[asterisk-bugs] [Asterisk 0015848]: [patch] segfault when transferring a queue caller

Asterisk Bug Tracker noreply at bugs.digium.com
Tue Nov 3 01:41:55 CST 2009


A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=15848 
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Reported By:                fhackenberger
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   15848
Category:                   Channels/General
Reproducibility:            random
Severity:                   crash
Priority:                   normal
Status:                     ready for review
Asterisk Version:           Older 1.6.1 
JIRA:                       SWP-327 
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): 1.6.1 
SVN Revision (number only!): 209515 
Request Review:              
====================================================================== 
Date Submitted:             2009-09-08 03:55 CDT
Last Modified:              2009-11-03 01:41 CST
====================================================================== 
Summary:                    [patch] segfault when transferring a queue caller
Description: 
The segfault happens after transferring (SIP re-invite) a queue call.

https://issues.asterisk.org/view.php?id=0  0xb7d5da04 in
__pthread_mutex_unlock_usercnt () from
/lib/tls/i686/cmov/libpthread.so.0
https://issues.asterisk.org/view.php?id=0  0xb7d5da04 in
__pthread_mutex_unlock_usercnt () from
/lib/tls/i686/cmov/libpthread.so.0
https://issues.asterisk.org/view.php?id=1  0xb61ef779 in sip_hangup
(ast=0xb3902a20) at
/tmp/buildd/asterisk-1.6.1.0~dfsg/include/asterisk/lock.h:1708
https://issues.asterisk.org/view.php?id=2  0x08087eb9 in ast_do_masquerade
(original=0x8722db8) at channel.c:4406
https://issues.asterisk.org/view.php?id=3  0x0808b589 in ast_waitfor_nandfds
(c=0xb41fa200, n=2, fds=0x0, nfds=0,
exception=0x0, outfd=0x0, ms=0xb41fa214) at channel.c:2004
https://issues.asterisk.org/view.php?id=4  0x0808bb4f in ast_waitfor_n
(c=0xb41fa200, n=2, ms=0xb41fa214) at
channel.c:2318
https://issues.asterisk.org/view.php?id=5  0x0808e2d4 in ast_channel_bridge
(c0=0x8722db8, c1=0x87407f8,
config=0xb41fa61c, fo=0xb41fa2ec, rc=0xb41fa2e8) at channel.c:4733
https://issues.asterisk.org/view.php?id=6  0x080b06b6 in ast_bridge_call
(chan=0x8722db8, peer=0x87407f8,
config=0xb41fa61c) at features.c:2538
https://issues.asterisk.org/view.php?id=7  0xb5bfa2e9 in dial_exec_full
(chan=0x8722db8, data=0xb41fd138,
peerflags=0xb41fae90, continue_exec=0x0) at app_dial.c:1986
https://issues.asterisk.org/view.php?id=8  0xb5bfc174 in dial_exec
(chan=0x8722db8, data=0xb41fd138) at
app_dial.c:2060
https://issues.asterisk.org/view.php?id=9  0x080ddab5 in pbx_exec (c=0x8722db8,
app=0x82b0d00, data=0xb41fd138)
at pbx.c:960
https://issues.asterisk.org/view.php?id=10 0x080e4632 in pbx_extension_helper
(c=0x8722db8, con=0x0,
context=0x8723028 "agents", exten=0x8723078 "972605572372450", priority=9,
    label=0x0, callerid=0x848d910 "0314224713906", action=E_SPAWN,
found=0xb41ff244, combined_find_spawn=1) at pbx.c:3201
https://issues.asterisk.org/view.php?id=11 0x080e6ecc in __ast_pbx_run
(c=0x8722db8, args=0x0) at pbx.c:3745
https://issues.asterisk.org/view.php?id=12 0x080e89d0 in pbx_thread
(data=0x8722db8) at pbx.c:4032
https://issues.asterisk.org/view.php?id=13 0x0811ca47 in dummy_start
(data=0x86be7f8) at utils.c:968
https://issues.asterisk.org/view.php?id=14 0xb7d5a4fb in start_thread () from
/lib/tls/i686/cmov/libpthread.so.0
https://issues.asterisk.org/view.php?id=15 0xb6f7be5e in clone () from
/lib/tls/i686/cmov/libc.so.6

====================================================================== 

---------------------------------------------------------------------- 
 (0113026) TomS (reporter) - 2009-11-03 01:41
 https://issues.asterisk.org/view.php?id=15848#c113026 
---------------------------------------------------------------------- 
fhackenberger: Our new problem with the lack of incomming SIP connections
was not related with your patch. After downgrading from 1.6.1.6 (with your
patch) to 1.6.0.15 and then to 1.6.0.13 (because of hint-issues with .15) I
didnt apply your patch since there is no known issue with transfers on the
1.6.0-tree.
The new problem with no sip connections reapeard on the downgraded system
too.
Yesterday I replaced the asterisk-hardware and switch and downgraded to
1.4.26.2 to get hopefully finally rid of all problems.

I would have liked to continue testing your patch which certenly changed
the behavier of 1.6.1.6 on transfers in a positive way, but the productive
system was to important for longer tests.
My other (testing) platform with 1.6.1.4 (without patch) hadn't any
problems with transfers yet but on the other hand we use the
transfer-function not very often there.

I just want to repeat: my problem with the missing incomming sip
connections was not related to your patch. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-11-03 01:41 TomS           Note Added: 0113026                          
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