Hi Micheal,

You are correct.  The problem has not occurred yet.  Good to know that the 401 
can be ignored.
Will post it again when it occurs.

Thanks,
Richard
  ----- Original Message ----- 
  From: Michael Zhang 
  To: Richard (Rogers @ work) ; Aloysius Thevarajah Lloyd 
  Cc: [email protected] 
  Sent: Thursday, March 15, 2007 10:22 AM
  Subject: RE: [on-asterisk] Debugging drop call problem


  It seems you did not provide sufficient traces for the problem you described. 
The "401" error is a normal challenge for "Register" authentication. You might 
need to keep your Ethereal or whatever sniffing tool open until the problem 
happens again.

  Cheers,
  Michael



------------------------------------------------------------------------------
  From: Richard (Rogers @ work) [mailto:[EMAIL PROTECTED] 
  Sent: Thursday, March 15, 2007 7:34 AM
  To: 'Aloysius Thevarajah Lloyd'
  Cc: [email protected]
  Subject: RE: [on-asterisk] Debugging drop call problem


  From the log, I see some 401 errors.  I wonder if this would mean and if it 
would be the one to blame?

  But 3 lines below it, the status was OK with 1 bind?

   

  Any suggestion is appreciated.

   

  3470.289876 192.168.0.111 -> 192.168.0.222 SIP Status: 100 Trying    (1 
bindings)

  3470.289951 192.168.0.111 -> 192.168.0.222 SIP Status: 401 Unauthorized    (0 
bindings)

  3470.360641 192.168.0.222 -> 192.168.0.111 SIP Request: REGISTER 
sip:192.168.0.111:5060

  3470.360838 192.168.0.111 -> 192.168.0.222 SIP Status: 100 Trying    (1 
bindings)

  3470.364668 192.168.0.111 -> 192.168.0.222 SIP Status: 200 OK    (1 bindings)

  3474.370689 192.168.0.111 -> 192.168.0.222 SIP Request: NOTIFY sip:[EMAIL 
PROTECTED]

  3474.438354 192.168.0.222 -> 192.168.0.111 SIP Status: 200 OK

  3483.629054 192.168.0.111 -> 192.168.0.222 SIP Request: OPTIONS sip:[EMAIL 
PROTECTED]

  3483.686565 192.168.0.222 -> 192.168.0.111 SIP Status: 200 OK

   

  Thanks,

  Richard

   

  -----Original Message-----
  From: Aloysius Thevarajah Lloyd [mailto:[EMAIL PROTECTED] 
  Sent: Tuesday, March 13, 2007 9:23 PM
  To: Richard (Rogers @ work)
  Cc: [email protected]
  Subject: Re: [on-asterisk] Debugging drop call problem

   

  Use

   

  tethereal -l -n udp port 5060 >sip.log

   

  capture the sip messages.

   

  U will see who is sending the BYE.

   

  Thanks

  Lloyd

   

  On 3/13/07, Richard (Rogers @ work) <[EMAIL PROTECTED]> wrote: 

  Hi,

  Occasionally, some calls get dropped for no obvious reasons.
  What is the best way to track it down? 
  Is there a way to log all call-terminations and the reasons associated
  with them?

  Thanks,
  Richard




  ---------------------------------------------------------------------
  To unsubscribe, e-mail: [EMAIL PROTECTED]
  For additional commands, e-mail: [EMAIL PROTECTED]

   

Reply via email to