I had a customer let me know that they had improved their call quality from WiFi and 3G connections by turning on the Asterisk jitter buffers for SIP connections. If you have any extensions where connection quality is intermittent it could be worth trying.
This can by done with the FreePBX SIP Settings module or by adding the following lines to –
After changing you should restart Asterisk or ‘sip reload’ from the console.
If you are just using Asterisk the change would go in the [general] section of sip.conf
If you try this option and notice any difference to call quality please post a comment below. Thanks!