inbound routes

Discussion in 'General' started by akibsayyed, Mar 17, 2011.

  1. akibsayyed

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    hello all

    i am bit new to elastix.
    i am trying to make inbound call from one asterisk server to elastix server but i am unable to make call.my asterisk server is in usa & elastix server in india.

    here is details of config.


    asterisk server

    extension 1000 is used as trunk in elastix server.


    ie. extension 1000 will act as trunk for elastix server.

    but when i place call on 1000 exten i get following error



    == Using SIP RTP CoS mark 5
    -- Executing [1001@default:1] Dial("SIP/1002-00000009", "SIP/1000") in new stack
    == Using SIP RTP CoS mark 5
    -- Called 1000
    [Mar 17 18:43:48] WARNING[3109]: chan_sip.c:19048 handle_response_invite: Received response: "Forbidden" from '"1002" <sip:1002@115.252.189.231>;tag=as299e5792'
    -- SIP/1000-0000000a is circuit-busy
    == Everyone is busy/congested at this time (1:0/1/0)
    -- Auto fallthrough, channel 'SIP/1002-00000009' status is 'CONGESTION'


    & on cli of elastix server i get


    == Using SIP RTP TOS bits 184
    == Using SIP RTP CoS mark 5



    can any one help in this
     
  2. fmvillares

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    which technologies are u using in all the lines?????only sip??? 1000 is an extension or trunk???
     
  3. akibsayyed

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    i am using sip . 1000 is extension in asterisk server & used as trunk in elastix server.
     
  4. fmvillares

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    registered as a kind of loopback?...the reason is clear it is not registering...could u show the configs? and please read about how to ask questions in a smart way as bob fryer puts in his newbie guide in the first sticky post
     

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