I've done plenty of testing of IVR systems, but have never used SIPp to
do it.  It doesn't seem like the right tool for the job, in my opinion,
especially for anything but the most trivially simple IVR flows.
Timing-based navigation of an IVR flow is fraught with peril.  To do it
right, you need some way of waiting for a prompt and then entering DTMF
digits.
 
By coincidence, I just got an email in my Inbox announcing this seminar
on IVR testing:
 
https://event.on24.com/eventRegistration/EventLobbyServlet?target=regist
ration.jsp&eventid=158288&sessionid=1&key=78FE635632E3BF9513D8033081764D
DF&partnerref=SMWS&sourcepage=register
 
I have no idea who Cognizant is, or what they're trying to sell.  But
it's free information.
 
 
Cheers,
Todd.
 


________________________________

        From: Patrick Collins [mailto:normalbl...@gmail.com] 
        Sent: Saturday, October 17, 2009 8:58 AM
        To: Bradley, Todd
        Cc: sipp-users@lists.sourceforge.net
        Subject: Re: [Sipp-users] IVR Testing
        
        
        The call duration is quite short ~ 30 seconds, in any case I
will sort the maths. I was after some assistance with the the Sipp
configuration.

        Thanks.

        Patrick Collins
        +61 419 712 581
        
        
        
        2009/10/18 Bradley, Todd <todd.brad...@polycom.com>
        

                So when you start to ramp down you're going to have
something like 30,000 PSTN calls up?  Where do you plan to terminate all
these calls?  In other words, who's answering?  And you've already got a
service plan with Flowroute that will allow you to do that many calls?
                
                
                Cheers,
                Todd.

                



                -----Original Message-----
                From: Patrick Collins [mailto:normalbl...@gmail.com]
                Sent: Fri 10/16/2009 8:32 PM
                To: sipp-users@lists.sourceforge.net
                Subject: [Sipp-users] IVR Testing
                
                Hello,
                Can anyone please give me some hints on how to use Sipp
to do load testing
                on our IVR system. What I need to do is dial the IVR via
a PSTN number wait
                for it to play it's voice prompt and then send a DTMF 9,
wait some more and
                then hangup.
                
                I have successfully installed Sipp on FreeBSD 7.2 (a
virtual provided by
                RootBSD) and have created an account with Flowroute. I
want to use Sipp
                running on the FreeBSD machine to create calls to the
PSTN via Flowroute. I
                need to ramp up the call rate from 0 to 100 calls per
second over 50 seconds
                and then sustain it for 5 minutes and then ramp down
back to 0.
                
                I don't expect a a blow by blow on how to do this but
would appreciate any
                pointers you might have.
                
                Thanks.
                
                Patrick
                
                Patrick Collins
                +61 419 712 581
                
                

                


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