That would probably be me. You could use a lot of different things to do the testing, one would be the tcl script in your asterisk/contrib/scripts directory, some more can be found in the beginning of this presentation: http://astertest.com/astricon_performance.ppt
We started some callgenerator for asterisk a very long time ago. ( I have to admit its far from ready and contains many bugs). A howto for this tool can be found at : http://www.asteriskguru.com/tutorials/astertest.html If you want to use sipp, be sure to use playback on your asterisk server and not app_milliwatt, meetme or echo. (those applications will not send any rtp if nothing was received). SIPP only does echoing Zoa ----------- Asteriskguru tutorials: http://www.asteriskguru.com/tutorials/ Visit ClueCon - the asterisk developpers conference: http://www.cluecon.com/ - Dates: August 3, 4 and 5, 2005 Best Western Chicago West Matthew Boehm wrote:
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 doesn't seem to work right. I also setup a fake number in asterisk that when called by sipp, would dial another number via PRI, hoping that some 729 conversion would occur. Nothing. I was able to pump 10 simul calls that went this path: sipp -> asterisk -> pri -> telco ->pri ->asterisk ..and still no 729 usage or any other discernable load on the server. Can anyone offer suggestion on how to really simulate calls (using sipp or other tester) to asterisk to verify its ability to process X calls? I know someone out there has done this, but forget who it was. Thanks, Matthew _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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