Date: Tue, 9 Oct 2007 19:17:44 -0400 From: "Dave" <dmehler26@woh.rr.com> To: <freebsd-pf@freebsd.org> Subject: pf and sip Message-ID: <000301c80aca$99695db0$0200a8c0@satellite>
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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
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