Asterisk / VOIP for small business

[Greg Rundlett (freephile)]

== Intro / Background ==

I've inherited an Asterisk (based) system, and I'm pretty happy about
that.  On the other hand, it's complicated, new and a critical system
so I'm learning telephony as fast as I can.

I work for a Real Estate company.  We have 8 offices and we have ~200
agents in the field, all using cell phones.  We have tied in at least
one office with a VPN and VOIP phones.  In the future, I can imagine
perhaps using VOIP mobile handsets.

The current system is a PBXtra system (Asterisk-based) from Fonality
http://fonality.com/ hosted on-site in our Portsmouth, NH office.

When we acquire a field office, we have sometimes setup remote call
forwarding through the traditional carrier so that the line then
forwards into the PBXtra.  This breaks down for a few reasons:
1) Dealing with FairPoint communications for anything has been a
complete nightmare - delays, misinformation, repeating requests
without results.
2) Apparently call forwarding types are not all the same, and anything
but 'remote' call forwarding depends on the number of 'paths' to be
able to handle more than one call (resulting in busy signals for
caller b). (Explaining what you want to happen to Fairpoint does not
actually result in getting what you want - see 1)
3) Remote call forwarding solves the busy signal problem, but then you
lose your listing in the phone book.

Our main private branch exchange (PBX) phone system connects to the
PSTN / POTS through one of two trunks. (Trunks can be a plain old
RJ-45 hardware card, a T1/PRI card, or a VoIP trunk using the SIP
connection or IAX protocol.) The Portsmouth, NH PBX (PBXtra) has two
different trunks- one is a T1 with BayRing_Communications. The other
is an IAX trunk with Junction Networks.  So, the telephony routing is
like this:
1.) POTS -> Bayring -> T1/PRI -> PBX

The main corporate phone number is pointed to the T1 which is provided
by Bayring and connected to a hardware T1 card in the PBX system

Satellite offices have the following connection:
2.) POTS->Junction Networks/OnSip->IAX/SIP->PBX

In order to solve the problem of having callers receive a busy signal,
 I'm about to move the branch office line to a SIP line using
OnSip.com.  This means that we can retain the published phone number,
but make it a VOIP line so that it can be handled by the PBXtra.  Two
obvious concerns are:

a) checking the total bandwidth capacity at our office to ensure the
additional call volume can be handled (already done when we initially
just forwarded the line)
b) getting the phone number listed in the White pages and Yellow
pages. (already doing this
http://www.junctionnetworks.com/knowledgebase/junction-networks/getting-listed-in-white-pages)

== Question / Discussion ==

I'm just sharing my experience in case it is helpful to others on the
list.  And I'm also hoping that people can tell me if they've been
down this road before and can share suggested tips or best practices.
I know there are list members using VOIP, Asterix, etc., and since I'm
new I'd even appreciate tips on what are the best tech forums for this
area.

p.s. Since a port of an existing number can take 4-6 weeks, I guess I
will have to try yet again to get Fairpoint to switch my service to
Remote Call Forwarding.  It would be totally impractical (and
expensive) to install additional copper lines just to get additional
simultaneous call capacity (aka a hunt group).  In fact, I already
have a couple extra lines in place.  Those are reserved for "back
channels" - calling from the main office into the branch office.  As
someone recently pointed out, the solution to problems comes from
writing a message to explain it - so I'll take at least one secondary
line and add it to the main line in a hunt-group configuration in case
Fairpoint can't / won't make the primary line a Remote call-forward
line.

Thanks,
~ Greg

Greg Rundlett

nbpt 978-225-8302
m. 978-764-4424
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