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 use
On 29 Jun 2005, at 04:51, 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
On 29 Jun 2005, at 04:51, 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
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
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 asteris
