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.. HTH, Austin -----Original Message----- From: Victor Pasten [mailto:[email protected]] Sent: Thursday, August 12, 2010 3:11 PM To: [email protected] Subject: Re: [pfSense Support] asterisk behind pfsense+remote sip clients ----- Mensaje original ----- De: "David Burgess" <[email protected]> Para: [email protected] Enviados: Miércoles, 11 de Agosto 2010 15:56:25 Asunto: Re: [pfSense Support] asterisk behind pfsense+remote sip clients On Wed, Aug 11, 2010 at 1:53 PM, Victor Pasten <[email protected]> wrote: > Hi Guys, recently I've installed a asterisk server (in my lan, behind pfsense > 1.2.3-release), everything it's ok, except for some remote sip extentions > (polycom device, and x-lite softphone) that periodically are loosing her > registration. >>Most voip problems with pfsense can be solved here: http://doc.pfsense.org/index.php/VoIP_Configuration --------------------------------------------------------------------- I've followed the instructions, but nothing... Now I'm testing with M0n0 sip+nat.... bad mix --------------------------------------------------------------------- To unsubscribe, e-mail: [email protected] For additional commands, e-mail: [email protected] Commercial support available - https://portal.pfsense.org
