Transcoding plays a major role.... you can find some info @ voip-info ...<br><a href="http://www.voip-info.org/wiki-Asterisk+dimensioning">http://www.voip-info.org/wiki-Asterisk+dimensioning</a><br><br>~Vamsi<br><br><div><span class="gmail_quote">
On 6/6/07, <b class="gmail_sendername">Vidura Senadeera</b> <<a href="mailto:vidurased@gmail.com">vidurased@gmail.com</a>> wrote:</span><blockquote class="gmail_quote" style="margin-top: 0; margin-right: 0; margin-bottom: 0; margin-left: 0; margin-left: 0.80ex; border-left-color: #cccccc; border-left-width: 1px; border-left-style: solid; padding-left: 1ex">
<div>Dear All,</div><div> </div><div>I looking to implement asterisk solution for 2000 sip registrations and expecting con-current call about 250.</div><div> </div><div>Can some one provide me guide line that what kind of server will fullfil the requirment.
</div><div> </div><div>what is the Processor, RAM ???<br clear="all"><br>-- <br>Thanks & Regards,<br>Vidura Senadeera,<br>Network Engineer,<br>Debug Solutions<br>Sri Lanka.<br>Tel - +94114520036<br>Mobile - +94777766596
<br>Web - <a href="http://www.debug.lk" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)"> www.debug.lk</a> </div><br>_______________________________________________<br>--Bandwidth and Colocation provided by
<a onclick="return top.js.OpenExtLink(window,event,this)" href="http://Easynews.com" target="_blank">Easynews.com</a> --<br><br>asterisk-users mailing list<br>To UNSUBSCRIBE or update options visit:<br> <a onclick="return top.js.OpenExtLink(window,event,this)" href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">
http://lists.digium.com/mailman/listinfo/asterisk-users</a><br><br></blockquote></div><br>