[Asterisk-Users] FXS or VOIP

Colin Anderson ColinA at landmarkmasterbuilder.com
Wed Jan 11 15:19:43 MST 2006


The nice thing about the Digium TDMXXX series of analog cards is that it
allows you to mix and match FXO or FXS as you see fit. And, in FXS's
defence, it is a good way (well, the ONLY way) to bridge analog to IP. 

A P3 class box should scale to a few dozen extensions no problem. The caveat
is if the box is doing other stuff (it shouldn't) and / or if it is
transcoding (converting) codecs from one format to another, you might run
into performance issues. Best practice is to always use the same codec
end-to-end if Asterisk remains in the media stream (which you select using
canreinvite=yes or canreinvite=no in sip.conf) and this avoids transcoding
on the Asterisk box.

Depending on your point of view and your luck in getting a box running,
Asterisk is either the coolest thing ever or a gigantic pain in the ass.
Patience and problem solving skills will serve you well. 

-----Original Message-----
From: Jim Freeze [mailto:asterisk at freeze.org]
Sent: Wednesday, January 11, 2006 2:57 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] FXS or VOIP


On 1/11/06, William Boehlke <william.boehlke at signate.com> wrote:
>
> A single computer will handle hundreds of telephones. Just get a card with
> more ports, or use an external gateway.

I am sorry, I don't understand. Are you talking about analog FXS phones?
All the PCI cards I have seen have a max of 4 FXS lines and the external
boxes
seem very expensive.

--
Jim Freeze
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