VoIP over NAT issues: Ring but no audio; disconnects

L1 Bithead

Yesterday i have sesion with support from LA and waiting for solusion.

L3 Networker


did u resolve problem


L1 Bithead

I had the same problem and finally pointed the outside interface of the VOIP PBX to the internet bypassing the PAN. It has a built in firewall that's satisfactory for it's purpose. I created a zone on the PAN to get to he voice subnet for secure management of the VOIP PBX and also to keep the voice subnet off my internel network. It works for now.

L1 Bithead

yes, my problem is resolved. support offered  in my case to use application ovverride for sip application, and it's work.

Not applicable


I'm experiencing the same problem.  Could you provide details of the application override fix please?

Mine looks like this, but is not working;


The 10. address is our PBX, the 80. address is the remote PBX and the 5. address is our external IP.

Not applicable

Over three hours on the phone to PA support this morning got it sorted.

My applictation override rules were not working, as they were pointing at predefined applications.  Custom Apps for both SIP and RTP were created which then started to allow some RTP through.

We were still getting quite a few calls with no audio (some were coming through).  The PA engineer discovered that this was due to us using the dynamic-ip-and-port option in our outgoing source translation.  Changing this to static-ip fixed it.

Here's what my working setup looks like;

Security Rules;


NAT Rules;


Application Override Rules;


Custom SIP Application;


Custom RTP/RTCP Application;


Hope this helps anybody experiencing similar problems.  PA have taken a support dump, so may have a more elegant fix at some point.

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