[Asterisk-Users] Integrating Asterisk

[mitcheloc at gmail.com]

Just be sure that if you ditch your POTS line that you have a proper
way to terminate 911 calls!

On 6/3/06, Martin Joseph <ast at stillnewt.org> wrote:
>
> On Jun 3, 2006, at 2:04 PM, Dakota Burns wrote:
>
> > What I was attempting to visualize is the following case:
> > 10 people in an organization pick-up their phones to make an outbound
> > call. Before integrating Asterisk, all calls route through their
> > current non-VoIP based phone provider. After integrating 1 trunk
> > from a VoIP service provider into their system that provides 4
> > simultaneous calls (Teliax's Corporate plan), and dropping 4 legacy
> > lines, if 10 people make calls simultaneously, some will be VoIP and
> > some will be legacy based. Based on the above example, I'm
> > questioning whether it would be best to configure a Sipura 3000 for
> > every analog phone (I'm guessing the non-profits will want to keep
> > their existing analog phones), or utilize another device (or devices)
> > to connect the company's internet service into their existing Trunks
> > or POTS. I think the former would be easier & something I know how to
> > do, but the latter may be smarter & more cost effective. So the
> > latter is what I'm questioning whether either of you have experience
> > implementing.
> >
> Personally I think it's better to get rid of the POTS lines and got to
> a "real" VoiP terminator.
>
> I am really an experimenter only,  but my initial goal was to setup a
> way to share my existing PSTN line via an FXO like the wellgate 3701a.
> This turns out to be quite a bit of a problem due to crappy hardware (I
> started with the HT-488 but found it to be useless)  and problems with
> my local loop (ie echo).
>
> Even with all the fussing I have done, I still have a very bad echo for
> the first few seconds of some calls, until the echo can. trains and
> knocks the echo out.
>
> Conversely,  with Voip providers like Teliax (very good), Nufone.net
> (very good), I found that there are no such issues, and the most
> serious QUALITY issues are due to the routing of my data over the
> public internet to these companies.
>
> SO, in conclusion.  Just because a particular Voip terminator is good,
> doesn't mean they will work well for you.  Check the routes to them!
> Having said that, I found a third Voip call terminator that is very
> close to me (sellvoip.net), and have configured that as my primary
> terminator (asterisk will fail over to nufone and teliax if needed).
> This arrangement works great, allows for inward dialing, and is very
> cost efficient.  If I had realized this to begin with, I would have
> skipped the whole PSTN aspect of my setup.
>
> Asterisk is SUPER flexible.  you can set it up to route calls based on
> many criteria.  For example, my setup routes 7 digit calls through my
> PSTN, because I already pay qwest 18$(us) per month, so these calls are
> "free".  If I dial 10 digits (US long distance) the calls are routed
> through sellvoip.net. If I dial an Israeli cell phone, the calls are
> routed through teliax (better rate).
>
> Hope this helps a bit.
> Marty
>
>
>
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