what is required to used a SIP phone behind a NAT router

Discussion in '3CX Phone System - General' started by thefinisher, Nov 28, 2007.

  1. thefinisher

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    I have 3CX PBX behind a NAT device and SIP phone that is located on another network be hind a NAT device. I have configured the firewall to forward ports 5060 UDP, 9000-9007 UDP 7000-7000 UDP to the 3CX PBX. I have configure the stun settings on the SIP phone to stun2.3cx.com.
    the problem is that phone registered with the PBX but I get no audio when I make calls. what other settings are neccessary to for this case to work
     
  2. RobLloyd

    RobLloyd Member

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    Do you have Extension is External checked off on the ext settings in 3CX?
     
  3. webguyz

    webguyz New Member

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    Show us a portion of the log (in debug) of a call. A lot of times in the log you see the remote extension using a non-routeable IP like 10.x.x.x or 192.x.x.x and its because your sip device has to be able to to give out its NAT ip and not its actual lan ip.

    I was setting up a linksys PAP2 and it has a setting to use NAT IP and until I turned on that option my remote audio would not work because the server would attempt to open an audio channel to my remote using 10.x.x.x which was my remotes local lan ip, but that is not routable on the internet.
     
  4. Ralph

    Ralph Member

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    Good evening

    You didn"t mention the location and network status of the SIP phone itself. Is there another firewall in front of the SIP phone? If so have you checked to ensure that it isn't blocking return traffic?

    Take care,
     

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