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Cisco ASA/FreePBX One Way Audio Issue

Posted on 2012-04-13
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Last Modified: 2012-06-08
We currently have a Cisco ASA 5505 Firewall and a FreePBX server that is setup.

The ASA is running Version 7.2

I am using ASDM via the GUI to make all my changes (I am a rookie when it comes to Cisco)

When I make calls extension to extension we get two way audio and it works perfectly. However when I make calls directly out from my extension to my cell phone or any other phone outside of our extension the person can hear me but I cannot hear them. The phone call then disconnects the caller but keeps my line open.

I have the ports open on the firewall and I have setup SIP traffic inside the  NAT Policy (see pictures)

My gut tells me this is a NAT issue but my gut has been wrong in the past! PLEASE PLEASE PLEASE HELP!
NatPolicy.png
FirewallPolicy.png
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Question by:nerdeasy
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TimotiSt earned 500 total points
ID: 37847063
You might need the allow high-port UDP traffic from outside (from your peer media-gateways) to the FreePBX. SIP is only used for session setup, the actual audio travels in RTP. The port-range used is usually determined by the SIP-trunk setup in Asterisk.

Tamas
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