Yes, Bradley, thank you all! 2009/8/12 Bradley, Todd <[email protected]>: > Perhaps that's a good question for the Asterisk user group. This is the > SIPp user group. > > >> -----Original Message----- >> From: Tincho ylm [mailto:[email protected]] >> Sent: Wednesday, August 12, 2009 10:25 AM >> To: Dmitry Goncharov >> Cc: [email protected] >> Subject: Re: [Sipp-users] Only 12 simultaneous calls >> >> 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 <[email protected]>: >> > >> > >> > 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 >> > >> >> -------------------------------------------------------------- >> ---------------- >> Let Crystal Reports handle the reporting - Free Crystal >> Reports 2008 30-Day trial. Simplify your report design, >> integration and deployment - and focus on what you do best, >> core application coding. Discover what's new with Crystal >> Reports now. http://p.sf.net/sfu/bobj-july >> _______________________________________________ >> Sipp-users mailing list >> [email protected] >> https://lists.sourceforge.net/lists/listinfo/sipp-users >> >
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