Hi Liviu, The Aastra is very similar to the Cisco phones when it comes to NAT. Although STUN is not the most reliable mechanism for NAT, it's handy. However, it all depends on how well your provider or Asterisk box handles NAT.
There is a "Nortel NAT traversal" feature, but it's not recommended. In your SIP settings leave everything else blank (or zero), and enter the following: - Proxy server - Registrar server Phone number and Authentication Name is your SIP username. When playing around with SIP ports you soon realize that 5060 or 0 (defaults to 5060) is your only alternative through NAT. My phone at my home office is connected to our (AtlasVoice) service (through NAT), plus that one line is connected to my personal Asterisk (on my home LAN). Works well. We use SER/Mediaproxy though, so it might not be totally realistic. Having said all this, the guys at Aastra are keen on getting this debugged, but I haven't had time to work on it. Being an enterprise phone it's really meant for LAN environments, where the PBX would take care of NAT/FXO/external lines. -- Bjorn -----Original Message----- From: Liviu Toma [mailto:[EMAIL PROTECTED] Sent: Wednesday, August 30, 2006 8:43 PM To: [email protected] Subject: [on-asterisk] Aastra 9133i Hello, I picked up one of the Aastra 9133i demo units yesterday at the meeting last night. I'm trying to get it configured and it looks like it doesn't support STUN. Does that mean I can't configure it to work from behind a NAT router with dynamic WAN address ? Thanks, Liviu --------------------------------------------------------------------- To unsubscribe, e-mail: [EMAIL PROTECTED] For additional commands, e-mail: [EMAIL PROTECTED]
