2 thoughts on - SIP calls destroyed after 1:20

  • Are you using originate? Check your originate timeout.
    Are you limiting your call length on Dial()… check your L options.

    Asterisk will send a BYE if it hits an internal timer that’s set to
    destroy the call at a specific time.

    For instance… this is almost guaranteed to cause problems

    Action: Originate
    Timeout: 30

    Timeout is in milliseconds if I remember correctly, so after 30
    milliseconds, which isn’t nearly enough time to establish a call,
    asterisk will kill the call.

  • On Mon, Nov 15, 2010 at 3:11 PM, Jeremy Kister
    wrote:

    Play around with the session-timers in sip.conf. We had an issue with
    our sip provider, and this turned out to be a workaround. Their end
    was okay with supported session timers, but not session timers which
    were marked as required.

    -M