[asterisk-bugs] [Asterisk 0012508]: handle_response_peerpoke floods asterisk cli with notices, cli crashes

[noreply at bugs.digium.com]

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=12508 
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Reported By:                kactus
Assigned To:                
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Project:                    Asterisk
Issue ID:                   12508
Category:                   Channels/chan_sip/Registration
Reproducibility:            always
Severity:                   crash
Priority:                   normal
Status:                     feedback
Asterisk Version:           SVN 
SVN Branch (only for SVN checkouts, not tarball releases): 1.6.0 
SVN Revision (number only!): 114601 
Disclaimer on File?:        N/A 
Request Review:              
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Date Submitted:             04-23-2008 22:35 CDT
Last Modified:              07-02-2008 02:51 CDT
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Summary:                    handle_response_peerpoke floods asterisk cli with
notices, cli crashes
Description: 
Since upgrading asterisk to svn later than 1.6 beta 8 cli is flooded with 

NOTICE[7435]: chan_sip.c:16045 handle_response_peerpoke: Peer 'PEERNAME'
is now Reachable. (71ms / 2000ms)

several times a second. Did not happen with svn releases around the time
of beta 6 and 7 aprox around april 17 (sorry didn't keep the old revision
numbers) or earlier revisions. 

disconnecting phone or rebooting causes:
[Apr 24 11:22:31] NOTICE[7435]: chan_sip.c:19841 sip_poke_noanswer: Peer
'DRA-997896-0043' is now UNREACHABLE!

being spammed in the same fashion.

after running it for a while we end up seeing

[root at voip02 asterisk]# /usr/sbin/safe_asterisk: line 117:  7427
Segmentation fault      (core dumped) nice -n $PRIORITY
${ASTSBINDIR}/asterisk -f ${CLIARGS} ${ASTARGS} >&/dev/${TTY} </dev/${TTY}
Asterisk ended with exit status 139
Asterisk exited on signal 11.
Automatically restarting Asterisk.
mpg123: no process killed

I'll need to recompile for "don't optimise" so I haven't attached a core
debug, let me know if you need it. I will attached the sip info with sip
debug as well.
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---------------------------------------------------------------------- 
 kactus - 07-02-08 02:51  
---------------------------------------------------------------------- 
Hi Corydon

The SVN-trunk-r126681 box was the first to stop spamming the screen with
output. core show locks shows nothing locked. Rebooting the phone with sip
debugging on shows that it is still recieving sip registers from the phone
but it throws an error that there is "No database handle available with the
name DBNAME"

odbc show all reports that all 5 connections are "in use".

Top shows:
  PID USER      PR  NI  VIRT  RES  SHR S %CPU %MEM    TIME+  COMMAND
30425 root      15   0  700m 133m 4236 S 99.9 52.2  88:32.90 memcheck


The asterisk 1.4.21.1 looks to be not responding at all but sip set debug
shows that it is constantly reliably transmitting nat CSeq 102 options 

It however only has one db connection in use but the phone is unable to
register (reports Network Failure and timeout in the logs) and sip trace
just shows it send and recieve 102 options.

The 1.4 box did however have these locks:

=== Thread ID: 147204416 (do_monitor           started at [15834]
chan_sip.c restart_monitor())
=== ---> Lock http://bugs.digium.com/view.php?id=0 (chan_sip.c): MUTEX 15778
do_monitor &monlock 0x8a1e740
(1)
=== ---> Lock http://bugs.digium.com/view.php?id=1 (sched.c): MUTEX 223
ast_sched_add_variable &con->lock
0x6ced22c (1)


let me know if you want me to give you anything else or if stopping
asterisk now to get you the valgrind output is sufficient.

Tried to catch you on the #asterisk-bugs but looks like you weren't
reading it. I'll be hanging around for another 45 min so if you would like
me to test anything for you before tommorow, please let me know.

Cheers - Kactus 

Issue History 
Date Modified   Username       Field                    Change               
====================================================================== 
07-02-08 02:51  kactus         Note Added: 0089575                          
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