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Summary:ASTERISK-07904: Notice: sched.c:296
Reporter:nguyencongtriet (nguyencongtriet)Labels:
Date Opened:2006-10-09 09:56:10Date Closed:2006-11-15 08:02:55.000-0600
Priority:CriticalRegression?No
Status:Closed/CompleteComponents:Core/General
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:
Description:Hi,
I'm using CentOS 4.4 and Asterisk 1.2.12, PostgreSQL  database, Zap: 1.2.9.1

The server log this:
sched.c:296 ast_sched_del attempted to del noexistent schedule entry 7744!

Then asterisk service is stop. How can I fix this?

Also, How can I update from CVS if the fixed is in CVS.

Thanks
Comments:By: nguyencongtriet (nguyencongtriet) 2006-10-09 09:58:27

Please help me how to install the patch to the Asterisk also,

Thanks
By: Anthony LaMantia (alamantia) 2006-10-09 15:19:07

you are going to need to supply us with a full backtrace from a copy of asterisk built with make don't optmisize as well as any relevment console logs to help us debug this problem..

if you have a patch, you just run

patch -p0 < patch.diff  to install it.
By: nguyencongtriet (nguyencongtriet) 2006-10-09 22:07:05

Hi,

I'm a newbie in Asterisk
Server console log

sched.c:296 ast_sched_del attempted to del noexistent schedule entry 3941!

Then asterisk is disconnected, asterisk service is stop.
I'm just building the Asterisk by copy the Asterisk 1.2.12 from the digium website and build without "don't optmisize"

My Client softphone is Express Talk 1.03

Adding info when I set debug is 5:
chan_sip.c: 11644 sip_poke_peer: still have qualify dialog active, deleting

And disconnect server.
By: Anthony LaMantia (alamantia) 2006-10-10 12:40:46

please see this

http://www.digium.com/bugguidelines.html
By: nguyencongtriet (nguyencongtriet) 2006-10-10 22:54:20

"disconnect problem": i can not reproduce this trouble.
Only this info for me now:
chan_sip.c: 11644 sip_poke_peer: still have qualify dialog active, deleting

then stop everything.

I use CenOS 4.4, Asterisk 1.2.12, no SVN, trunk.
This problem only have when I have sip client connect to

Please help
By: Anthony LaMantia (alamantia) 2006-10-11 11:29:49

I see, in that case we are going to need you to provide as much of a console debug log as possible so we can pinpoint which portion of the code is causing the trouble for you and what leads up to it.

do you still have an interactive cli? durring the disconnect problem?
or is asterisk just freezing on you?
By: Anthony LaMantia (alamantia) 2006-10-30 12:56:40.000-0600

is this issue still effecting you using the letest asterisk 1.2 release?
By: Anthony LaMantia (alamantia) 2006-10-30 12:58:03.000-0600

1) Prepare test environment (reduce the amount of unrelated traffic on the server);
2) Make sure your logger.conf has the following line:
  console => notice,warning,error,debug
3) restart Asterik.
4) Enable SIP transaction logging with the following CLI commands:
set debug 4
set verbose 4
sip debug
5) Save complete console log to file and _attach_ said file to the
By: Serge Vecher (serge-v) 2006-10-30 13:18:12.000-0600

fyi: the latest asterisk release is 1.2.13.
By: Anthony LaMantia (alamantia) 2006-11-09 11:49:24.000-0600

nguyencongtriet any updates ?
By: Serge Vecher (serge-v) 2006-11-15 08:02:55.000-0600

no answer from the original poster