[Asterisk-Users] Asterisk and Sipp

[jean-marie.goupil at telintrans.fr]

Hi there!

Does anyone knows how to test Asterisk load with sipp? I am using uac.xml
to call a 'playback extensions' via a SIP channel. When I increase the Call
rate (about 20cps), I begin to have INVITE/200/BYE retransmissions
meanwhile the RedHat box is not loaded at all (made a TOP). Where is the
pb?

[root at 10.54.196.38 sipp]# sipp 10.54.196.32 -s 9001 -sf uac.xml -d 100 -i
10.54.196.38 -r 20

                        Messages  Retrans
INVITE ---------->         71        52
100    <----------         70        0
200    <---------- E-RTD   69        44
BYE    ---------->         69        51
200    <----------         69        0

Thanx!