|Summary:||ASTERISK-18650: Asterisk hangs after failed directed call pickup attempt, logs show "Fixup failed on channel SIP/xxx, strange things may happen."|
|Reporter:||M. Anderson (adv99)||Labels:|
|Date Opened:||2011-09-30 10:45:43||Date Closed:||2011-10-04 10:51:43|
|Environment:||FreePBX Distro (CentOS 5.5, FreePBX 2.9, Asterisk 126.96.36.199)||Attachments:|
|Description:||This morning, I was testing my system. I called my own DID, which does not ring at my phone. I hung-up the call and almost immediately, inadvertently pushed a button to do a directed call pickup of one of the ringing extensions. The pick-up failed, since the call had terminated. Shortly thereafter, I noticed that all phones showed no service and Asterisk had crashed.|
Immediately before the crash, the log shows two curious entries:
[2011-09-30 07:51:46] WARNING channel.c: Fixup failed on channel SIP/44-xxx, strange things may happen.
[2011-09-30 07:51:46] WARNING channel.c: Hangup failed! Strange things may happen!
When I looked at my FreePBX status screen, it showed 1 active channel (more precisely 1 of 0), but no calls in progress.
My concern might be related to this one:
|Comments:||By: Leif Madsen (lmadsen) 2011-10-03 12:00:15.669-0500|
Thank you for your bug report. In order to move your issue forward, we require a backtrace from the core file produced after the crash. Also, be sure you have DONT_OPTIMIZE enabled in menuselect within the Compiler Flags section, then:
After enabling, reproduce the crash, and then execute the backtrace instructions. When complete, attach that file to this issue report.
By: M. Anderson (adv99) 2011-10-03 12:04:16.109-0500
I have no means to reproduce this issue.
Also, I'm not sufficiently familar with Asterisk to perform the tasks you've requested without more detailed instructions. All of my interaction with asterisk occurs either via FreePBX or reading the "full" logs at /var/asterisk/logs.
By: Leif Madsen (lmadsen) 2011-10-03 12:53:30.407-0500
I've already linked a page on how to provide a backtrace. If you can't reproduce the issue or provide the required backtrace information I'll have to close this issue as there is not anything a developer can do here as it stands.
By: Leif Madsen (lmadsen) 2011-10-04 10:51:05.850-0500
Thank you for taking the time to report this bug and helping to make Asterisk better. Unfortunately, we cannot work on this bug because your description did not include enough information. You may find it helpful to read the Asterisk Issue Guidelines http://www.asterisk.org/developers/bug-guidelines. We would be grateful if you would then provide a more complete description of the problem. At a minimum, we need:
1. the specific steps or actions you took that caused you to encounter the problem,
2. the behavior you expected, and
3. the behavior you actually encountered (in as much detail as possible).
This likely includes output from the console with debug level logging, a SIP trace (if this is SIP related), and configuration information such as dialplan (e.g. extensions.conf) and channel configuration (e.g. sip.conf). Thanks!
By: Leif Madsen (lmadsen) 2011-10-04 10:51:43.980-0500
Suspended for now. Please reopen or ask a bug marshal to reopen for you if you're able to provide the required information. Thanks!