[asterisk-users] concurrent channels limit

[Syco]

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 at 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 at 192.168.200.64:5060>;tag=as4ad7b2e8^M
    To: sipp <sip:sipp at 192.168.200.185:52281>;tag=2001SIPpTag0015^M
    Call-ID: 15-2001 at 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 at 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 at 192.168.200.64:5060>;tag=as4ad7b2e8^M
    To: sipp <sip:sipp at 192.168.200.185:52281>;tag=2001SIPpTag0015^M
    Call-ID: 15-2001 at 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 at 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 at 192.168.200.64:5060>;tag=as43adc103^M
    To: sipp <sip:sipp at 192.168.200.185:38844>;tag=2001SIPpTag009^M
    Call-ID: 9-2001 at 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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