Adding Asterisk between existing phone system and PSTN Re: [Asterisk-Users] Integrating Asterisk

Mike Fedyk mfedyk at mikefedyk.com
Sat Jun 3 15:04:05 MST 2006


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)
First of all, don't use a service that limits the number of calls.  Get 
the per minute plan.  That way you won't have to worry about hitting any 
soft-caps.
> , and dropping 4 legacy lines
Don't drop the lines until you have a setup that works reliably without 
hickups for at least 3 months (more if you can convince them to keep the 
lines that long).
> , 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)
Only use ATAs when you have to.  They cost about $80 anyway, why not get 
a spa-841 instead?  And why are you guessing?  You should know if they 
want to keep the phones they have.  And what type of phones are they?
> , 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. 
Let me be frank.  I'm relatively new to phone systems, but I can tell 
you need to do a lot more research before even thinking about doing an 
implementation.

If you want to keep the analog phones, they probably already go to a 
wiring closet.  You'll want to put either an asterisk box with a 
tdm2400p with 12 FXS and 12 FXO (look up the tdm2400p before asking why 
I say 12 instead of 10).  Or if you have voice T1s at that location you 
may want a channel bank instead.  I haven't used any channel banks so 
others will have to step in to give suggestions on that.

My point is that you need to post what you want your client's results to 
be instead of how to do what you think should be done.  The details I 
mentioned above are only part of one possible direction to go in, and 
there is more to it than that also and it may not even be the best for 
your situation too. 

Have you looked at their network to see if can handle the large number 
of small packets that voip produces?  What about their Internet 
connection? What is it that your client wants in a phone system that 
their current one isn't doing?  How is adding asterisk and an ATA for 
each analog phone going to help?

So, post what you already have and what you want the end result to be 
from an end-user's perspective and we can probably point you in the 
right direction.



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