[Asterisk-Users] Anyone using SipP to produce RTP load?

[Matthew Boehm]

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