Hello Steve, On Thu, Oct 16, 2008 at 12:42 PM, Steve Totaro <[EMAIL PROTECTED]> 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 _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
