do this in sip.conf [youruser] type=friend secret=adsds host=dynamic nat=yes qualify=yes and other paramters for your user. They key is nat=yes and qualify=yes. This assumes you have a real IP for your Asterisk server and you are trying to connect a SIP phone which is behind NAT.
David ----- Original Message ----- From: "Heison Chak" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Monday, February 23, 2004 7:50 PM Subject: Re: [Asterisk-Users] SIP over NAT > SIP works fine behind NAT if you have externip, localnet & localmask > defined in sip.conf. I believe it was committed since 0.7.1. > > -Heison > > On Mon, Feb 23, 2004 at 08:51:23PM +0100, Marc Fargas wrote: > > Assuming that getting H323 to work over NAT is almost really hard? What is > > about having both SIP clients venid different NAT?s ? is it posible or as > > hard as H.323? > > > > Thanks! > > Marc. > > > > > > > > _______________________________________________ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users