Subject: Re: Internet telephony. (In Turnette, tell a phony?)
To: Richard Rauch <>
From: Heison Chak <>
List: netbsd-users
Date: 03/25/2005 07:57:20
On Fri, Mar 25, 2005 at 04:06:15AM -0600, Richard Rauch wrote:
> On Wed, Mar 23, 2005 at 09:01:26PM -0800, Andrew Gillham wrote:
> > Asterisk is supposed to work pretty well on NetBSD.  I am currently
>  [...]
> Thanks for the extensive answer.  Sorry for not replying earlier.
> I guess that the good news is that it sounds like I can do what
> I want, but the bad news is that I have to sift through quite a
> bit of info.
> Alas, there is no pkgsrc package for it.  (^&
> Maybe sometime this weekend I'll see if I can build it by hand.
> > but I am fairly familiar with getting the basics setup with Asterisk,
> > inter-asterisk-exchange connections, Digium hardware setup, etc.  Feel
> > free to ask questions.
> Okay.  Thanks.  I'll take you up on that; here are a few offhand:
>  * If I ever want (or for some reason need) a traditional phone,
>    what would I be looking at by way of hardware to add to the
>    computer?  (I am familiar with none of this...)

You will need to source an FXS interface for POTS phones, Digium 
carries those in 2 forms:
X100P - a single port FXS interface
FXS modules for their 4 port cards

>  * What kind of CPU would be required?  My fastest system should be
>    more than sufficient (AMD64), but every 15 seconds or so it seems
>    to have a system-wide freeze for a second (this started around August,
>    last year, give or take, and persists with a 3.99.1 kernel that I
>    built a couple of days ago).  The next fastest computer that is
>    directly on the 'net is an old 450MHz (I think) Pentium III.
>    (Inbetween, I have an 800MHz Athlon that is off of the 'net...
>    I thought that I read something about NAT causing problems for
>    this, though, and that computer is not likely to be put directly
>    on the 'net.))

System requirement and performance depend highly on what your intend of 
use will be. Whether you'll be using it's conference feature to join 
2 or more audio streams (e.g. 3 way calling, conference room, music on hold,
etc.); and whether or not the server is doing transcoding for the 
converstaion (i.e. when CODEC on the 2 communicating channels are different,
Asterisk will have to translate before sending)

> -- 
>   "I probably don't know what I'm talking about."