[Asterisk-Users] Integrating Asterisk

[Martin Joseph]

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