[Asterisk-Users] SIP w/NAT on Grandstream 496 and Call-Waiting

[Dave Wise]

Hello All;
I have a Grandstream 496 ATA and it is behind a NAT Router.  The phone 
service works well, but it is setup to support Call-Waiting, which it 
does not do.  When I am on the phone and someone calls, instead of 
getting a ring, they go straight to Voicemail with the busy message.  I 
used Ethereal to watch what happens and I notice a SIP Redirect 3XX.  Is 
this normal?  Does anyone know if Call-Waiting will work behind a NAT 
router (with a Stun Server)?