pf and sip

[Asbjørn Clemmensen]

> 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.

Try looking into siproxd from the ports system. Also check their website[1]
which details what ports need to be forwarded. This of course requires your
phones to be able to use a proxy.

[1] http://siproxd.sourceforge.net/

> 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

--
Asbjørn Clemmensen