> 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
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  • pf and sip Dave
    • Re: pf and sip Asbjørn Clemmensen

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