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20cpu
2004 May 25
0
Asterisk and Sipp
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@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...