Call Drop after 20 Sec

Discussion in 'General' started by kileak, Nov 18, 2008.

  1. kileak

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    Hi All,

    Good day all, anyone faced the problem where the calls will drop after 20 seconds? The call just got hang-up signal and will end. Anyone that could help on this?

    Thanks in advance.
     
  2. Bob

    Bob

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    kileak,
    Can you confirm what your system is e.g.

    1) Elastix Version (on the top of your Elastix GUI)
    2) Whether you are running zaptel or SIP or IAX trunks
    3) What provider you are using
    4) If you are using VoIP trunks, what router you have

    If you give this information up front, you will get a lot more replies.

    As a guide however you are describing an issue that can occur with SIP trunks (in particular), where the NAT on the router is not working correctly, or portforwarding may be needed. Typical issues that can occur
    that are related to router issues (NAT/Port forwarding etc)

    1) 20-30 seconds dropout - generally depends on VoIP provider, but basically their system is expecting a response from your system and does not get it. Usually because your system didn't get the initial request for response due to a NAT/Port forwarding issue

    2) 5min - usually a NAT session with no further traffic times out usually about 5 minutes (typical but not a standard) if no further outgoing traffic occurs. Normally this kept alive by a simple Ping/Pong method between your Elastix system and the VoIP provider, but if this is somehow being blocked (even by misconfiguraton of Elastix), then the NAT Session will time out. Some NAT implementations on routers will reset the session timer on outbound only, and some on outbound and inbound.

    Usually setup of permanent port forwarding will eliminate some of these NAT issues, if you have a router that is not handling SIP NATing well.

    Some things you can also try is look at sip_nat.conf. Normally this requires a couple of lines such as Externip and localnet which may help with some SIP Trunks. Also you may find that you need to add a NAT line into your trunk, or remove it.

    Hope this helps

    Bob
     
  3. Bob

    Bob

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    Kileak,

    You can ignore the previous message. Found another of your postings relating to the same issue, and have answered that as well.

    Will leave the post above for others having SIP NAT issues.


    Regards

    Bob
     
  4. kileak

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    Bob,

    Thanks a lot, will look into it as we are using Port Forwarding and I was totally forgotten about that.So I will need special reconfigurations for port forwardings as well in the server right?

    Thanks a lot for the reminder.
     

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