Yes! you're right!

But, what can be the error on Asterisk? some kind of call limitation?
I have installed Freepbx, may be that is the problem.

Anyone knows?

thanks!

2009/8/12 Dmitry Goncharov <dgoncha...@unison.com>:
>
>
> Tincho ylm wrote:
>
> Hi all!
>
> My SIPp only allow 12 simultaneous calls. If a use -l 10 everything
> work perfect!
>
> If I put -l 25, I get this error at 13th call:
>
> Aborting call on unexpected message for Call-Id '92-4...@ip-uac':
> while expecting '100' (index 1), received 'SIP/2.0 500 Server internal
> error
> Via: SIP/2.0/UDP IP-UAC:5061;branch=z9hG4bK-4219-92-0;received=IP-UAC
> From: sipp <sip:s...@ip-uac:5061>;tag=92
> To: sut <sip:1...@ip-asterisk:5060>;tag=as210e1c01
> Call-ID: 92-4...@ip-uac
> CSeq: 1 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Content-Length: 0
>
> I'm using this commands:
>
> UAC:
> ./sipp -sf uac_pcap.xml IP-Asterisk -m 100 -s 1234 -l 25 -mp 9000 -trace_err
> (The XML is by default - AUC with Media)
>
> UAS:
> sipp -sf uas.xml -mi Local-IP -rtp_echo -mp 11000 -p 5070
>
> ASTERISK:
>
> sip.conf
>
> [sipp]
> type=friend
> context=sipp-test
> dtmfmode=rfc2833
> host=IP-UAC
> canreinvite=no
> disallow=all
> allow=g729
> allow=alaw
> allow=ulaw
> port=5060
> nat=yes
>
> [sipp_uas]
> type=friend
> context=sipp-test
> dtmfmode=rfc2833
> host=IP-UAS
> canreinvite=no
> disallow=all
> allow=g729
> allow=alaw
> allow=ulaw
> port=5070
> nat=yes
>
> extensions.conf
>
> [sipp-test]
> exten => 1234,1,Dial(SIP/sipp_uas,100,Tt)
> exten => 1234,n,Hangup
>
> exten => _X.,1,NoOp()
> exten => _X.,n,Answer()
> exten => _X.,n,Playback(demo-instruct)
> exten => _X.,n,Playback(demo-instruct)
> exten => _X.,n,Playback(demo-instruct)
> exten => _X.,n,Playback(demo-instruct)
> exten => _X.,n,Hangup()
>
> Why is this?
> Thanks all!
>
>
>
> You are overloading your sip server
>
> BR, Dmitry
>

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