Perhaps that's a good question for the Asterisk user group. This is the SIPp user group.
> -----Original Message----- > From: Tincho ylm [mailto:sad...@gmail.com] > Sent: Wednesday, August 12, 2009 10:25 AM > To: Dmitry Goncharov > Cc: sipp-users@lists.sourceforge.net > 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 <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 > > > > -------------------------------------------------------------- > ---------------- > 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 > Sipp-users@lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/sipp-users > ------------------------------------------------------------------------------ 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 Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
