We are currently using Freepbx ver 12.0.68 with the default audio codec. It works great most of the time. Other times I have calls randomly drop or the audio completely drops off or other times I have intermitted audio drops for like 1 sec.
Do I need to enable Jitter Buffer and set this up. Or do I need to buy the g.729 codec from Digium and install it. Some of my remote sites that VPN back into this PBX server have limited bandwith but even the places with high bandwidth experience this sometimes. I don’t very high usage on the systems resources either.