voice-port 2/0
!
dial-peer voice 1 voip
destination-pattern …
session protocol sipv2
session target sip-server
session transport udp
dtmf-relay rtp-nte
codec g711ulaw
!
dial-peer voice 6363 pots
description Test analog phone.
destination-pattern 6363
port 2/0
authentication username 6363 password xxxx
!
gateway
timer receive-rtp 1200
!
sip-ua
registrar ipv4:192.168.1.200 expires 300
sip-server ipv4:192.168.1.200
!
Note that if it’s on the same router, it may not even use FreePBX, working on that part soon.
On FreePBX: All in the GUI
Create extension.
User extension : 6363
Display name : test
secret : xxxx
No different than any other IP Phone.