Voip (SIP) switch
Andrew Kaplan
akaplan at netshack.co.il
Mon Dec 28 20:17:04 IST 2009
P
From: linux-il-bounces at cs.huji.ac.il [mailto:linux-il-bounces at cs.huji.ac.il] On Behalf Of shimi
Sent: Monday, December 28, 2009 5:38 PM
To: geoffrey mendelson
Cc: linux-il list
Subject: Re: Voip (SIP) switch
On Mon, Dec 28, 2009 at 5:16 PM, geoffrey mendelson <geoffreymendelson at gmail.com<mailto: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
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