[asterisk-bugs] [Asterisk 0015627]: [patch] Asterisk runs out of sockets

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=15627 
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Reported By:                falves11
Assigned To:                
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Project:                    Asterisk
Issue ID:                   15627
Category:                   Core/Netsock
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     ready for review
Asterisk Version:           SVN 
JIRA:                       SWP-255 
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): 1.6.2 
SVN Revision (number only!): 209626 
Request Review:              
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Date Submitted:             2009-07-31 17:12 CDT
Last Modified:              2009-11-06 07:08 CST
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Summary:                    [patch] Asterisk runs out of sockets
Description: 
The Parallels engineers have found a bug that takes down asterisk because
the server runs out of sockets, and also it degrades the performance
because over time it takes more and more time for the processor to find an
empty socket. The load on the processor grows over time,

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Relationships       ID      Summary
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related to          0015716 [patch] chan_sip fails to destroy chann...
child of            0015356 After a few thousand calls, or at rando...
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 (0113273) atis (reporter) - 2009-11-06 07:08
 https://issues.asterisk.org/view.php?id=15627#c113273 
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I was running tests and I discovered the following errors, which probably
indicate this issue:

    -- Executing [8006772943 at did-dial:1]
Dial("Local/8006772943 at did-dial-3c2e;2", "SIP/8006772943 at external_calls")
in new stack
  == Using SIP RTP CoS mark 5
[Nov  6 03:26:51] ERROR[5968]: acl.c:472 ast_ouraddrfor: Cannot create
socket
[Nov  6 03:26:51] WARNING[5968]: channel.c:830 __ast_channel_alloc_ap:
Channel allocation failed: Can't create alert pipe!
[Nov  6 03:26:51] WARNING[5968]: chan_sip.c:5870 sip_new: Unable to
allocate AST channel structure for SIP channel
[Nov  6 03:26:51] WARNING[5968]: app_dial.c:1528 dial_exec_full: Unable to
create channel of type 'SIP' (cause 0 - Unknown)
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [8006772943 at did-dial:2]
Hangup("Local/8006772943 at did-dial-3c2e;2", "") in new stack
  == Spawn extension (did-dial, 8006772943, 2) exited non-zero on
'Local/8006772943 at did-dial-3c2e;2'
[Nov  6 03:26:51] NOTICE[5967]: pbx_spool.c:338 attempt_thread: Call
failed to go through, reason (1) Hangup


I ported proposed patch to 1.6.1 branch, and I'm starting to test 

Issue History 
Date Modified    Username       Field                    Change               
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2009-11-06 07:08 atis           Note Added: 0113273                          
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