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<div class="ecxgmail_quote">2013/3/7 Steve Edwards <span dir="ltr"><<a href="mailto:asterisk.org@sedwards.com">asterisk.org@sedwards.com</a>></span><br><blockquote style="padding-left: 1ex; border-left-color: rgb(204, 204, 204); border-left-width: 1px; border-left-style: solid;" class="ecxgmail_quote">
Please don't top-post.<div class="ecxim"><br>
<br>
On Thu, 7 Mar 2013, Bharat Lalcheta wrote:<br>
<br>
<blockquote style="padding-left: 1ex; border-left-color: rgb(204, 204, 204); border-left-width: 1px; border-left-style: solid;" class="ecxgmail_quote">
You can use ATA box with pstn phone to reduce cost.<br>
</blockquote>
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Are you wiring a building where multiple-line SIP gateways make sense?<br>
<br>
How about a description of what you are trying to do?<br>
<br>
Personally, I like Polycom SIP phones but I don't have to buy 1,000 of them :)<div class="ecxHOEnZb"><div class="h5"><br><br></div></div></blockquote><div><br></div><div>I bet it is a school assignment ... home work or the way you like to call them. However I have a box with 972 peers, no reinvite (but no transcoding), average usage of conference call and other audio mix feature, reaching a max of 60 CPS and an average of 150 channels without problems. The cpu is a double Intel(R) Xeon(R) CPU E5-2630 0 @ 2.30GHz, but it works fine even on the old hardware, a double Intel(R) Xeon(R) CPU 5150 @ 2.66GHz</div>
<div><br></div><div>Leandro,</div></div>
<BR> <BR>This is not school assignment or home work :) We need to setup in society buildings. Each flat will have SIP extension (hard phone) registered on asterisk server. Calling between SIP extensions is required. No PSTN / ITSP SIP trunking. Just like inter-com feature.<BR> <BR>One way is to install 1000 IP Phones one at each flat<BR>Secondly, install multiple-line SIP gateways with RJ-11 cabling.<BR> <BR>Is there any other low budget solution for this setup?<BR> <BR> <BR>Thanks,<BR>Kamlesh <BR><br>--
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