[asterisk-bugs] [Asterisk 0016608]: [patch] Deadlock on &(&channels)->lock

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=16608 
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Reported By:                sergee
Assigned To:                tilghman
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Project:                    Asterisk
Issue ID:                   16608
Category:                   Channels/General
Reproducibility:            random
Severity:                   major
Priority:                   normal
Status:                     assigned
Target Version:             1.6.0.27
Asterisk Version:           SVN 
JIRA:                       SWP-732 
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): 1.6.0 
SVN Revision (number only!): 237060 
Request Review:              
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Date Submitted:             2010-01-14 15:04 CST
Last Modified:              2010-03-28 03:45 CDT
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Summary:                    [patch] Deadlock on &(&channels)->lock
Description: 
I've got 2 deadlocks in 3 days. It happens in a peak hours, with more then
a hundred calls in the system.

I've got 2 files 1 with "core show locks" - from the first deadlock (the
day before yesterday), second file - with output from gdb (intho thread,
thread apply all bt, thread apply all bt full) - from today's deadlock.
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 (0119964) sergee (reporter) - 2010-03-28 03:45
 https://issues.asterisk.org/view.php?id=16608#c119964 
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Now i have false-positives for rtp-timeout timers,

but rtp is works,

    -- Executing DIAL("SIP/1010-000017a4", "SIP/voipgw4/........")
  == Using SIP RTP CoS mark 5
  == Using SIP VRTP CoS mark 6
  == Using UDPTL CoS mark 5
    -- Called voipgw4/09689262415703
    -- SIP/voipgw4-000017a5 is making progress passing it to
SIP/1010-000017a4
    -- SIP/voipgw4-000017a5 is making progress passing it to
SIP/1010-000017a4
    -- SIP/voipgw4-000017a5 answered SIP/1010-000017a4
    -- Packet2Packet bridging SIP/1010-000017a4 and SIP/voipgw4-000017a5

Disconnecting call 'SIP/1010-000017a4' for lack of RTP activity in 61
seconds



I've checked "core show channel SIP/1010-000017a4" several times during a
call. It seems that Asterisk stops updating packet counters (Frames in,
Frames out) after a few seconds of call. 

Issue History 
Date Modified    Username       Field                    Change               
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2010-03-28 03:45 sergee         Note Added: 0119964                          
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