Hi List,<div><br></div><div>I'm trying to get working SIPp with media but something is wrong (it's working well without media), please help:</div><div><br></div><div>This is the command I send at SIPp server: </div>
<div><span class="Apple-style-span" style="font-family: Arial; font-size: 13px; "> ./sipp -sn uac_pcap -d
5000 -s 2006 192.168.1.18 -l 20 -trace_err</span></div><div><span class="Apple-style-span" style="font-family: Arial; font-size: 13px; "><br></span></div><div><span class="Apple-style-span" style="font-family: Arial; font-size: 13px; ">This is the result I see:</span></div>
<div><font class="Apple-style-span" face="Arial"><div> Last Error: Aborting call on unexpected message for Call-Id '19-12768@12...</div><div><br></div><div>What I see at logs:</div><div><br></div><div><div>2011-06-28 14:32:57:624 1309289577.624809: Aborting call on unexpected message for Call-Id '<a href="mailto:1-12768@127.0.0.1">1-12768@127.0.0.1</a>': while expecting '100' (index 1), received 'SIP/2.0 488 Not acceptable here^M</div>
<div>Via: SIP/2.0/UDP 127.0.0.1:5061;branch=z9hG4bK-12768-1-0;received=192.168.25.253^M</div><div>From: sipp <<a href="http://sip:sipp@127.0.0.1:5061">sip:sipp@127.0.0.1:5061</a>>;tag=12768SIPpTag091^M</div><div>To: sut <<a href="http://sip:2005@192.168.1.18:5060">sip:2005@192.168.1.18:5060</a>>;tag=as3614adc3^M</div>
<div>Call-ID: <a href="mailto:1-12768@127.0.0.1">1-12768@127.0.0.1</a>^M</div><div>CSeq: 1 INVITE^M</div><div>Server: Asterisk PBX 1.8.4.1^M</div><div>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH^M</div>
<div>Supported: replaces, timer^M</div><div>Content-Length: 0^M</div></div><div><br></div></font><span class="Apple-style-span" style="font-family: Arial; ">This is my asterisk 1.8's configuration:</span><font class="Apple-style-span" face="Arial"><div>
<div><b>sip.conf</b></div><div>[sipp]</div><div>type=friend</div><div>context=sipp</div><div>host=dynamic</div><div>port=6000</div><div>user=sipp</div><div>canreinvite=no</div><div>disallow=all</div><div>allow=ulaw</div></div>
<div><b><br></b></div><div><b>extensions.conf:</b></div><div><div>[sipp]</div><div>exten => 2005,1,Answer</div><div>same=>n,Dial(SIP/intern,30)</div></div></font><span class="Apple-style-span" style="font-family: Arial; ">same=>n,Hangup()</span><font class="Apple-style-span" face="Arial"><div>
<div><br></div><div>exten => 2006,1,Answer()</div><div>same=> n,WaitMusicOnHold(4)</div><div>same=> n,Hangup()</div></div><div><br></div></font><div class="gmail_quote"><br></div><div class="gmail_quote">I'm using sipp.3.1.src.tar.gz and I have installed it this way:</div>
<div class="gmail_quote">..sip.svn# make pcapplay</div><div class="gmail_quote"><br></div><div class="gmail_quote">Thanks in advance.</div><div class="gmail_quote"><br></div><div class="gmail_quote">Elder</div><div class="gmail_quote">
On Thu, May 12, 2011 at 2:51 PM, Steve Totaro <span dir="ltr"><<a href="mailto:stotaro@asteriskhelpdesk.com">stotaro@asteriskhelpdesk.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
<a href="http://tinyurl.com/3hx5652" target="_blank">http://tinyurl.com/3hx5652</a><br><br><div class="gmail_quote"><div><div></div><div class="h5">On Thu, May 12, 2011 at 11:52 AM, Daniel - Asterisk <span dir="ltr"><<a href="mailto:earohuanca@gmail.com" target="_blank">earohuanca@gmail.com</a>></span> wrote:<br>
</div></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></div><div class="h5">Hello Everyone,<div><br></div><div>I wonder if someone could share a manual about using SIPp for Asterisk's testing.</div>
<div><br></div><div>I'll be gratefull</div><div><br></div><div><br></div><div>Regards,</div>
<div><br></div><div>Elder Arohuanca</div><div>Lima - Peru<div><div></div><div><br><br><div class="gmail_quote">On Tue, Sep 30, 2008 at 12:59 PM, zac wolfe <span dir="ltr"><<a href="mailto:zac.wolfe@gmail.com" target="_blank">zac.wolfe@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"><div dir="ltr">Sipp looks pretty good! I don't know how I missed this one. This would've saved me tons of time a couple months ago.<div>
<br></div><div>I plan on using it to load test using 2 Asterisk servers, one to initiate the SIP calls, the other to receive. Thanks for the tip Alex.</div>
<div><br></div><font color="#888888"><div>Zac Wolfe</div><div>Safi Systems LLC</div><div><a href="http://www.safisystems.com" target="_blank">www.safisystems.com</a></div></font><div><div></div><div><div><br><br>
<div class="gmail_quote">On Sat, Sep 27, 2008 at 5:58 AM, Alex Balashov <span dir="ltr"><<a href="mailto:abalashov@evaristesys.com" target="_blank">abalashov@evaristesys.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">What you are looking for is SIPP: <a href="http://sipp.sourceforge.net/" target="_blank">http://sipp.sourceforge.net/</a><br>
<br>
It won't intrinsically tell you anything about the data; it's up to you<br>
to appropriate the findings. But it accomplishes the generation of<br>
traffic (and dummy media!) on a technical level.<br>
<div><br>
Igor Hernandez wrote:<br>
<br>
> Sam Tam wrote:<br>
>> Hello everyone<br>
>><br>
>><br>
>><br>
>> I am trying to look for a free test call generator that will get me some<br>
>> stats like PDD, ASR and call quality etc on each route. As well as do<br>
>> test at every interval too<br>
>><br>
>><br>
>> If you know something like this please enlighten me.<br>
>><br>
>> Sam<br>
>><br>
>><br>
>> ------------------------------------------------------------------------<br>
>><br>
>> _______________________________________________<br>
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><br>
> Hey Sam,<br>
><br>
> I've been looking for such a tool also. I can't seem to find a tool that<br>
> does those things.<br>
><br>
> If nothing comes up in the next couple of weeks I'm going to code<br>
> something up, I wouldn't mind letting you and anyone else who might be<br>
> interested have the source once its done.<br>
><br>
> Let me know if you find anything thats already out there in the<br>
> meantime, might just save me a few hours of work.<br>
><br>
> Regards,<br>
><br>
><br>
<br>
<br>
--<br>
</div><font color="#888888">Alex Balashov<br>
Evariste Systems<br>
Web : <a href="http://www.evaristesys.com/" target="_blank">http://www.evaristesys.com/</a><br>
Tel : (+1) (678) 954-0670<br>
Direct : (+1) (678) 954-0671<br>
Mobile : (+1) (706) 338-8599<br>
</font><div><div></div><div><br>
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