[asterisk-users] One Way Audio Problem

[Jeff LaCoursiere]

On Thu, 16 Oct 2008, GNUbie wrote:

> Hello,
>
> On Tue, Oct 14, 2008 at 12:07 AM, Jeff LaCoursiere <jeff at jeff.net> wrote:
> >
> > A packet trace will probably show exactly what is happening.  Try:
> >
> > tcpdump -nlXs 8192 -i eth0 port 5060
> >
> > You should be able to see the SIP information going back and forth and
> > will probably show you that your NAT rules are applying when they
> > shouldn't.  I agree with first turning off your firewall and testing...
> > but if that actually solves the problem you need to know why.  This should
> > tell why.
>
> Why eth0 when in fact it is not being used AFAIK? My Asterisk box is
> connected to the LAN via its eth1 interface and the SIP phone is
> calling from the LAN to the analog telephone via FXO/POTS. Again,
> below is the call scenario diagram:
>
> [SNOM] ==LAN==> eth1 [ASTERISK] fxo ==POTS==> [ANALOG_TELEPHONE]
>                                             eth0
>                                               ||
>                                         INTERNET

You should try on both interfaces.  If you see packets on eth0 then your
NAT rules are leaking!  Try on eth1 to see the SIP headers and tell if
your NAT rules are doing what you expect.

This is always my first attack...

j

>
> Please advice.  Thank you in advance.
>
> Regards,
>
> GNUbie
>
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