I have a question here.<div><br></div><div>How can we test the quality of voice upon increasing the call load?</div><div><br></div><div>Can we try passing a voice file using sipp and record the same in dial plan record application ? Is this reliable enough to simulate near real world scenario?</div>
<div><br></div><div>Mitul<span></span><br><br>On Wednesday, May 22, 2013, Tommy Cooper wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div style="font-size:12pt;font-family:times new roman,new york,times,serif">
<div><span>Thank you for your help I finally solved this issue. Is it possible that my setup can achieve 212 concurrent calls, I am running Asterisk on just 1 core using 3.5 GHz, and 1Gb of RAM?<var></var><br></span></div>
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<div dir="ltr">----- Forwarded Message -----<br><font face="Arial"><b><span style="FONT-WEIGHT:bold">From:</span></b> Marie Fischer <<a href="javascript:_e({}, 'cvml', 'marie@vtl.ee');" target="_blank">marie@vtl.ee</a>><br>
<b><span style="FONT-WEIGHT:bold">To:</span></b> Asterisk Users Mailing List - Non-Commercial Discussion <<a href="javascript:_e({}, 'cvml', 'asterisk-users@lists.digium.com');" target="_blank">asterisk-users@lists.digium.com</a>> <br>
<b><span style="FONT-WEIGHT:bold">Sent:</span></b> Wednesday, May 22, 2013 1:16 PM<br><b><span style="FONT-WEIGHT:bold">Subject:</span></b> Re: [asterisk-users] Stress testing Asterisk<br></font></div>
<div><br><br>On 21.05.2013, at 0:05, Tommy Cooper <<a href="javascript:_e({}, 'cvml', 'tomcooper83@yahoo.com');" target="_blank">tomcooper83@yahoo.com</a>> wrote:<br><br>> Hi,<br>> I just installed Sipp 3.3 on CentOS 6.3 and all of the calls Sipp is generating are failing. I am trying to run Sipp on the same machine as Asterisk PBX using the ./sipp -sn uac 192.168.1.115 command.<br>
<br>Do you have a peer and extension configured for SIPP in your Asterisk configuration? You also needat least the -s <extension_to_dial> option on your sipp command line.<br><a href="http://hasnainali.wordpress.com/2009/03/12/using-sipp-for-stress-testing-asterisk/" target="_blank">http://hasnainali.wordpress.com/2009/03/12/using-sipp-for-stress-testing-asterisk/</a>has some simple instructions which should get you started.<br>
If the calls still fail, Asterisk console output would be
helpful.<br><br><br><br>--<br>_____________________________________________________________________<br>-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com/" target="_blank">http://www.api-digital.com/</a>--<br>
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Mitul Limbani,<br>Chief Architech & Founder,<br>Enterux Solutions Pvt. Ltd.<br>110 Reena Complex, Opp. Nathani Steel, <br>Vidyavihar (W), Mumbai - 400 086. India<br><a href="http://www.enterux.com/" target="_blank">http://www.enterux.com/</a><br>
<a href="http://www.entvoice.com/" target="_blank">http://www.entvoice.com/</a><br>email: <a href="mailto:mitul@enterux.in" target="_blank">mitul@enterux.in</a><br>DID: +91-22-71967121<br>Cell: +91-9820332422<br><br><br>