It's been over six months now that Twilio has opened up its Elastic Sip Trunking to the masses. This video will show you how to hook up your FreePBX box to Twilio's Sip network and take advantage of their super low per minute rates.
This setup uses chan_sip and NOT chan_pjsip. By default, if you install FreePBX 13 with asterisk 13 your install will set the chan_pjsip protocol to the standard 5060 bind port and chan_sip to bind to port 5160.
To force chan_sip (if you installed asterisk 13) go to:
Settings > Advanced Settings > then change "Sip Channel Driver" to chan_sip.
Next go to Settings > Asterisk Sip Settings and update the Chan_Sip Bind Port to 5060 and the TLS Bind Port to 5061.
You will also need to update the chan_pjsip Ports to 5160 and 5161.
If you installed asterisk 11 from the start then the chan_pjsip tab will not appear in the Asterisk Sip Settings menu, but you may still have to update the ports in the Settings > Asterisk Sip Settings chan_sip tab.