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Summary:ASTERISK-25348: Asterisk Crashes after caller records name with the Privacy Options
Reporter:Pramod Venugopal (pramsky)Labels:
Date Opened:2015-08-26 14:08:02Date Closed:2020-01-14 11:13:43.000-0600
Priority:MajorRegression?
Status:Closed/CompleteComponents:Applications/app_dial Applications/app_privacy
Versions:12.8.0 12.8.1 Frequency of
Occurrence		Frequent
Related
Issues:
Environment:Centos 6.6. Hosted on Amazon EC2 Asterisk 12.8 with FreePBX12 Attachments:( 0) backtrace.txt
( 1) crashlog.txt
( 2) extensions_additional.conf
( 3) extensions.conf
( 4) sip_additional.conf
( 5) sip_general_additional.conf
( 6) sip.conf
Description:Every so often, when a call comes in to an account with Privacy Screening, after the caller records their name, the phone rings and is answered. When the recorded caller name is played to the user, asterisk crashes.

The log /var/log/asterisk/full shows around 1.6million lines for that call with the following:

"[2015-08-26 09:13:38] ERROR[18320][C-00000469] astobj2.c: bad magic number for object 0x7fd1acb994e8. Object is likely destroyed."

Just prior to this are the following lines:

[2015-08-26 09:13:20] VERBOSE[18320][C-00000469] app_dial.c:     -- SIP/709-00000605 is ringing
[2015-08-26 09:13:25] VERBOSE[18320][C-00000469] app_dial.c:     -- SIP/709-00000605 answered SIP/from-trunk-00000604
[2015-08-26 09:13:25] VERBOSE[18320][C-00000469] file.c:     -- <SIP/709-00000605> Playing 'priv-callpending.ulaw' (language 'en')
[2015-08-26 09:13:30] VERBOSE[18320][C-00000469] file.c:     -- <SIP/709-00000605> Playing 'priv-callerintros/2066337888.slin' (language 'en')
[2015-08-26 09:13:31] VERBOSE[18320][C-00000469] file.c:     -- <SIP/709-00000605> Playing 'screen-callee-options.ulaw' (language 'en')

I have the full call log ( without the 1.6 million astobj2.c ). I also have a backtrace setup , but since I am using the official RPM, its not built with DONT_OPTIMIZE.
Comments:By: Asterisk Team (asteriskteam) 2015-08-26 14:08:03.968-0500

Thanks for creating a report! The issue has entered the triage process. That means the issue will wait in this status until a Bug Marshal has an opportunity to review the issue. Once the issue has been reviewed you will receive comments regarding the next steps towards resolution.

A good first step is for you to review the [Asterisk Issue Guidelines|https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines] if you haven't already. The guidelines detail what is expected from an Asterisk issue report.

Then, if you are submitting a patch, please review the [Patch Contribution Process|https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process].
By: Rusty Newton (rnewton) 2015-08-28 08:51:58.557-0500

Thanks for the report! Please attach the sip.conf and extensions.conf needed to reproduce the issue.
By: Pramod Venugopal (pramsky) 2015-09-04 12:57:41.452-0500

Sorry for responding so late, the sip.conf and extensions.conf are from freepbx. Adding sip.conf, sip_additional.conf, extensions.conf and extensions_additional.conf
By: Rusty Newton (rnewton) 2015-09-04 18:21:00.739-0500

To hunt down the issue we may need a reference count log.

Is there any way you could compile from source such that you could compile with REF_DEBUG ?

https://wiki.asterisk.org/wiki/display/AST/Reference+Count+Debugging
By: Rusty Newton (rnewton) 2015-09-04 18:29:38.782-0500

Of course - if you are able to go to that trouble then you might as well get a backtrace where you compiled with DONT_OPTIMIZE as well..
By: Asterisk Team (asteriskteam) 2015-09-19 12:00:20.979-0500

Suspended due to lack of activity. This issue will be automatically re-opened if the reporter posts a comment. If you are not the reporter and would like this re-opened please create a new issue instead. If the new issue is related to this one a link will be created during the triage process. Further information on issue tracker usage can be found in the Asterisk Issue Guidlines [1].

[1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines