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Ok, this was a stupid thing (my fault), with -r 1000 I get easily
1000 concurrent calls that terminate in 20 seconds.<br>
This calls just answer, play a file the first 2 seconds and then
wait.<br>
Then sipp close because of two many errors, this is the log:<br>
<blockquote>sipp: The following events occured:<br>
2012-03-30>-----15:17:07:081>---1333117027.081757:
Discarding message which can't be mapped to a known SIPp call:<br>
BYE <a class="moz-txt-link-freetext" href="sip:sipp@192.168.200.185:52281">sip:sipp@192.168.200.185:52281</a> SIP/2.0^M<br>
Via: SIP/2.0/UDP
192.168.200.64:5060;branch=z9hG4bK2a711f73;rport^M<br>
Max-Forwards: 70^M<br>
From: sut <a class="moz-txt-link-rfc2396E" href="sip:17000@192.168.200.64:5060"><sip:17000@192.168.200.64:5060></a>;tag=as4ad7b2e8^M<br>
To: sipp
<a class="moz-txt-link-rfc2396E" href="sip:sipp@192.168.200.185:52281"><sip:sipp@192.168.200.185:52281></a>;tag=2001SIPpTag0015^M<br>
Call-ID: <a class="moz-txt-link-abbreviated" href="mailto:15-2001@192.168.200.185^M">15-2001@192.168.200.185^M</a><br>
CSeq: 102 BYE^M<br>
User-Agent: Asterisk PBX 1.8.11.0^M<br>
X-Asterisk-HangupCause: Normal Clearing^M<br>
X-Asterisk-HangupCauseCode: 16^M<br>
Content-Length: 0^M<br>
^M<br>
.<br>
2012-03-30>-----15:17:07:580>---1333117027.580847:
Discarding message which can't be mapped to a known SIPp call:<br>
BYE <a class="moz-txt-link-freetext" href="sip:sipp@192.168.200.185:52281">sip:sipp@192.168.200.185:52281</a> SIP/2.0^M<br>
Via: SIP/2.0/UDP
192.168.200.64:5060;branch=z9hG4bK2a711f73;rport^M<br>
Max-Forwards: 70^M<br>
From: sut <a class="moz-txt-link-rfc2396E" href="sip:17000@192.168.200.64:5060"><sip:17000@192.168.200.64:5060></a>;tag=as4ad7b2e8^M<br>
To: sipp
<a class="moz-txt-link-rfc2396E" href="sip:sipp@192.168.200.185:52281"><sip:sipp@192.168.200.185:52281></a>;tag=2001SIPpTag0015^M<br>
Call-ID: <a class="moz-txt-link-abbreviated" href="mailto:15-2001@192.168.200.185^M">15-2001@192.168.200.185^M</a><br>
CSeq: 102 BYE^M<br>
User-Agent: Asterisk PBX 1.8.11.0^M<br>
X-Asterisk-HangupCause: Normal Clearing^M<br>
X-Asterisk-HangupCauseCode: 16^M<br>
Content-Length: 0^M<br>
^M<br>
.<br>
2012-03-30>-----15:17:07:982>---1333117027.982422:
Discarding message which can't be mapped to a known SIPp call:<br>
BYE <a class="moz-txt-link-freetext" href="sip:sipp@192.168.200.185:38844">sip:sipp@192.168.200.185:38844</a> SIP/2.0^M<br>
Via: SIP/2.0/UDP
192.168.200.64:5060;branch=z9hG4bK66a86c70;rport^M<br>
Max-Forwards: 70^M<br>
From: sut <a class="moz-txt-link-rfc2396E" href="sip:17000@192.168.200.64:5060"><sip:17000@192.168.200.64:5060></a>;tag=as43adc103^M<br>
To: sipp
<a class="moz-txt-link-rfc2396E" href="sip:sipp@192.168.200.185:38844"><sip:sipp@192.168.200.185:38844></a>;tag=2001SIPpTag009^M<br>
Call-ID: <a class="moz-txt-link-abbreviated" href="mailto:9-2001@192.168.200.185^M">9-2001@192.168.200.185^M</a><br>
CSeq: 102 BYE^M<br>
User-Agent: Asterisk PBX 1.8.11.0^M<br>
X-Asterisk-HangupCause: Normal Clearing^M<br>
X-Asterisk-HangupCauseCode: 16^M<br>
Content-Length: 0^M<br>
^M<br>
.<br>
2012-03-30>-----15:17:08:504>---1333117028.504334: Unable to
get a UDP socket (3).<br>
</blockquote>
<br>
But if I change the dialplan, remove background and wait functions,
add play with a g729 audio file instead, I could do again just 80
concurrent call.<br>
<br>
<br>
<br>
<br>
On 30/03/2012 14:50, Danny Nicholas wrote:
<blockquote cite="mid:005b01cd0e7c$01c0d280$05427780$@debsinc.com"
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<p class="MsoNormal"><span
style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D">Change
–r 100 to –r 300.</span><o:p></o:p></p>
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