Sip.conf not is parsing correctly?


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


Take a look at /var/log/asterisk/full to see if the full log file tells you anything.