Hi, I am running SME Server 6.0 with Asterisk PBX using SIP. My SME also acts as a NAT router between my cable modem and my office LAN. All calls from the inside to IAX or SIP clients on another proxy go through fine, however any user connected as a SIP peer to my Asterisk server has no inbound nor outbound audio. I've tried numerous times to implement a template so that UDP ports 10000 to 20000 are open but everytime a SIP user tries to call in or out of the switch, I see it in the denylog and he gets no in/out audio. Signalling and connection are fine (port 5060 seems to work correctly) but everything else is on the fritz.
Now, can anybody help me write a template that'll FORCE open those ports so I don't get them in the denylog? I've even tried to put that port range in the INPUT policy instead of InboundUDP, to no avail. This is very frustrating as many telecommuters can't get access to my phone system.
Thanks for your help...