<br><br><div class="gmail_quote">2013/3/7 Steve Edwards <span dir="ltr"><<a href="mailto:asterisk.org@sedwards.com" target="_blank">asterisk.org@sedwards.com</a>></span><br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
Please don't top-post.<div class="im"><br>
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On Thu, 7 Mar 2013, Bharat Lalcheta wrote:<br>
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<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
You can use ATA box with pstn phone to reduce cost.<br>
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Are you wiring a building where multiple-line SIP gateways make sense?<br>
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How about a description of what you are trying to do?<br>
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Personally, I like Polycom SIP phones but I don't have to buy 1,000 of them :)<div class="HOEnZb"><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>