Hi All, I think I have found the bloody problem. It is to do with the Digium 400 series cards where ZAP drops calls randomly. Well, my thorough testing revealed it was not RANDOM. It relates to the voice level. It will disconnect when voice fluctuation is high - when you speak loud, laugh etc.,
There are links on the internet about it but I have not seen a fix yet. E.g., http://lists.digium.com/pipermail/asterisk-users/2004-June/044158.html I can not believe Digium card has such a fundamental problem in their products. I am very disappointed. But any suggestion, other than getting rid of the cards, will be appreciated. Thanks, Richard -----Original Message----- From: Ovidiu Sas [mailto:[EMAIL PROTECTED] Sent: Saturday, March 24, 2007 11:26 PM To: Richard (Rogers @ work) Cc: [email protected] Subject: Re: [on-asterisk] Debugging drop call problem - due to voice level 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] > > --------------------------------------------------------------------- To unsubscribe, e-mail: [EMAIL PROTECTED] For additional commands, e-mail: [EMAIL PROTECTED]
