On 06/29/05 11:51 Matthew Boehm said the following:
Hey gang,
I've been able to use sipp to produce some call volume on our asterisk
server. The server has no problems handling 50 simul calls. But then again,
no RTP is being done. I tried to use the rtp echo ability of sipp but that
i've used the following sipp command line,
sipp -d 30000 -r 5 -t un -sn uac -l 50 -m 100 -s 20 -mp 10000 <asterisk ip>
which will generate 100 calls of 30 seconds each, limiting it to 50
simultaneous calls at a time to extension 20 on asterisk. extensions 20 was
a simple
exten => 20,1,Answer()
exten => 20,2,Playback(demo-instruct)
exten => 20,3,Goto(1)
this had asterisk send back the Playback output on RTP port 10000 to sipp.
if you wanted to test ulaw<-->g729 conversions between two asterisk
servers, have the above exten lines in the second asterisk server with the
exten lines in the first just being
exten => 20, 1, Dial(IAX2/asterisk2/20)
--
Regards, /\_/\ "All dogs go to heaven."
[EMAIL PROTECTED] (0 0) http://www.alphaque.com/
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