(My) Asterisk bias (Mk. II) (was: Asterisk / VOIP for small business)

Ken D'Ambrosio ken at jots.org
Thu Oct 15 20:03:39 EDT 2009


Forgot one other advantage of do-it-yourself: embedded systems. For
example, I've got a Sheeva Wall Wart.  For a VoIP-only solution, it's just
about perfect.  But I'm unlikely to find a turnkey solution.  With Ubuntu
for Arm, however, I had Asterisk installed with a simple "apt-get".  (The
great thing about VoIP-only Asterisk solutions is the distribution's stock
Asterisk is fine -- generally, the only kernel modules you have to worry
about are dummy timing devices that are usually included as a source
package or somesuch.)

-Ken

<soapbox>
I like Asterisk.  A *lot*.  And not just because it's cheap, but because
you can get your hands as dirty as you'd like.  Unfortunately, most (I'm
tempted to say "all," but lack of proof prevents me) distributions of
Asterisk don't really really let you get in there.  Most of them (for
example) create the configuration files from an intermediate storage
mechanism that they "understand."  This allows for easy back-end
manipulation without having to parse Asterisk's (sometimes esoteric)
configuration files -- but prevents you from doing tricks.  For example, a
tweak I did that I particularly enjoy is I attach in-bound faxes to a
user's e-mail... but also have an HTTP link to an upside-down version on
the server, should it have been fed into the fax machine incorrectly.
Also, due to flaming incompetence on the part of our telecom system folks
(General Communications: just say "No."), they screwed up all sorts of
things that were most easily fixed by throwing a quad-span T1 card in
front of our two other PBX systems, and letting the Asterisk box do the
routing, call accounting (which, once you have Asterisk put your CDR's
into MySQL, was a not-too-difficult Python script), VoIP headsets, etc.
All things that happy, friendly GUIs get in the way of to some extent or
another.

Now, for a simple fire-and-forget small-office setup (especially where
POTS lines are involved, which minimizes the fun you can have with DIDs),
a turnkey solution is superfine.  But when you get bigger, I find a
well-documented, from-scratch Asterisk install is easier in the long run.

</soapbox>

-Ken

P.S.  One other advantage: the *insanely* helpful asterisk-users mailing
list generally won't help out much if you're using a shrinkwrap Asterisk
solution.  Kind of like posting to LKML with a tainted kernel.


On Thu, October 15, 2009 4:43 pm, Gerry Hull wrote:
> 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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>>
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