<br><br><div class="gmail_quote">On Tue, Oct 5, 2010 at 1:02 PM, Danny Dias <span dir="ltr"><<a href="mailto:ing.diasdanny@gmail.com">ing.diasdanny@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
Hello my friend Ingmar,<div><br></div><div>I would like to know the cable you used? how was the connection? i'm using this one:</div><div><br></div><div><a href="http://wiki.sangoma.com/Pinouts#A108" target="_blank">http://wiki.sangoma.com/Pinouts#A108</a> Loop Back</div>
<div><br></div><div>Is this ok? what should i do my friend, my problems are "understand" the fisicall connection :(</div><div><br></div><div>Best Regards!!!<br><br><div class="gmail_quote">2010/9/24 Ingmar Steen <span dir="ltr"><<a href="mailto:i.steen@teleknowledge.nl" target="_blank">i.steen@teleknowledge.nl</a>></span><br>
<blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
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<p class="MsoNormal"><span style="font-size: 11pt;">Hi
DD,</span></p>
<p class="MsoNormal"><span style="font-size: 11pt;"> </span></p>
<p class="MsoNormal"><span style="font-size: 11pt;" lang="EN-US">We
usually use loopback cables and use the open source SIP test tool “SIPp”
to initiate SIP calls that are sent from one group of 4 ports to another group
of 4 ports.</span></p>
<p class="MsoNormal"><span style="font-size: 11pt;" lang="EN-US"> </span></p>
<p class="MsoNormal"><span style="font-size: 11pt;">Met
vriendelijke groet,</span></p>
<p class="MsoNormal"><span style="font-size: 11pt;">Ingmar
Steen</span></p>
<p class="MsoNormal"><span style="font-size: 11pt;">Teleknowledge</span></p>
<p class="MsoNormal"><span style="font-size: 11pt;"> </span></p>
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<p class="MsoNormal"><b><span style="font-size: 10pt;">Van:</span></b><span style="font-size: 10pt;">
<a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank">asterisk-users-bounces@lists.digium.com</a>
[mailto:<a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank">asterisk-users-bounces@lists.digium.com</a>] <b>Namens </b>Danny Dias<br>
<b>Verzonden:</b> vrijdag 24 september 2010 11:05<br>
<b>Aan:</b> Asterisk Users Mailing List - Non-Commercial Discussion<br>
<b>Onderwerp:</b> [asterisk-users] How to test BIG traffic through
DAHDI/WANPIPEinterfaces</span></p>
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<p class="MsoNormal">Hello Community,</p>
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<p class="MsoNormal">I need to test or simulate many calls through dahdi/wanpipe,
i have a Sangoma A108D, and i need to test the stability of the
card/drivers/firmwares with a test environment, do you think is possible?</p>
</div>
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<p class="MsoNormal"> </p>
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<div>
<p class="MsoNormal">What should i do? using some loopback cable maybe?</p>
</div>
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<p class="MsoNormal"> </p>
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<p class="MsoNormal">Thanks in advance</p>
</div>
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<p class="MsoNormal"> </p>
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<p class="MsoNormal">DD </p></div></div></div></div></div></blockquote></div></div></blockquote><div><br>I set up two machines with T1 interfaces, and connected the two with an appropriate t1 cable.<br>One was acting as a network (master), the other as a subscriber (slave) (for timing). wrote two dialplans, one for each machine,<br>
that would answer an incoming call on one dahdi line, and call to the next numbered line on the other<br>machine. The other machine was similarly outfit. I'd define the extension for the first line on the t1,<br>and call it with any phone you desire. That call will cascade into 23 separate interlinked calls. If you are<br>
clever, the last call in should dial another real phone you have on-hand.<br><br>You get the picture... right? Phone A dials the exten to call the first exten on the other machine. The<br>dialplan should use the first channel on the t1 to place a call to the first exten on the other machine.<br>
On the other machine, the incoming call on channel 1 is answered, and then a dial to the second extension<br>on the first machine, over the 2nd t1 channel. The first machine answers, and uses the 3rd channel<br>to call the other machine.... and so on till all channels are being used. The last exten answers and dials<br>
a phone (dahdi or SIP, no matter) that you pick up. Such a looped call should probably be awful, but<br>it's going thru 23 t1 channels!<br><br>If you have two t1 interaces in a single card (or two cards), then you do this on one machine.<br>
<br>Another approach: set up equal numbers of FZO and FXS lines, and similarly loop s single call thru all the<br>channels.This would require just one machine.<br><br>Other approaches would involve running multiple threads to call an extension and then hang up and<br>
repeating this over and over again on all channels to ascertain the load placed just by call setup and tear-down.<br>This kind of load is different than when all lines are just shoveling data back and forth.<br><br>Watch your load averages, your %cpu, your swap, etc, as the tests are in full swing.<br>
<br>murf<br><br><br><br> <br></div><blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;"><div><div class="gmail_quote"><blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
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