Looks like you got it right. IAX works well with NAT to overcome some of the headaches of SIP. As stated from some of the others, it does sound like a keep alive issue at this point - your config checks out..
As previously stated, check the sip registration time and the keep alive timeout.. depending on the phone you use, it could very well have some nat friendly settings too.. -----Original Message----- From: Victor Pasten [mailto:[email protected]] Sent: Thursday, August 12, 2010 4:26 PM To: [email protected] Subject: Re: [pfSense Support] asterisk behind pfsense+remote sip clients ----- Mensaje original ----- >> De: "Austin G. Smith" <[email protected]> Para: [email protected] Enviados: Jueves, 12 de Agosto 2010 15:32:55 Asunto: RE: [pfSense Support] asterisk behind pfsense+remote sip clients I overcome this issue most of the time by defining your port range w/ asterisk for RTP in the rtp.conf file. Then redirect those ports from the nat device to the asterisk box inside. Make sure you do what needs to be done for nat keepalive if you have states enabled. Also, don’t forget to open 5060 udp on nat to the inside asterisk box. Also note, you can adjust the amount of ports for RTP needed based on how many phones you have. The lower the amount of phones, the lower amount of ports to forward. Mess with port address translation (PAT) or port forwarding, and also try 1:1 nat if you have the public ip's to spare.. my pat is: 5060 -> asterisk(ip_internal) 10000-20000 -> asterisk(ip_internal) 4569 -> asterisk(ip_internal) nat 1:1, impossible.... We've only 1 public ip address --------------------------------------------------------------------- To unsubscribe, e-mail: [email protected] For additional commands, e-mail: [email protected] Commercial support available - https://portal.pfsense.org
