sip.conf not is parsing correctly?

Discussion in 'General' started by nancy, Oct 13, 2009.

  1. nancy

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    Hello

    Each time is restarted the server, the VoIP phones is listening busy, when I try of call, then I should do

    1. sip reload
    2. open sip.con and save it,
    3. sip reload
    4. dialplan reload

    With this, the phone give tone correctly

    Why, not simply work fine without do it? without have to than do this each time is restarted the asterisk server?

    Seem be not is parsing correctly the sip.conf when asterisk is started, How can I verify this?

    Best Regards
     
  2. arusnet

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    Is a new installation?
    If it were not new, it began to fail from that time?
    Maybe you have some backup before the failure?

    The problem you have is nothing more normal and logical to reinstall the system. But it would be very useful to know where the problem is never to discuss the same mistake.
     
  3. nancy

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    Hi

    I reinstall the system, but the problem is the same.
    How can I fix it?
    This is what shows me in the CLI>

    [Oct 13 14:32:35] WARNING[3508]: chan_sip.c:4181 create_addr: No such host: ProveedorSip
    [Oct 13 14:32:35] WARNING[3508]: app_dial.c:1450 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown)
    == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [85214921354564@internas:2] Hangup("SIP/401-0888f898", "" ) in new stack
    == Spawn extension (internas, 85214921354564, 2) exited non-zero on 'SIP/401-0888f898'

    Best Regards
     

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