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<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>Cost and ease of deployment, yes. At this
specifc location we are currently using Centrex lines (AT&T hosted) and are
looking for a way to move into something cheaper without throwing away the
existing phones. I like the idea of using a channel bank – I’ll look into that
as an option as well.<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>-Justin<o:p></o:p></span></font></p>
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<p class=MsoNormal><b><font size=2 face=Tahoma><span style='font-size:10.0pt;
font-family:Tahoma;font-weight:bold'>From:</span></font></b><font size=2
face=Tahoma><span style='font-size:10.0pt;font-family:Tahoma'>
asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] <b><span style='font-weight:
bold'>On Behalf Of </span></b>Carlos Alvarez<br>
<b><span style='font-weight:bold'>Sent:</span></b> Thursday, October 25, 2012
1:57 PM<br>
<b><span style='font-weight:bold'>To:</span></b> <st1:PersonName w:st="on">Asterisk
Users Mailing List - Non-Commercial Discussion</st1:PersonName><br>
<b><span style='font-weight:bold'>Subject:</span></b> Re: [asterisk-users] high
capacity analog <-> sip gateway</span></font><o:p></o:p></p>
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<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'><o:p> </o:p></span></font></p>
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<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'>On Thu, Oct 25, 2012 at 1:21 PM, Justin Killen <<a
href="mailto:jkillen@allamericanasphalt.com" target="_blank">jkillen@allamericanasphalt.com</a>>
wrote:<o:p></o:p></span></font></p>
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<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><font
size=2 face=Arial><span style='font-size:10.0pt;font-family:Arial'>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:</span></font><o:p></o:p></p>
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<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
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<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'>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.<o:p></o:p></span></font></p>
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12.0pt'><o:p> </o:p></span></font></p>
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<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'>Are you doing this instead of VoIP phones for cost reasons?<o:p></o:p></span></font></p>
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<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'><o:p> </o:p></span></font></p>
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<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'>-- <o:p></o:p></span></font></p>
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<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'>Carlos Alvarez<o:p></o:p></span></font></p>
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<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'>TelEvolve<o:p></o:p></span></font></p>
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<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'>602-889-3003<o:p></o:p></span></font></p>
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12.0pt'><o:p> </o:p></span></font></p>
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