canreinvite=no in the sip entry of the phone on the inside of the NAT.
See other posts on the list. Just search with google and add site:lists.digium.com Greetings, Tjardick ----- Original Message ----- From: "George Lin" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Thursday, September 25, 2003 9:33 AM Subject: [Asterisk-Users] SIP problem with asterisk > > > HI List, > > I have two SIP phones. one is 6002, w: > > which is behind a NAT, and another is 5009 which has public IP . > > When a call between 6002 to 5009, the 6002 cannot hear any from 5009, and > 5009 did hear from 6002. > > And in the sip debug, I see following message > > Sip read: > SIP/2.0 481 CallLeg/Transaction Does Not Exist > > > and I specified in sip.conf > [6002] > > type=friend > host=dynamic > nat=1 > qualify=yes > > [5009] > > type=friend > host=dynamic > nat=1 > qualify=yes > > Can anyone help me what can be wrong ??? > > Thanks, > > George Lin > > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > > _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
