[asterisk-bugs] [Asterisk 0018065]: sometimes crash after Unable to write to alert pipe
Asterisk Bug Tracker
noreply at bugs.digium.com
Tue Sep 28 06:35:12 CDT 2010
A NOTE has been added to this issue.
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https://issues.asterisk.org/view.php?id=18065
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Reported By: wdoekes
Assigned To:
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Project: Asterisk
Issue ID: 18065
Category: Core/Channels
Reproducibility: sometimes
Severity: minor
Priority: normal
Status: new
Asterisk Version: 1.6.2.14-rc1
JIRA:
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: 2010-09-28 03:02 CDT
Last Modified: 2010-09-28 06:35 CDT
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Summary: sometimes crash after Unable to write to alert pipe
Description:
Hi,
I was updating my dialplan and running some tests calls, and suddenly
asterisk started crashing. I'm guessing the dialplan change did something,
but I can't figure out what yet.
I won't be posting the entire dialplan, and I have yet to distill it down
to something small that still crashes.
I've cleaned asterisk of all my patches and had both 1.6.2.8 and
1.6.2.14-rc1 crash with MALLOC_DEBUG and DONT_OPTIMIZE on. (Oddly enough,
it never core dumps, so I have to run it under gdb to get proper info.)
======================================================================
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(0127453) wdoekes (reporter) - 2010-09-28 06:35
https://issues.asterisk.org/view.php?id=18065#c127453
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Okay, it seems that the Pipe warning is not the cause but a side-effect.
I've also figured out that the warnings (WARNING: Freeing unused memory at
0x122ec08, in __ast_module_user_remove of loader.c, line 222) is probably
very serious. (Shouldn't this be an ERROR?)
I've reverted my dialplan to before my most recent edits and I can still
get asterisk to crash (currently running an unpatched 1.6.2.14-rc1). But
it's hard to reproduce. What I do is call the same number quite a number of
times and hang up pretty quickly. Sometimes the phone is answered in time,
sometimes it isn't.
After every call, the used-modules-list is always set to 0, except after
one of these errors:
*CLI> module show
...
chan_local.so Local Proxy Channel (Note: used internal 0
...
-- Executing [+310 at voipgrid_out:1] Dial("Local/+310 at world_out-f7f7;2",
"SIP/+310 at 126680007") in new stack
== Using SIP RTP CoS mark 5
== Using UDPTL CoS mark 5
-- Called +310 at 126680007
-- Called +310 at world_out
-- Local/ID2 at osvpi_route_fixeddestination-a20c;1 is ringing
-- Local/+311 at osvpi_world-7df6;1 is ringing
-- SIP/126680007-00000031 answered Local/+310 at world_out-f7f7;2
-- Local/+310 at world_out-f7f7;1 answered
Local/ID2 at osvpi_route_fixeddestination-a20c;2
-- Local/ID2 at osvpi_route_fixeddestination-a20c;1 stopped sounds
-- Local/ID2 at osvpi_route_fixeddestination-a20c;1 answered
Local/+311 at osvpi_world-7df6;2
-- Local/+311 at osvpi_world-7df6;1 stopped sounds
-- Local/+311 at osvpi_world-7df6;1 answered SIP/126680002-00000030
WARNING: Freeing unused memory at 0x1a9f658, in __ast_module_user_remove
of loader.c, line 222
[2010-09-28 13:21:58] WARNING[27190]: channel.c:1072 __ast_queue_frame:
Unable to write to alert pipe on Local/+310 at world_out-f7f7;1<ZOMBIE> (qlen
= 0): Bad file descriptor!
== Spawn extension (voipgrid_out, +310, 1) exited non-zero on
'Local/+310 at world_out-f7f7;2'
== Spawn extension (osvpi_world_real_call, +310, 3) exited non-zero on
'Local/ID2 at osvpi_route_fixeddestination-a20c;2'
-- Packet2Packet bridging SIP/126680002-00000030 and
SIP/126680007-00000031
== Spawn extension (osvpi_route_fixeddestination_step, +311, 1) exited
non-zero on 'Local/+311 at osvpi_world-7df6;2'
== Spawn extension (osvpi_world_call, +311, 7) exited non-zero on
'SIP/126680002-00000030'
*CLI> module show
...
chan_local.so Local Proxy Channel (Note: used internal -1
...
Now I can call the number some more and hang up quickly, but if the other
side picks up, the crash happens.
Sometimes I don't get a second chance and it crashes immediately, like
this:
-- Executing [+310 at voipgrid_out:1] Dial("Local/+310 at world_out-0efe;2",
"SIP/+310 at 126680007") in new stack
== Using SIP RTP CoS mark 5
== Using UDPTL CoS mark 5
-- Called +310 at 126680007
-- Called +310 at world_out
-- Local/ID2 at osvpi_route_fixeddestination-8a76;1 is ringing
-- Local/+311 at osvpi_world-5a84;1 is ringing
-- SIP/126680007-0000000f answered Local/+310 at world_out-0efe;2
-- Local/+310 at world_out-0efe;1 answered
Local/ID2 at osvpi_route_fixeddestination-8a76;2
-- Local/ID2 at osvpi_route_fixeddestination-8a76;1 stopped sounds
-- Local/+311 at osvpi_world-5a84;1 stopped sounds
-- Local/ID2 at osvpi_route_fixeddestination-8a76;1 answered
Local/+311 at osvpi_world-5a84;2
-- Local/+311 at osvpi_world-5a84;1 answered SIP/126680002-0000000e
WARNING: Freeing unused memory at 0x153b808, in __ast_module_user_remove
of loader.c, line 222
== Spawn extension (osvpi_world_real_call, +310, 3) exited non-zero on
'Local/ID2 at osvpi_route_fixeddestination-8a76;2'
WARNING: Freeing unused memory at 0x153ab48, in local_pvt_destroy of
chan_local.c, line 240
Segmentation fault
Issue History
Date Modified Username Field Change
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2010-09-28 06:35 wdoekes Note Added: 0127453
======================================================================
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