Voip (SIP) switch

[shimi]

On Mon, Dec 28, 2009 at 5:16 PM, geoffrey mendelson <
geoffreymendelson at gmail.com> wrote:

> I am looking for advice or pointers to web sites with advice (in English)
> about setting up a SIP switch and would appreciate any help.
>
> What I have is several SIP accounts which I want to integrate into one soft
> phone. The soft phone will run on MacOS ( I already have it) and the switch
> will run on UBUNTU, I have a choice of a system running 9.04 and another
> running 9.10.
>
> It's very simple. My computer will make outgoing connections to the various
> SIP servers, none will attempt to connect to mine. Once registered, calls,
> probably one a day between all of them, will be all routed to the soft
> phone.
>
> Dialing out will be more complicated. calls begining with 1 or 0 will go
> out on one connection, I can use prefixes such as 9 or 8, etc to call on the
> other connections.
>
> While I assume that Asterisk is the software of choice, it does not matter.
> In fact, something simple would be prefered over something complicated with
> more features I won't use.
>
>
>
If it's possible to remove the "UBUNTU" requirement, you can simply throw a
Trixbox installation CD to your CDROM, hit an enter key, wait, and then have
a fully installed Asterisk (on centos), with a Web UI that can easily
configure anything you wish.

HTH,

-- Shimi
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://mailman.cs.huji.ac.il/pipermail/linux-il/attachments/20091228/2e5f8ddf/attachment.html>