[asterisk-bugs] [Asterisk 0013235]: Memory leak in Asterisk 1.4 and Trunk

Asterisk Bug Tracker noreply at bugs.digium.com
Fri Sep 5 23:21:01 CDT 2008


A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=13235 
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Reported By:                falves11
Assigned To:                murf
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Project:                    Asterisk
Issue ID:                   13235
Category:                   Core/General
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     feedback
Asterisk Version:           SVN 
SVN Branch (only for SVN checkouts, not tarball releases):  trunk 
SVN Revision (number only!): 13058 
Disclaimer on File?:        N/A 
Request Review:              
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Date Submitted:             2008-08-04 18:31 CDT
Last Modified:              2008-09-05 23:20 CDT
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Summary:                    Memory leak in Asterisk 1.4 and Trunk
Description: 
Both versions of Asterisk have a huge memory leak. I thought that it was
Trunk only and ported my app to 1.4. After 1 day and 17 hours the memory
has gone up 1.2 GB. I only have 300 open calls. My machine is open for
inspection. I am not using "malloc debug" and "don't optimize", for
performance reasons,but I will restart tonight the server. if somebody
wants to suggest any diagnostic technique, please let me know before I
restart.
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Relationships       ID      Summary
----------------------------------------------------------------------
related to          0013409 [patch] Huge memory leak because memory...
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 (0092132) falves11 (reporter) - 2008-09-05 23:20
 http://bugs.digium.com/view.php?id=13235#c92132 
---------------------------------------------------------------------- 
I think that I found a probable cause for the memory growth. I have two
boxes, identical dialplan, version, etc. In one of them I type "sip show
channels" and I have 60 active channels. In the one where the memory goes
crazy I do the same and I get: 
7.110.179.253   207.2.123.  44938D3E-7A  00102/00102  0x0 (nothing)    No 
     Rx: BYE
67.110.179.253   207.2.123.  43589C8D-7A  00102/00102  0x0 (nothing)    No
      Rx: BYE
67.110.179.253   207.2.123.  3D4438C4-7A  00102/00102  0x0 (nothing)    No
 (d)  Rx: BYE
67.110.179.253   207.2.123.  375ED38F-7A  00102/00102  0x0 (nothing)    No
 (d)  Rx: BYE
67.110.179.253   207.2.123.  36CD8F87-7A  00102/00102  0x0 (nothing)    No
 (d)  Rx: BYE
67.110.179.253   207.2.123.  34A66BA1-7A  00102/00102  0x0 (nothing)    No
      Rx: BYE
5552 active SIP channels

Please notice the "(d)" in almost every line. It does not happen in the
first box. In the second box, with the 5552 active channels, I have only 9
active calls, while in the first one I have 30 open calls.

The second box receives calls only from a Cisco 7301, while the other one
receives calls from switches unknown. Maybe something on the SIP interop is
not understanding the Cisco dialogs and this the calls never close. Both
boxes generate CDR's successfully. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2008-09-05 23:20 falves11       Note Added: 0092132                          
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