take a look on Originate command for Asterisk manager interface to get web page generating calls between the two boxes.<br><br>Easier I believe is to use SIPp to be used as an UAC that starts dialing to your box1 and in the dialplan of this box make a dial for a Zap channel on Box2.
<br><br><br>You need to compile <span id="st" name="st" class="st">sipp</span>
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" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">http://<span id="st" name="st" class="st">sipp</span>.sourceforge.net/doc/reference.html#Installing+
<span id="st" name="st" class="st">SIPp</span></a><br><ul><li>
<strong>With <a href="http://sipp.sourceforge.net/doc/reference.html#pcapplay" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">PCAP play</a> and without <a href="http://sipp.sourceforge.net/doc/reference.html#authentication" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">
authentication</a> support</strong>:
<pre># gunzip <span id="st" name="st" class="st">sipp</span>-xxx.tar.gz<br># tar -xvf <span id="st" name="st" class="st">sipp</span>-xxx.tar<br># cd <span id="st" name="st" class="st">sipp</span><br># make pcapplay</pre>
</li></ul><br><ul><li>
<strong>With <a href="http://sipp.sourceforge.net/doc/reference.html#pcapplay" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">PCAP play</a> and <a href="http://sipp.sourceforge.net/doc/reference.html#authentication" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">
authentication</a> support</strong>:
<pre># gunzip <span id="st" name="st" class="st">sipp</span>-xxx.tar.gz<br># tar -xvf <span id="st" name="st" class="st">sipp</span>-xxx.tar<br># cd <span id="st" name="st" class="st">sipp</span><br># make pcapplay_ossl<br>
</pre>
</li></ul>Example:<br><ul><li><span id="st" name="st" class="st">Sipp</span> 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> ./<span id="st" name="st" class="st">
sipp</span> -sf uac_pcap.xml -d 10000 <a href="http://192.168.34.6/" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">
192.168.34.6</a> -trace_err<br><br>Where this IP is my * .<br><br><br><br><div><span class="gmail_quote">On 2/1/07, <b class="gmail_sendername">Mik Cheez</b> <<a href="mailto:michael_bulk@wildgate.com">michael_bulk@wildgate.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;">Use auto dial. You can have as many calls as you wish.<br><br><a href="http://www.voip-info.org/wiki-Asterisk+auto-dial+out">
http://www.voip-info.org/wiki-Asterisk+auto-dial+out</a><br><br><br>Porier, Jeremy M. wrote:<br>> Are there any scripts out there that would help me stress test two boxes<br>> that are setup back to back with 4 PRI connections? We're having
<br>> problems with Sangoma cards w/ PCI-e on HP DL385 G2 hardware and I'm<br>> tired of "testing" them in a production environment. As Sangoma<br>> provides firmware updates (and various other shots in the dark) I'd like
<br>> to be able see if the problem is fixed in an isolated environment. I<br>> just need a way to simulate call volume on 4 t1s.<br>><br>> Thanks,<br>> Jeremy Porier<br>> Senior Director of IST<br>> Colorado Christian University
<br>> <a href="mailto:jporier@ccu.edu">jporier@ccu.edu</a><br>> _______________________________________________<br>> --Bandwidth and Colocation provided by <a href="http://Easynews.com">Easynews.com</a> --<br>>
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