globally set the session transport to be udp
bind the control and media source interface to be the loopback port
set the control channel to use secure sip

turn on SRTP
configure SRTP to fallback to RTP if SRTP not available
forward the display name with the caller id
substitute CLID number for non-existant display names

set sip-ua authentication username and password
allow sip gw to register e.164 nums on external phones
set ip address for sip server
set timers for INVITE response to 1000ms
set timers for ACK response to 1000ms
set timers for INVITE to be valid for 1000ms
set retry type for invites to 2
bind both control and media to use source interface fa0/1

set up dial peer for sipv2
any 4 digit number starting with 3
send to the sip-server you defined in the above sip-ua section
use secure sip for this dial peer

setup a dial peer for sipv2
strip the CLID if it's designated as private on incoming PRI