[asterisk-users] One Way Audio Problem

[GNUbie]

Hello Steve,

On Thu, Oct 16, 2008 at 12:42 PM, Steve Totaro
<stotaro at totarotechnologies.com> wrote:
> canreinvite defaults to yes, whether specified or not.
>
> http://www.voip-info.org/wiki/view/tips
>
> If you follow these directions adapting to your particular
> circumstances and it doesn't work, post your whole sip.conf
>
> Start asterisk with verbose set to 3 or so and turn on sip debugging.
> I get somewhere in the debug, you will see local NAT IPs that don't
> belong there, or it will just work.

My /etc/asterisk/sip.conf is at
http://lists.digium.com/pipermail/asterisk-users/2008-October/220256.html
and my SIP phone is located within the LAN where the Asterisk box is
also part of it.

Regards,

GNUbie