Remote Extension incoming/outgoing no voice heard

Discussion in 'General' started by japok25, Jan 3, 2011.

  1. japok25

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    The asterisk server w/ a router box at the office works perfectly suddenly the hardware crashes I have backup setting of elastix so I tried to restore it into another server with the same version of elastix 1.6.2-1 and I think it restored all the settings. Now I buy a new router and the brand is TRENDnet TW100-BRF114 I have already open all the ports of the router. Then I test it locally using ATA’s and Softphones where SIP Extension incoming and outgoing calls the voice it is works fine and now I test it remotely using Softphones same procedure incoming and outgoing calls it can ring anyone but you cannot hear any voice both ways.

    Also I already tried to change the Sip_Nat.conf
    nat=yes
    externip=xxx.xxx.xxx.xx (external static ip at office)
    localnet=172.16.2.0/255.255.255.0 (local network at office)


    Can anyone have any clues that may fix the problem?
     
  2. circuitid

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    I would recommend looking at your firewall settings. Note that the SIP protocol uses port 5060 for signaling and RTP ports 8766 - 35000. If you can hear audio internally but not externally then it most likely has to do with the RTP ports being blocked to the external network.

    For NAT and VoIP, I would also recommend looking at this article:
    http://www.voip-info.org/wiki/view/NAT+and+VOIP
     
  3. dicko

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    A quick note for those setting up their NAT routers.

    Although 5060 is the default and conventional port to use, some would suggest that it is a security risk, it can easily be changed in

    /etc/asterisk/sip.conf (or one of it's included files)

    with

    bindport=50XX

    Your external extensions and VSP must of course agree on this.

    the open ports for rtp (the audio load) are set in

    /etc/asterisk/rtp.conf

    by default they are

    rtpstart=10000
    rtpend=20000

    your firewall needs only match what you have here.
     
  4. fmvillares

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    hi my great friend dick!!! 1 add...udp only by default...tcp is not used by default in asterisk (but could be used in sip since 1.6.x)
     
  5. dicko

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    Fernando, my friend and fellow black sheep :) welcome back . . .

    Of course no tcp needed for rtp ;)

    regards

    Dick
     
  6. japok25

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    Gentlemen,

    Thank you very much for giving an ideas on how to passthrough the NAT routers. I figure it out how to configure my router TRENDnet TW100-BRF114 In this kind of router the Port Forwarding is a virtual server so I created a SIP Server 5004 – 5082 and RTP Server 8766 – 35000. When I test it in a remote extension it works thank again to all of you.

    Here the link for your reference: http://trendnet.com/kb/kbp_viewquestion ... &catId=193


    Regards to all,

    Japok
     

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