Voip (SIP) switch

[Geoff Shang]

On Mon, 28 Dec 2009, geoffrey mendelson wrote:

> 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.

I can understand you wanting something simpler than Asterisk, Callweaver 
(completely open Asterisk fork) or Freeswitch, but I'm not personally 
aware of any.  Doing this under Asterisk or Callweaver is pretty trivial 
though.  I have a similar setup here, and the main reason why I use 
Callweaver is that I have an ATA with two phone ports.

> 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.

In Asterisk/Callweaver, sip.conf is the file where you'll want to set this 
up.  Note that you'll want to use Callweaver if your Linux box is behind a 
NAT, as AFAIK Asterisk doesn't support STUN.

> 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.

Again pretty simple, just set this up in extensions.conf.

HOpe you find something that suits you.

Cheers,
Geoff.