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.

Full article here:
Improving Asterisk call quality with SIP jitter buffers (SysAdminMan Blog)

Note that the article mentions that “This can by done with the FreePBX SIP Settings module”, but does not mention the settings when using that method. Although they should be fairly obvious, we’ll mention them anyway:

Under Settings / Asterisk SIP Settings in the Jitter Buffer Settings section:
Jitter Buffer: Enabled
Force Jitter Buffer: No
Implementation: Adaptive
Jitter Buffer Logging: Disable
Jitter Buffer Size: 200 (jbmaxsize) 1000 (jbresyncthreshold)

The two items in bold text correspond to the ones discussed in the above-mentioned article.