[asterisk-bugs] [JIRA] (ASTERISK-24768) res_timing_pthread: file descriptor leak

Private Name (JIRA) noreply at issues.asterisk.org
Sun Feb 22 08:25:34 CST 2015


    [ https://issues.asterisk.org/jira/browse/ASTERISK-24768?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=225053#comment-225053 ] 

Private Name edited comment on ASTERISK-24768 at 2/22/15 8:24 AM:
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The problem is back. I am using SIP, not PJSIP. With 220 open calls, playing tons of voice prompts, it is a free roulette over the phone, you may test it 1 573 764 2870 (free free free)
 lsof | grep asterisk | grep FIFO | wc -l
646110
after like 1/2 hour.

Note: SVN-branch-11-r432098M has the same issue, so this is a regression issue.
I think this may be a new leak, unknown so far. I went back to rev SVN-branch-13-r431807M and with 120 calls it shows
lsof | grep asterisk| grep FIFO | wc -l
190950
The only thing I do is playing prompts and constantly reading variables
Note:
This is turning simpler. In a completely empty, not even a single line, dialplan, with 

SIP/demo-0000037d    (None)               Up      Echo()
SIP/demo-0000037e    (None)               Up      Echo()
SIP/demo-0000037f    (None)               Up      Echo()
501 active channels
0 active calls
0 calls processed
lsof | grep asterisk| grep FIFO | wc -l
1028105

This is SVN-branch-13-r432154M
Is this normal?
By the way, there are no codecs loaded that do not come with Asterisk.
All the channels are originated using a call file and the app Echo.
Can somebody give instructions as to how to debug this inside gdb?




 


was (Author: falves11):
The problem is back. I am using SIP, not PJSIP. With 220 open calls, playing tons of voice prompts, it is a free roulette over the phone, you may test it 1 573 764 2870 (free free free)
 lsof | grep asterisk | grep FIFO | wc -l
646110
after like 1/2 hour.

Note: SVN-branch-11-r432098M has the same issue, so this is a regression issue.
I think this may be a new leak, unknown so far. I went back to rev SVN-branch-13-r431807M and with 120 calls it shows
lsof | grep asterisk| grep FIFO | wc -l
190950
The only thing I do is playing prompts and constantly reading variables
Note:
This is turning simpler. In a completely empty, not even a single line, dialplan, with 

235 active channels
0 active calls
0 calls processed
lsof | grep asterisk| grep FIFO | wc -l
240080
This is SVN-branch-13-r432154M
Is this normal?
By the way, there are no codecs loaded that do not come with Asterisk.
All the channels are originated using a call file and the app Echo.
Can somebody give instructions as to how to debug this inside gdb?




 

> res_timing_pthread: file descriptor leak
> ----------------------------------------
>
>                 Key: ASTERISK-24768
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-24768
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Resources/res_timing_pthread
>    Affects Versions: 13.2.0
>         Environment: Current Debian (jessie/testing), i386, up-to-date
>            Reporter: Matthias Urlichs
>            Assignee: Joshua Colp
>         Attachments: timer.patch
>
>
> Pthread timers are never deallocated because their link into the pthread_timers chain is never undone.
> This causes a file descriptor leak (at least two per incoming call).
> The locking in this patch probably needs review; the ao2_unlink() call does not. :-P
> \[Edit:\] *Inline patch removed*



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