Hi Richard, The logs are showing to restarts and that's it: 23:00:52 23:04:29
Did you collected those logs from the /var/log/asterisk/messages file? You don't have any type of call related logs there. You need to enable debug and verbose logs on the console and capture there. Regards, Ovidiu Sas On 3/24/07, Richard (Rogers @ work) <[EMAIL PROTECTED]> wrote:
Hi Ovidiu, The log has been included after full debug is turned on. I did not see anything out of the ordinary. May be you can catch some things for me? Thanks, Richard -----Original Message----- From: Ovidiu Sas [mailto:[EMAIL PROTECTED] Sent: Saturday, March 24, 2007 8:20 PM To: Richard (Rogers @ work) Cc: [email protected] Subject: Re: [on-asterisk] Debugging drop call problem - due to voice level Hi Richard, Since the asterisk is dropping the call, you can enable debug logs on the asterusk server and try to debug it. Edit /etc/asterisk/logger.conf and fix the following line: console => notice,warning,error,debug Restart asterisk and enable debug logs: set debug 10 Place the call, shout and check what's going on ... Regards, Ovidiu Sas PS: My name is Ovidiu, Ovi for short :) On 3/24/07, Richard (Rogers @ work) <[EMAIL PROTECTED]> wrote: > 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] > > --------------------------------------------------------------------- To unsubscribe, e-mail: [EMAIL PROTECTED] For additional commands, e-mail: [EMAIL PROTECTED]
