[asterisk-dev] Spring is here :: Test
the Asterisk Spring Collection 2006!
Kristian Kielhofner
kris at krisk.org
Mon Apr 10 13:07:25 MST 2006
Olle E Johansson wrote:
> Oh, I have no perfect configuration... The SIP accounts are minimal,
> just names, passwords, contexts and IP address.
>
>
> This is the interesting part on both sides:
>
> [enterprise]
> exten => _X.,1,set(number=$[${EXTEN} + 1])
> exten => _X.,n,verbose
> (-----------------------------------------------------------------------
> ------- ${EXTEN} ------------)
> exten => _X.,n,gotoif($[${number} = 250]?ready,1)
> exten => _X.,n,dial(SIP/enterprise/${number})
> exten => _X.,n,hangup
> exten => ready,1,answer
> exten => ready,n,wait(1)
> exten => ready,n,playback(tt-monkeys)
> exten => ready,n,wait(1)
> exten => ready,n,hangup
>
> I start the call by calling 10 at enterprise from the console and then
> wait until I either hear monkeys
> or see a total mess on the screen.
>
> We should be able to create a svn repository with various test
> configurations, that are fixed.
> - One SIP stress test
> - One IAX2 stress test
> - One Realtime stress test
> - Codec conversion tests
>
> Good idea.
> /O
Olle,
I see. One of my scenarios uses a bash script to generate calls like so:
if [ ! $1 ]
then
echo "Usage:
$0 account calls delay
"
exit 1
fi
for i in `seq 1 $2`
do
echo "Channel: Local/$1 at call
MaxRetries: 1
RetryTime: 60
WaitTime: 30
Context: loadtest
Extension: $1
Priority: 1" > /tmp/scratch/$i.call && mv \
/tmp/scratch/*.call /tmp/scratch/spool/
sleep $3
done
I'll find the related Asterisk config portions, but it allows me to
test the number of calls (with a limit) as well as "calls per second".
It doesn't have perfect resolution, but it has been accurate enough for
my tests.
/tmp/scratch is mounted as tmpfs, and /var/spool/asterisk/outgoing is
symlinked to "/tmp/scratch/spool".
I also try to record the audio stream to /tmp/scratch with Monitor or
Record. I have been able to get Record to work, but Monitor only writes
out 160 bytes before stopping (while the RTP continues to flow). Oh
well, another issue, another thread.
--
Kristian Kielhofner
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