Hi Ovidus, The log below does not seem to have anything out of the ordinary other than show the BYE from the asterisk server to the phone.
All IPs below have been masked out and are not the real IPs. But the 192.168.1.333 if the IP for the AAstra 480iCT 192.168.0.111 is the asterisk server. Any suggestion is appreciated. Is there a way I can limit the outbound voice level on either the phone or asterisk? I highly suspect this is the cause of the problem. But if this is the case, I should not be the first one to encounter it, right? Would this be a hardware problem i.e, the digium card? Thanks, Richard. 11.337723 192.168.0.111 -> 192.168.1.333 SIP/SDP Request: INVITE sip:[EMAIL PROTECTED], with session description 11.456023 192.168.0.111 -> 192.168.1.333 SIP/SDP Request: INVITE sip:[EMAIL PROTECTED], with session description 11.492035 192.168.1.333 -> 192.168.0.111 SIP Status: 100 Trying 11.499646 192.168.1.333 -> 192.168.0.111 SIP Status: 180 Ringing 11.525902 192.168.1.333 -> 192.168.0.111 SIP Status: 180 Ringing 12.272209 192.168.1.333 -> 192.168.0.111 SIP/SDP Status: 200 OK, with session description 12.273121 192.168.0.111 -> 192.168.1.333 SIP Request: ACK sip:[EMAIL PROTECTED] < I am shouting very loud here!!!!> 23.962274 192.168.0.111 -> 192.168.1.333 SIP Request: BYE sip:[EMAIL PROTECTED] 24.076478 192.168.1.333 -> 192.168.0.111 SIP Status: 200 OK 45.969554 192.168.1.333 -> 192.168.0.111 SIP Status: 200 OK 48.005881 192.168.0.111 -> 75.190.223.32 SIP Request: REGISTER sip:75.190.223.32 52.006652 192.168.0.111 -> 75.190.223.32 SIP Request: REGISTER sip:75.190.223.32 Thanks, Richard -----Original Message----- From: Ovidiu Sas [mailto:[EMAIL PROTECTED] Sent: Saturday, March 24, 2007 12:16 AM To: Richard (Rogers @ work) Cc: [email protected] Subject: Re: [on-asterisk] Debugging drop call problem 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] > > --------------------------------------------------------------------- To unsubscribe, e-mail: [EMAIL PROTECTED] For additional commands, e-mail: [EMAIL PROTECTED]
