[asterisk-users] Newbie h/w Q, and confirming basic concepts

Brian Candler B.Candler at pobox.com
Fri Oct 6 03:36:53 MST 2006


On Thu, Oct 05, 2006 at 07:22:16PM -0700, Mike Morris wrote:
>    I'm preparing for my first asterisk install, and would like to ask a
>    hardware question & confirm my understanding of some basics:
>      * The Q: I'm looking for 2 FXO ports to have asterisk answer 2
>        incoming lines. There are single FXO port cards for about $30...
>        but dual cards, or the Digium "400" cards, are all several hundred
>        dollars. Why is this? Are the chipsets so different, or am I
>        missing something?

FXS ports are a little bit more sophisticated - they have to provide voltage
to ring the phone for example. However, the reason the FXO cards are so
cheap is that they are basically WinModems (and hence obsolete
consumer-grade gear being shifted out)

You do have another alternative: buy an ATA (analogue telephone adaptor)
which has, say, one FXO and two FXS ports, and connects to your LAN using
ethernet. It talks to your Asterisk server using SIP. This probably works
out cheaper than a TDM400P. You also get the advantage that it may reduce
the CPU load on your box, since you can arrange for the media streams to run
directly between the ATA and your local softphones (i.e. the Asterisk box
handles signalling but not audio). This needs the ATA and your softphones to
support "reinvite", so that Asterisk can switch itself back into the audio
stream when necessary (e.g. for conferencing, voicemail etc)

The other advantage of ATAs is that they let you build simpler networks. If
you don't want the features and complexity of a local Asterisk server, you
can point your ATAs and VoIP phones all at an upstream SIP provider like
sipgate. You can either give your phones separate accounts, or register them
all with the same account (sipgate handles multiple registrations and
forking, so that a call to your number will ring all the phones)

If you're running behind NAT then you probably need to add a simple SIP
proxy like siproxd, but that's something very simple and tiny compared to
Asterisk.

OTOH, if you do it that way, you deprive yourself of the experience of
building and running your own softswitch :-)

HTH,

Brian.


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