Asterisk / VOIP for small business

Thu Oct 15 16:43:48 EDT 2009

Hi Greg,

I hope Fonality has removed the fact that they have a daemon with root
access in PBXtra, which phones home.  There has been a lot of press about
that in the past.  My previous employer had Fonality for a while... but they
swiched to a non-phone-home version of Asterisk.
I guess I'm a bit prejudiced, but I have been using the completely
open-source PBX-in-a-Flash distribution (http://www.pbxinaflash.com)  since
it's inception, and it's been a rock-solid performer in a home office
environment.   They offer high-level paid support if you need it.   The
forum is very active and informative.   The Distro is CentOS5 + FreePBX +
Asterisk;   very unique is the rebuild asterisk from the source any time you
need to, and nothing breaks.  It has 32 and 64-bit flavors.

Many business customers use PSTN for the primary number, and roll over or
all outbound on SIP/IAX trunks.  It saves a lot, and you limit the risk of
not having your inbound down to some internet issue.

Did you know that you can call cellular users outbound and have no minutes
charged on outbound calls?   The SIP provider  Gizmo5.com has peering
arrangements with all the US cell providers to route calls for no carriage
fee.

There are several sites for white-page listing SIP/IAX DIDs in the telephone
book.. you pointed out one.

On Thu, Oct 15, 2009 at 2:44 PM, Greg Rundlett (freephile) <
greg at freephile.com> wrote:

> == 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
> -skype/aim/irc/twitter freephile
> http://profiles.aim.com/freephile
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