[Asterisk-Users] 30 button vip 1 way audio

[Bill Coward]

what module should i use, and how? Thanks for you input...

-Bill


On 5/3/05, Eric Wieling aka ManxPower <eric at fnords.org> wrote:
> Bill Coward wrote:
> > It seems this is a redundant question (or at least problem) within
> > this group, but I'm unable to find a solution/combination.
> >
> > I have 3 30 button VIP phones running behind 3 different firewalls/servers (NAT)
> >
> >  My asterisk server is running great with a public IP address (no NAT)
> >
> >  The 30 button phone work fine to a point (load, register, date&time,
> > and process calls to and from) but
> >
> > Only get one way audio, and my skinny debug shows the internal ip
> > address (10.x.x.x) rtp session packets leaving (attempting,
> > firewalled) to leave my server.  Which geek nob do I need to turn to
> > get my server to initiate the rtp on the public address...
> >
> > I've tried the nat=yes (and no...) statement in the skinny.conf tried
> > host=dynamic host= (ip address)
> 
> As I understand it, chan_skinny does not support phones behind NAT.
> 
>