------------------------------------------
>From : W. Chris Shank <shankwc@acetechgroup.com>
To : plug <plug@lists.phillylinux.org>
Subject : [PLUG] asterisk
Date : Wed, 7 Nov 2007 15:30:13 -0500 (EST)
> We have been using VoicePulse as a terminator for our asterisk server
but about once per week we run into an outage where we drop calls and we
can't sync for several minutes. I'm looking for a reliable VoIP
alternative and/or considering getting line cards and sending the calls
directly to us.
>
> can anyone make a suggestion for a good local provider? anyone familiar
with asterisk the digium cards? this isn't a project I have time for.
>
>
> --
> W. Chris Shank
> ACE Technology Group, LLC
> www.myremoteITdept.com
> (610) 640-4223
>
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