I have a fairly new freepbx 14 installation, the phones are on a remote network from the hosted pbx. Generally things are running swimmingly, although we’ve recently just enabled g729 in order to save on bandwidth.
There are some intermittent network issues with the internet provider, causing occasional call quality issues. All the boring stuff yup, a bit of extra packet loss and jitter that we can’t predict or prevent.
So we need to somehow visualize the per-call quality, in some sort of automated manner.
From the asterisk CLI, I can manually “pjsip show channelstats” and see live info on current active calls. Or I can script a “asterisk -rx ‘pjsip show channelstats’” to see everything at a periodic interval.
But what would be much more helpful would be for asterisk to log the per-call stats at the end of each call.
My google fu repeatedly brings up a command “rtcp set stats on”, but … nothing changes in the logs. How can I see the produced jitter data so that I can start to visualize it?