[asterisk-bugs] [Asterisk 0017699]: user get unreachable after some minutes, deadlock in ao2_lock

[Asterisk Bug Tracker]

The following issue requires your FEEDBACK. 
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https://issues.asterisk.org/view.php?id=17699 
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Reported By:                schmidts
Assigned To:                
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Project:                    Asterisk
Issue ID:                   17699
Category:                   Channels/chan_sip/General
Reproducibility:            sometimes
Severity:                   minor
Priority:                   normal
Status:                     feedback
Asterisk Version:           1.6.2.10 
JIRA:                       SWP-1949 
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
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Date Submitted:             2010-07-24 17:30 CDT
Last Modified:              2010-09-14 15:52 CDT
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Summary:                    user get unreachable after some minutes, deadlock in
ao2_lock
Description: 
i´ve tried the second time to upgrade from 1.2.40 to 1.6.2.9 / 10 and
after 10 to 15 minutes it looks like asterisk couldnt send any sip packets
out.
i thought that the last revision r278465 could also solve this problem so
i´ve tried 1.6.2.10 today but i got the same problem.

i have around 3100 peers in static config files and normally around 2100
are registered and reachable in asterisk. there are also around 1300 hints
in the extensions.conf and more than 1300 subcribes on that hint.
when i restart asterisk, the peers get the qualify packet and are
reachable but after some time, i see more and more of these messages:
 == Extension Changed userB[outcust] new state Idle for Notify User userA
(queued)
also the phones cant reach asterisk any more and show they are offline,
and in asterisk i can see them as unreachable.
with 1.6.2.9 after 1 hour only 100 phones has keep rechable and in
1.6.2.10 only 1000 after 30 mins.

i can also see that their are many (>500) open sip channels which i dont
see in 1.2.40
this server is a hosted pbx solution for our customers and when i´ve
tested this there is no phone traffic on the server just registrations and
subscribes.
with load this server handles > 200 concurrent calls without any problems.
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Relationships       ID      Summary
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related to          0017741 [patch] sip_poke_noanswer launch ast_de...
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 (0126961) russell (administrator) - 2010-09-14 15:52
 https://issues.asterisk.org/view.php?id=17699#c126961 
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Now that we put in the astdb improvement, is this still an issue we should
keep open? 

Issue History 
Date Modified    Username       Field                    Change               
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2010-09-14 15:52 russell        Note Added: 0126961                          
2010-09-14 15:52 russell        Status                   acknowledged =>
feedback
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