Hope everyone is safe.
I built this FreePBX VM a while ago but now client wants to use it. I use Sangoma as a SIP trunk. When caller dials menus prompt the sound, quality becomes bad (it fluctuates too). The volume goes up and down, and it distorted.
Build: 220.127.116.11 and most modules are updated.
Phones are on same LAN where server is.
I know I must check codecs, but I never done it. If someone could give me right direction what exactly should I check.
Thank you in advance!
Could be any number of things. We have noticed persisting degraded call quality when the free memory is consumed (see any use of swap).
htop will let you monitor resources from SSH.
You can measure jitter and packet loss from Asterisk CLI:
sip show channelstats
Might be good to pull the logs, incase they show anything.
The swap usage currently about 212M, memory utilization is 1Gb/1.7Gb.
Should I increase it? It easiest of all.
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