I have finally nailed down and resolved the problem. Thanks for all the responses. It has been very helpful and educational for me.
ztmonitor showed the TX gain on the phone + asterisk setting combined was way too high. After making iterative adjustments on both, the drop call problem can not be reproduced. Thanks, Richard ----- Original Message ----- From: "Phil Oxrud" <[EMAIL PROTECTED]> To: "Richard (Rogers @ work)" <[EMAIL PROTECTED]> Cc: <[email protected]> Sent: Sunday, March 25, 2007 2:47 PM Subject: Re: [on-asterisk] Debugging drop call problem - due to voice level > Try playing around with the txgain and rxgain values. I've also had > many problems with Digium's TDM2400E card, even though their tech > support was great. I've tried for over a month to fix my problems > (echo, calls dropping, breaking up) then finally got a Sangoma card > and the problems went away. > > On 3/25/07, Richard (Rogers @ work) <[EMAIL PROTECTED]> wrote: > > 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] > > > > > > --------------------------------------------------------------------- > > 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] >
