Thanks for responding! You guys are one of my favorite companies to deal with in terms of support. I can tell you really want to have a great product.
We were having the issue between 20-50 calls. Had 500, 700, and 1,200 sip phones connected at various points. Some sites (maybe 10-20 sites) had 20 phones with 20 BLF’s all watching one another along with ~2 parks. Most of our phones are set to 300 second registers.
When we first tried we converted 1,200 phones (“converted” a bit less since we had preexisting PJSIP extensions but you get the idea) and had 40-70 active calls soon as it hit 9am.
I can say based on symptoms:
- Was the hold music choppy and horrible using the default freepbx hold music
- Was the IVR and recording choppy to the point where you could miss words or options, or perhaps the entire recording
- Were calls choppy where you miss parts of words or multiple words?
If all of those were happening then it was happening to us. The very first symptom is hold music and recordings getting progressively worse. Call quality issues happen once hold music and recordings get to a certain point.
We were running a live machine with most Freepbx features enabled and users constantly transferring, putting people on hold, and doing what they do. Users are always experts at stress testing
We split our 1,200 phone server between 3 servers. One had 500 phones and the full call flow of the 1,200 server. It would work ok with under 20 calls but I noticed when it got to 20-30 we could hear MOH and recording quality issues. We gave up and had to fix the issue asap so we converted to chan_SIP.