[asterisk-users] high capacity analog <-> sip gateway

Valer Nur valernur at yahoo.com
Fri Oct 26 08:51:29 CDT 2012


Hi Carlos,
To solve the echo problem from your 96 analog ports, you can use the PBXMate.
Valer.




________________________________
 From: Carlos Alvarez <carlos at televolve.com>
To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> 
Sent: Thursday, October 25, 2012 10:57 PM
Subject: Re: [asterisk-users] high capacity analog <-> sip gateway
 



On Thu, Oct 25, 2012 at 1:21 PM, Justin Killen <jkillen at allamericanasphalt.com> wrote:

I’m looking for an fxs <-> sip
gateway/router/switch for about 100 existing analog phones.  I’d
like to get this done cheaply, but I want to make sure that whatever we buy
works well with asterisk as well.  As far as I can tell, digium make no
such device.  The only ones I’ve been able to find with a 48 port
capacity are these two:

I have a deployment of 96 analog ports using a Digium T1 card ($500 on eBay) and Rhino analog channel banks (also cheap on eBay).  We have extremely high reliability from this configuration.  In fact, other than the normal analog annoyances like occasional echo, they are rock solid.

Are you doing this instead of VoIP phones for cost reasons?
-- 

Carlos Alvarez
TelEvolve
602-889-3003


--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
              http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20121026/d3bbc47c/attachment-0001.htm>


More information about the asterisk-users mailing list