[Freeswitch-users] Port question again

[Frank Carmickle]

Hello

On Wed, Jan 20, Eduardo Casarero wrote:
> Hi list, i'm a brand new freeswitch user (without previous asterisk/voip
> experience), after reading all wiki pages, google searchs, etc i need some
> help to solve a problem.
> 
> configuration:
> 
> Freeswitch -> Firewall (nat) -> internet -> Sip Provider
> 
> In my current configuration the gateway is REGED and inbound calls (from
> provider to freeswitch) works ok with good audio quality. However outbound
> calls don't. When i call through the gateway the destination phone rings,
> and when is answered there is no audio.
> 
> I've check with "show channels" in fs_cli and i cant see any codec in the
> read_codec write_codec part, they are blank. I've reviewed all sip profiles
> configuration, but obviously i'm missing something.

Sounds to me like your firewall is blocking outbound ports.  If it's a linux machine you'll want something like

-A OUTPUT -p udp --dport 16384:32768 -j ACCEPT

> 
> I will really appreciate any comment,guidance,help,etc. (if someone is in
> Buenos Aires/Argentina i can also offer a free beer!)

I am not but I'd sure love to try your local beer.

HTH
--FC