Re: concurrent channels limit

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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.
This calls just answer, play a file the first 2 seconds and then wait.
Then sipp close because of two many errors, this is the log:
sipp: The following events occured:
2012-03-30>-----15:17:07:081>---1333117027.081757: Discarding message which can't be mapped to a known SIPp call:
BYE sip:sipp@192.168.200.185:52281 SIP/2.0^M
Via: SIP/2.0/UDP 192.168.200.64:5060;branch=z9hG4bK2a711f73;rport^M
Max-Forwards: 70^M
From: sut <sip:17000@192.168.200.64:5060>;tag=as4ad7b2e8^M
To: sipp <sip:sipp@192.168.200.185:52281>;tag=2001SIPpTag0015^M
Call-ID: 15-2001@192.168.200.185^M
CSeq: 102 BYE^M
User-Agent: Asterisk PBX 1.8.11.0^M
X-Asterisk-HangupCause: Normal Clearing^M
X-Asterisk-HangupCauseCode: 16^M
Content-Length: 0^M
^M
.
2012-03-30>-----15:17:07:580>---1333117027.580847: Discarding message which can't be mapped to a known SIPp call:
BYE sip:sipp@192.168.200.185:52281 SIP/2.0^M
Via: SIP/2.0/UDP 192.168.200.64:5060;branch=z9hG4bK2a711f73;rport^M
Max-Forwards: 70^M
From: sut <sip:17000@192.168.200.64:5060>;tag=as4ad7b2e8^M
To: sipp <sip:sipp@192.168.200.185:52281>;tag=2001SIPpTag0015^M
Call-ID: 15-2001@192.168.200.185^M
CSeq: 102 BYE^M
User-Agent: Asterisk PBX 1.8.11.0^M
X-Asterisk-HangupCause: Normal Clearing^M
X-Asterisk-HangupCauseCode: 16^M
Content-Length: 0^M
^M
.
2012-03-30>-----15:17:07:982>---1333117027.982422: Discarding message which can't be mapped to a known SIPp call:
BYE sip:sipp@192.168.200.185:38844 SIP/2.0^M
Via: SIP/2.0/UDP 192.168.200.64:5060;branch=z9hG4bK66a86c70;rport^M
Max-Forwards: 70^M
From: sut <sip:17000@192.168.200.64:5060>;tag=as43adc103^M
To: sipp <sip:sipp@192.168.200.185:38844>;tag=2001SIPpTag009^M
Call-ID: 9-2001@192.168.200.185^M
CSeq: 102 BYE^M
User-Agent: Asterisk PBX 1.8.11.0^M
X-Asterisk-HangupCause: Normal Clearing^M
X-Asterisk-HangupCauseCode: 16^M
Content-Length: 0^M
^M
.
2012-03-30>-----15:17:08:504>---1333117028.504334: Unable to get a UDP socket (3).

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.




On 30/03/2012 14:50, Danny Nicholas wrote:

Change –r 100 to –r 300.

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