Re: [asterisk-users] sip error logging
On Sat, Apr 16, 2011 at 6:05 PM, Jeremy Kister asterisk...@jeremykister.com wrote: bumping once before sending it to the tracker. Original Message Subject: [asterisk-users] sip error logging Date: Fri, 15 Apr 2011 03:39:23 -0400 I recently noticed that asterisk is not logging unknown sip connections. I'm not sure if I've broken something or if asterisk itself has been broken. the last entry I have is: /var/log/asterisk/messages.9:[Feb 10 23:58:48] NOTICE[9868] chan_sip.c: Registration from 'sip:22942@10.0.0.3' failed for '10.0.0.228:5060' - No matching peer found my logger.conf looks like: # grep -v '^;' /etc/asterisk/logger.conf [general] [logfiles] console = notice,warning,error,dtmf messages = notice,warning,error,verbose,dtmf,fax if i send 'options' or 'register' from a non-configured sip peer, i dont see anything in the log. am I missing something ? * i can replicate this behavior on 1.8.2.3 and 1.8.3.2 -- Jeremy Kister http://jeremy.kister.net./ This may sound like a stupid question, but what are your verbosity and debug levels set at currently? Sherwood McGowan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip error logging
On 4/17/2011 3:16 AM, Sherwood McGowan wrote: This may sound like a stupid question, but what are your verbosity and debug levels set at currently? nope, thats exactly the type of thing i'm wondering if i'm missing :) but, i tried with verbose 3/debug 0 (which worked in 1.6), and i also tried with verbose 10/debug 10 before posting. no dice. -- Jeremy Kister http://jeremy.kister.net./ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip error logging
On Sun, Apr 17, 2011 at 2:24 AM, Jeremy Kister asterisk...@jeremykister.com wrote: On 4/17/2011 3:16 AM, Sherwood McGowan wrote: This may sound like a stupid question, but what are your verbosity and debug levels set at currently? nope, thats exactly the type of thing i'm wondering if i'm missing :) but, i tried with verbose 3/debug 0 (which worked in 1.6), and i also tried with verbose 10/debug 10 before posting. no dice. Ah right on mate! Glad to see that you checked it *and* didn't mind being asked (after all, we're all IT/VOIP professionals, and we all know the first thing to ask is the simplest possible solution ;-] ) Cheers! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sip error logging
bumping once before sending it to the tracker. Original Message Subject: [asterisk-users] sip error logging Date: Fri, 15 Apr 2011 03:39:23 -0400 I recently noticed that asterisk is not logging unknown sip connections. I'm not sure if I've broken something or if asterisk itself has been broken. the last entry I have is: /var/log/asterisk/messages.9:[Feb 10 23:58:48] NOTICE[9868] chan_sip.c: Registration from 'sip:22942@10.0.0.3' failed for '10.0.0.228:5060' - No matching peer found my logger.conf looks like: # grep -v '^;' /etc/asterisk/logger.conf [general] [logfiles] console = notice,warning,error,dtmf messages = notice,warning,error,verbose,dtmf,fax if i send 'options' or 'register' from a non-configured sip peer, i dont see anything in the log. am I missing something ? * i can replicate this behavior on 1.8.2.3 and 1.8.3.2 -- Jeremy Kister http://jeremy.kister.net./ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip error logging
I listened to your email using DriveCarefully and will respond as soon as I can. Download DriveCarefully for free at www.drivecarefully.com -- Sent from my BlackBerry® Senior Support Engineer US Numbers: 561-886-0664 Nicaragua Mobile: +505.8488.6876 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sip error logging
I recently noticed that asterisk is not logging unknown sip connections. I'm not sure if I've broken something or if asterisk itself has been broken. the last entry I have is: /var/log/asterisk/messages.9:[Feb 10 23:58:48] NOTICE[9868] chan_sip.c: Registration from 'sip:22942@10.0.0.3' failed for '10.0.0.228:5060' - No matching peer found my logger.conf looks like: # grep -v '^;' /etc/asterisk/logger.conf [general] [logfiles] console = notice,warning,error,dtmf messages = notice,warning,error,verbose,dtmf,fax if i send 'options' or 'register' from a non-configured sip peer, i dont see anything in the log. am I missing something ? -- Jeremy Kister http://jeremy.kister.net./ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip error logging
On 4/15/2011 3:39 AM, Jeremy Kister wrote: I recently noticed that asterisk is not logging unknown sip connections. I'm not sure if I've broken something or if asterisk itself has been broken. forgot to mention that I can replicate this behavior on 1.8.2.3 and 1.8.3.2 -- Jeremy Kister http://jeremy.kister.net./ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
