Don't forget to be sure your power supplies are reliable, and if necessary redundant.<br><br>
<div><span class="gmail_quote">On 6/13/06, <b class="gmail_sendername">Colin Anderson</b> <<a href="mailto:ColinA@landmarkmasterbuilder.com">ColinA@landmarkmasterbuilder.com</a>> wrote:</span>
<blockquote class="gmail_quote" style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid">Maybe you are over thinking it. I have 200 users on a quad Xeon 700 with 2<br>PRI's and we regularly have 40-60 channels up, no problem (believe me, if
<br>there was a problem I'd have 200 guys freaking on my head). I rarely see ><br>30% single-CPU usage, and that's only when Sendmail is invoked to send out a<br>voicemail.<br><br>But yes, transcoding and reasonable echocancel values is key. If you are
<br>connecting to the PSTN, ulaw all the way. If you are connecting to a<br>provider, use the codec of your choice as long as your provider supports it,<br>and make sure every phone and endpoint is set to use the same codec.
<br><br>I also have 30 IAX remote sites that support from 1 to 5 users, on P-II<br>233's. I use them because they are bulletproof and they are so cheap if<br>something gets hosed we just throw it away and put in another one. Again, no
<br>problem<br><br>Maybe try your cheapo machine and if it doesn't work try a better box. You<br>already have the cheap machine, and the card will remain the same regardless<br>of what box you use.<br><br><br>-----Original Message-----
<br>From: Erick Perez [mailto:<a href="mailto:eaperezh@gmail.com">eaperezh@gmail.com</a>]<br>Sent: Tuesday, June 13, 2006 9:15 AM<br>To: Asterisk Users Mailing List - Non-Commercial Discussion<br>Subject: Re: [Asterisk-Users] Can this config sustain 30 users?
<br><br><br>Well thanks all for your responses. My original intention was to<br>address the mistic know-how about machine calculations, and I still<br>feel the shadows remain.<br><br>Why? Because to achieve a 24 user PBX-only/One E1, I was going to
<br>install a Dual Xeon with 2 GB of RAM and a 3ware card runing RAID-1<br>with two sata3 disks.<br>Now This thread tells me that my dual core pentium d (a 700$ computer)<br>will do the work. (the other equipment costs about
3500.00$). I do<br>realize that i must minimize transcoding (ulaw all the way) but you're<br>telling me it will work for 24 users (let's say 30 for round numbers)<br>all with SIP phones in an IP network.<br><br>Below are some comments that i found googling and doing some
<br>calculations myself. I do not enforce or deny any of them, please feel<br>free to tell me if Im wrong.<br><br>(not confirmed)a- A voice channel takes 30mhz (60mhz in duplex mode).<br>So A 2.66 Ghz CPU can sustain about 43 calls (2600mhz/60mhz=43calls),
<br>not taking into account other factors that may increase/decrease the<br>number of calls at the same time.<br><br>b- 24 users talking ulaw (-+ 80kbps per channel) consume 1920kbps and<br>in full duplex they consume 3840kbps (about
3.75 megabits/s).<br><br>c- To Calculate the bandwidth DDR memory can achieve (example PC4200)<br>,to get the transfer rate, multiply the width of the module (8 Bytes)<br>by the rated speed of the memory module (in MHz): (8 Bytes) x (533
<br>MHz/second) = 4,264 Mbytes/second or 4.2 Gbytes/second (34gigabits/s),<br>hence the name PC4200<br><br>So, will all of this in mind,<br>CPU Dual Core 533FSB, 2.66 Ghz speed<br>DDR533mhz, One gigabyte. (2x512)<br>Two Sata disks (each sata pumps
1.5 gigabits/s)<br>Motherboard Intel 945 at 533FSB<br><br>Means that the cpu,the ram and the board can achieve (see point b)<br>about 34 gigabits of data transfer, but 24 users only generate 3.75<br>megabits. So this is more than covered.
<br>However if we take into account the lowest performing component on<br>this system (the sata disks) we go down to 1.5gbits/s which still<br>seems to be enough.<br><br>Please please correct me if im wrong (or crazy)<br>
<br>Thanks,<br><br><br>References:<br><a href="http://en.wikipedia.org/wiki/Front_side_bus">http://en.wikipedia.org/wiki/Front_side_bus</a> (bus bandwidth table)<br><a href="http://www.acme.com/build_a_pc/bandwidth.html">
http://www.acme.com/build_a_pc/bandwidth.html</a><br><a href="http://www.lostcircuits.com/memory/ddrii/">http://www.lostcircuits.com/memory/ddrii/</a><br><a href="http://bvio.ngic.re.kr/Bvio/index.php?title=Front_side_bus">
http://bvio.ngic.re.kr/Bvio/index.php?title=Front_side_bus</a><br><br><br>On 6/13/06, Mike Fedyk <<a href="mailto:mfedyk@mikefedyk.com">mfedyk@mikefedyk.com</a>> wrote:<br>> Erick Perez wrote:<br>> > I just don't want to install it and then after a 5th user going to
<br>> > call someone the asterisk begin to crash due to lack of resuources.<br>> Check the wiki for SIP load generation tools you can use to test your<br>> setup on any number of calls you like.<br>> _______________________________________________
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http://lists.digium.com/mailman/listinfo/asterisk-users</a><br>><br><br><br>--<br>------------------------------------------------------------<br>Erick Perez<br>Panama Sistemas<br>Integradores de Telefonia IP y Soluciones Para Centros de Datos
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