Now, that you are able to reproduce the problem, sniff the traffic and find out who is dropping/disconnecting the call.
Regards, Ovidiu Sas On 3/23/07, Richard (Rogers @ work) <[EMAIL PROTECTED]> wrote:
Hi, I think I have finally figured out the direct or indirect cause of the problem. I was using trixbox 1.2 and later upgraded to 2.0 – latest and greatest. The problem can be reproduced in both. I have a Aastra 480i CT but I don't think its phone related. And here is my observations. When I talk loud consistently for 5-10 seconds, the call will be dropped. For a test, I shouted into the phone for 5-10 second, the call would for sure be dropped. If I talk softly in lower volume, the call will last. I think Asterisk is dropping it when talk vol is exceeded certain limit and the problem is not related to the phone. Any suggestion will be appreciated. Thanks, Richard -----Original Message----- From: Richard (Rogers @ work) [mailto:[EMAIL PROTECTED] Sent: Thursday, March 15, 2007 10:25 AM To: Michael Zhang; Aloysius Thevarajah Lloyd Cc: [email protected] Subject: Re: [on-asterisk] Debugging drop call problem 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]
