It is very simple take openser(opensips/openser/kamalio) the openser community is great, the project have been here and tested for a years in production, used by the biggest companyes (millions!) of users, it's a carrier grade soft ;) in combination of cdrtool + opensips + mediaproxy you can get 100% billing accuracy.<br>
<br><div class="gmail_quote">2008/11/28 Yehavi Bourvine <span dir="ltr"><<a href="mailto:yehavi.bourvine@gmail.com">yehavi.bourvine@gmail.com</a>></span><br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<div dir="ltr"><div>I did a test yesterday and did 1,000 registrations to Asterisk using SIPP. I did the register test since I am using the realtime DB and asterisk does periodic quesries to it for each registered user. Although Asterisk continued to function as usuall, it was in a steady loop querying the DB for the 1,000 users.</div>
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<div>OK, you convinced me to look at some front end to it. There are mainly three front ends mentioed here: OpenSer, SipExpress and FreeSwitch. Is there some comparison available which will save me from testing all three of them? Is there one which is more used than the others? (so it has more public QA :-)</div>
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<div> Thanks! __Yehavi:<br><br></div>
<div class="gmail_quote">2008/11/24 Steve Totaro <span dir="ltr"><<a href="mailto:stotaro@totarotechnologies.com" target="_blank">stotaro@totarotechnologies.com</a>></span><br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0px 0px 0px 0.8ex; padding-left: 1ex;">Fronting with OpenSER or FS, you should have no problems providing you<br>plan to use SIP extensions.<br>
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What is critical are the max simultaneous trunks you are going to use.<br><br>I would go TDM although universities have good bandwidth, and SUPERIOR<br>bandwidth between others.<br><br>I would think a TDM DS3 or two just to be safe. It should be pretty<br>
trivial besides gotchas, like cat3 to the rooms, although channel<br>banks may be an even better solution if phones are already in place.<br><br>Then you just use SIP when needed or wanted, and Asterisk is simple,<br>although more costly.<br>
<font color="#888888">--<br></font>
<div>Thanks,<br>Steve Totaro<br>+18887771888 (Toll Free)<br>+12409381212 (Cell)<br>+12024369784 (Skype)<br><br><br></div>
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<div>On Fri, Nov 21, 2008 at 6:24 PM, Wilton Helm <<a href="mailto:whelm@compuserve.com" target="_blank">whelm@compuserve.com</a>> wrote:<br>> Yet another option is a commercial system with in-house staff. I used to<br>
> maintain a NEC (NEAX 2400) for many years. I went to factory training and<br>> had total responsibility for it. Some manufacturers discourage or prevent<br>> this, but others are open to it. There are also 3rd party organizations<br>
> (such as Source) that can supply parts and even expertise for those going<br>> that direction. Whether the result would be higher availability than<br>> Asterisk, I don't know. Given I'm both a telco guy and a computer guru (CS<br>
> degree) I'd probably go the Asterisk route myself, because its open and I<br>> would have more control.<br>><br>> Wilton<br>><br>>>and bug fixes than any commercial product sold in the intra-industrial<br>
>> channel<br>><br>> ... and they won't charge you a $30,000 license fee for the upgrade.<br><br></div></div>
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