asterisk users - Mar 2012 - concurrent channels limit

Hi,
I've some problem setting up Asterisk 1.8:
the configuration of the server is
8 core Intel(R) Xeon(R) CPU E5310 @ 1.60GHz
4GB of ram
CentOS release 6.2 (Final)
asterisk-1.8.10.1

I'm using sipp to do some tests, the command I use is "sipp -sf 
/root/uac_pcap_G729.xml -s 17000 192.168.200.64 -l 1000"
the file "uac_pcap_G729.xml" just establishes a call, wait 120 seconds
and hangup.

Finally the problem is: I cannot manage more than 80 concurrent calls.
but I know that is a limit too low, the same machine in the production 
environment, with the same configurations and asterisk 1.4 manage up to 
400 concurrent calls.

How can I find my real upper call limit?

tnx

On 30 Mar 2012, at 10:14, Syco wrote:
> Finally the problem is: I cannot manage more than 80 concurrent calls.
What happens on the 81st call?..

S

Check the sip.conf.sample file.  I think it is the call-limit parameter that
is getting you.  The sample file should tell you what the default is.
Another possibility is that your rtp range is set too low;  the
"normal"
range is 10000-20000, which allows for 2500 calls(or 5000 if you set other
things "correctly").

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Steven Howes
Sent: Friday, March 30, 2012 7:45 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] concurrent channels limit

On 30 Mar 2012, at 10:14, Syco wrote:
> Finally the problem is: I cannot manage more than 80 concurrent calls.
What happens on the 81st call?..

S

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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 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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