Good evening all. So, we need to do a load test on our main SIP routing systems as well as our Conference Bridge, and I thought I’d use the XACT Dialer to do it, but ran into a little snag.
Ok, so, you call the number for the bridge, you’re asked for your moderator code - I used an announcement that had the touch tones in it, then I sent the call to a queue with no music on hold (to provide me a simple method of dropping the call after the queue timer expires).
Part of the issue, I think, was that as we started nailing up more and more calls, the system’s ability to actually play the sound file became a problem. There were several calls where the bridge never actually heard touch tones, yet the first few hundred calls worked fine (and, while those were nailed up, we could call in and join the conference - making me think the box itself wasn’t able to play the file reliably, or wasn’t establishing audio quickly enough, so the tones were missed).
I’ve got 10000 audio ports in the config (the typical 10000-19999), and I increased the processing/ram on the box to 8 processors and 16 gigs of ram. HTOP showed it was running in the 60-80% CPU range, but the web portal was still snappy, so it was still working well - and ram was nowhere near the 16igs (more like around 4-5 gigs).
My thought is, instead of using XACT dialer and these sound files, does anyone skilled in Asterisk scripting know how to create something that:
Set the quantity of the max calls (such as 1000), then have it
call external number (like maybe 10 per second))
sends DMTF (such as 12345678#) - true DTMF instead of a sound file
then keeps channel up with no audio for a timed interval (such as 15 minutes).
drops the call - maybe even saying “goodbye” before it drops.
Any help from the more skilled scripting folks is greatly appreciated.