My problem is I have an asterisk box at my house. I have dahdi lines, and 3 sip2sip trunks, and 3 ipcomms sip trunks going into box. I have two remote users on nortel 1535 phones with holes in FW for thier phones 5060, 10k-20k, 23k, 24k.

I also have vpn with my work where I have another polycom phone connected to work system.

I am getting alot of random one way audio, or remote caller can not call speed dial that hits server and dials an outside line. the phone rings but no audio. I can call an outside sip trunk that does once in a while get back to remote user phone and have two way audio if I call from my cel.

calling between the two extensions, local and remote seems to work fine and get video. my office line across vpn seems to work fine also.

I am just reading a bunch of things about issues with sip and nat in 1.2.3 and wondering if anyone has it working properly with a number of connections like I have.  If not wondering if I need to look for another product as I am getting frustrated with this issue.

I have already tried these steps
http://doc.pfsense.org/index.php/VoIP_Configuration

I have also tested another nortel phone connecting to sip2sip.info directly. When I first hooked it up it would not register on the site with my asterisk box live. I stopped my phone server and rebooted the firewall to clear the connections and the phone came up. I then started my server and it stayed up. I can even deregister and reregister. I can call my sip2sip trunk and it will ring but then hangs up after 20 seconds. 2 way audio. when I call this from my extension to sip2sip it rings with 2 way sound sometime then hang up sometimes after 20 sec.

any thoughts would be appreciated.

I was looking at www.untangle.com the free version or trying to learn one of the iptable front ends. I would love to stay with this if I could figure out.