debug audio or channel

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Asterisk Users 2 Comments

Does anyone have any short answers on how I can fix this problem:

A calls B.
B rings
Says connected.
But the call is not bridged and therefor no audio passes.
very simple dial plan.

Frustrated.
v 1.8

2 thoughts on - debug audio or channel

  • 0) Don’t use clouds.

    1) Start poking about with wireshark and see if audio is flowing.

    If the call is connecting, it sounds like 5060 is being passed OK. If
    audio is not being passed, it sounds like… Well, it doesn’t sound like
    anything, so the problem may be a router or firewall issue.

  • As I suggested yesterday, and in accordance with Steve’s answer, you should check if your A and B phones are behind a NAT and, if they are, review the configuration of your router / firewall. Although it is not clear from your e-mail which is the complete picture of your system, let’s suppose that your phones A and B are in your LAN, connected through a router to Internet, and that they both are registered against an Asterisk PBX deployed on a server hired in the net with possibly a fixed IP and no NAT, neither special firewall rules that could avoid any type of traffic.

    If it is the case, you can see http://www.voip-info.org/wiki/view/NAT+and+VOIP for information about the issue. There you can read:


    At first, for both the calling and the called party everything will appear just fine. The called party will see the calling party’s Caller ID and the telephone will ring while the calling party will hear a ringing feedback tone at the other end. When the called party picks up the telephone, both the ringing and the associated ringing feedback tone at the other end will stop as one would expect. However, the calling party will not hear the called party (one way audio) and the called party may not hear the calling party either (no audio).

    In the same place, you can find some avoidances and workarounds for the problem.

    Regards,
    José Miguel

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