[Asterisk-Users] SIPP load testing - unexpected message - anyone using sipp sucessfully ?

[Robert Rozman]

Hi,

I'd like to test Asterisk performance under more concurrent sip calls. I use
Sipp, but do get "Unexpected message for Call-ID ...", so I wonder if anyone
is using sipp succesfully with Asterisk and is willing to share more info
about his solution ...

Any other convenient way to load test Asterisk ?  Is sipp the right tool ?

Thanks in advance,

regards,

Rob.



sipp: The following events occured:
2005-02-08 00:23:36: Unexpected message for Call-ID
'1.3972.192.168.0.101 at sipp.call.id': while expecting '100' response,
received 'SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.0.101:5060;received=193.77.90.224;rport=5060
From: sipp <sip:sipp at 192.168.0.101:5060>;tag=1
To: sut <sip:service at 193.77.158.104:5060>;tag=as3e7533a6
Call-ID: 1.3972.192.168.0.101 at sipp.call.id
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:service at 193.77.158.104>
Content-Length: 0

' .
2005-02-08 00:23:36: Unexpected message for Call-ID
'2.3972.192.168.0.101 at sipp.call.id': while expecting '100' response,
received 'SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.0.101:5060;received=193.77.90.224;rport=5060
From: sipp <sip:sipp at 192.168.0.101:5060>;tag=2
To: sut <sip:service at 193.77.158.104:5060>;tag=as43cce205
Call-ID: 2.3972.192.168.0.101 at sipp.call.id
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:service at 193.77.158.104>
Content-Length: 0