sip_nat.con entries make my zap channels disconnect after 20-25 seconds into a call?

Asterisk 1.4 and Freepbx 2.4 Pbxinaflash

OK here is my real problem. When I put entries in the sip_nat.conf this causes my zap trunks to disconnect after 24 seconds into the call. I have tested with 2 distros, same deal when this remote phone goes online. It’s all making sense now, all the problems started when I put this remote phone online last week. I just removed the contents of the sip_nat.conf and the zaps are not dropping anymore calls. Any idea whats causing this?

Well you’ve given us no details to even take a guess. What you describe dare I say, is not possible as sip will not effect zap. Now that is not to say that your phones are not sip and it’s causing the phones to have a issue and not the zap.

Please provide complete details. Distro/OS build, version of asterisk and FreePBX, what you exactly you put in the sip_nat.conf file. What your network is like. One subnet, multiple subnets, etc. you get the point. since you’ve provided non of these details we can’t take a wild guess as then you’d be mad at us for attempting to do just that.