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?
[EMAIL PROTECTED] 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!
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