pf and sip

[Dave]

Hello,
    I've got a FreeBSD 6.2 gateway/router/firewall providing nat services 
among others. I've just tried to hook up voip phone services, i did some 
checking and it is using the sip protocol. I'm not getting a dial tone and 
calls aren't happening. According to the digital box i have it can't contact 
the login server. Below are my pf rules. If anyone has pf and sip working 
i'd be interested in hearing from you.
Thanks.
Dave.

ipphone1="192.168.0.9"
sip="5060"
sip1="5061"
# One translation line per IP phone. static-port is necessary to make pf 
retain the UDP
# ephemeral port, so that the remote SIP proxy knows what session we belong 
to
nat on $ext_if proto udp from $ipphone1 to any -> ($ext_if) static-port
# experimental sip for viatalk
pass in quick on $int_if inet proto udp from 192.168.0.9 port $sip to any 
keep state
pass in quick on $int_if inet proto udp from 192.168.0.9 port $sip1 to any 
keep state
pass out quick on $ext_if inet proto udp from $int_if port $sip to any keep 
state
pass out quick on $ext_if inet proto udp from $int_if port $sip1 to any keep 
state