As far as I know:<br><br>You need to compile sipp with media streaming and authentication or if you just want first to test you may provide an extension named "service" in the context defined in general section of your sip conf for external calls coming to your asterisk server without authentication:
<br><br><a href="http://sipp.sourceforge.net/doc/reference.html#Installing+SIPp">http://sipp.sourceforge.net/doc/reference.html#Installing+SIPp</a><br><ul><li>
<strong>With <a href="http://sipp.sourceforge.net/doc/reference.html#pcapplay">PCAP play</a> and without <a href="http://sipp.sourceforge.net/doc/reference.html#authentication">authentication</a> support</strong>:
<pre class="code"># gunzip sipp-xxx.tar.gz<br># tar -xvf sipp-xxx.tar<br># cd sipp<br># make pcapplay</pre></li></ul><br><ul><li>
<strong>With <a href="http://sipp.sourceforge.net/doc/reference.html#pcapplay">PCAP play</a> and <a href="http://sipp.sourceforge.net/doc/reference.html#authentication">authentication</a> support</strong>:
<pre class="code"># gunzip sipp-xxx.tar.gz<br># tar -xvf sipp-xxx.tar<br># cd sipp<br># make pcapplay_ossl<br></pre>
</li></ul>Example:<br><ul><li>Sipp being used as a SIP user agent Client:</li><ul><li> Call Duration 10000ms</li><li> Dialing Calls with RTP using ulaw </li></ul></ul><br> ./sipp -sf uac_pcap.xml -d 10000 <a href="http://192.168.34.6">
192.168.34.6</a> -trace_err<br><br>Where this IP is my * .<br><br>Hope this helps,<br><br>Plse provid some feedback.<br><br>I would like also to learn from community how to understand Load average results with Top command while incrementing calls dial from sipp to asterisk, and how to determine max calls on Asterisk. This max calls is defined when Sipp calls to * starts being discarded?
<br><br>Best regards,<br>Marco Mouta<br><br><div><span class="gmail_quote">On 1/23/07, <b class="gmail_sendername">Julian Lyndon-Smith</b> <<a href="mailto:asterisk@dotr.com">asterisk@dotr.com</a>> wrote:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
We are in the process of implementing realtime voicemail. I was wanting<br>to "stress-test" the system to see if or when it would fall over.<br><br>Is it possible to use sipp to create say 250 calls, each of which leaves
<br>a message in the voicemail ?<br><br>My dialplan is currently<br><br>[default]<br><br>exten => stress,1,Answer()<br>exten => stress,2(vm),Voicemail(7777|su)<br>exten => stress,3,Hangup()<br><br>however, if I use sipp to test this, I get
<br><br>[Jan 23 14:43:51] WARNING[22782]: app.c:599 __ast_play_and_record: No<br>audio available on SIP/sipp-b7c274b0??<br><br>I suspect that's because sipp itself is not sending audio.<br><br>Is there any tricks I can do in the dialplan to get an extension to
<br>answer sipp and then send it to voicemail, but play some audio for the<br>voicemail ?<br><br>Thanks.<br><br>Julian.<br>_______________________________________________<br>--Bandwidth and Colocation provided by <a href="http://Easynews.com">
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</a><br></blockquote></div><br>