[asterisk-bugs] [Asterisk 0016382]: SIP OPTIONS qualify message forever

Asterisk Bug Tracker noreply at bugs.digium.com
Mon Mar 22 10:42:37 CDT 2010


A NOTE has been added to this issue. 
====================================================================== 
https://issues.asterisk.org/view.php?id=16382 
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Reported By:                lftsy
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   16382
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     acknowledged
Asterisk Version:           SVN 
JIRA:                       SWP-478 
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:             2009-12-03 10:04 CST
Last Modified:              2010-03-22 10:42 CDT
====================================================================== 
Summary:                    SIP OPTIONS qualify message forever
Description: 
Hello, I have a trouble with different Asterisk versions (1.4.26, 1.4.27,
1.4.27.1). When I use the steps below, Asterisk starts to send SIP OPTIONS
to the previous IP/port used by a SIP realtime peer (that has been pruned)
and will keep trying to send SIP OPTIONS pings forever, event if the peer
is connected with a new IP/port address.

I have just checked with my old Asterisk 1.2.27 with the same sip.conf and
I do not have the problem, the SIP OPTIONS stops once register timer has
expired.

During my experience to reproduce the bug, I have been able to have 10
IP/port currently pinged by the Asterisk server for one single peer.
And the only way to stop it is to restart Asterisk...

Thank you for your attention!
Marc Leurent
======================================================================
Relationships       ID      Summary
----------------------------------------------------------------------
duplicate of        0016764 Sip Channels Colapse
related to          0015716 [patch] chan_sip fails to destroy chann...
related to          0015627 [patch] Asterisk runs out of sockets
====================================================================== 

---------------------------------------------------------------------- 
 (0119650) zerohalo (reporter) - 2010-03-22 10:42
 https://issues.asterisk.org/view.php?id=16382#c119650 
---------------------------------------------------------------------- 
Here's another one (1.4.27.1). Full chan_sip deadlock, forced a crash to
get a BT. Note that the 'core show locks' is slightly different:

=======================================================================
=== Currently Held Locks ==============================================
=======================================================================
===
=== <file> <line num> <function> <lock name> <lock addr> (times locked)
===
=== Thread ID: -1208841312 (do_devstate_changes started at [ 363]
devicestate.c ast_device_state_engine_init())
=== ---> Lock https://issues.asterisk.org/view.php?id=0 (chan_sip.c): MUTEX 2805
find_peer &(&peerl)->_lock
0x193220 (1)
=== -------------------------------------------------------------------
===
=== Thread ID: 1924000 (do_monitor started at [17041]
chan_sip.c restart_monitor())
=== ---> Lock https://issues.asterisk.org/view.php?id=0 (chan_sip.c): MUTEX
16987 do_monitor &monlock 0x190f60
(1)
=== ---> Lock https://issues.asterisk.org/view.php?id=1 (sched.c): MUTEX 220
ast_sched_add_variable &con->lock
0x96b8a58 (1)
=== -------------------------------------------------------------------
===
=======================================================================

Thread 41 (process 6583):
https://issues.asterisk.org/view.php?id=0 0x0810a26a in schedule (con=0x96b8a58,
s=0xb3b73430) at sched.c:178
https://issues.asterisk.org/view.php?id=1 0x0810a488 in ast_sched_add_variable
(con=0x96b8a58, when=1000,
callback=0x125f60 <retrans_pkt>, data=0xb6a06e90, variable=1) at
sched.c:231
https://issues.asterisk.org/view.php?id=2 0x00127770 in __sip_reliable_xmit
(p=0xb67168d8, seqno=102, resp=0, 
data=0x1d40cc "OPTIONS sip:2XX.XX.XXX.XXX SIP/2.0\r\nVia: SIP/2.0/UDP
XX.XXX.X.XX:5060;branch=z9hG4bK1f6e4ce4;rport\r\nFrom: \"unknown\"
<sip:unknown at XX.XXX.X.XX>;tag=as51db4c55\r\nTo:
<sip:2XX.XX.XXX.XXX>\r\nContact: <sip:unkn"..., len=491, fatal=1,
sipmethod=3) at chan_sip.c:2130
https://issues.asterisk.org/view.php?id=3 0x00129435 in send_request
(p=0xb67168d8, req=0x1d3eb0,
reliable=XMIT_CRITICAL, seqno=102) at chan_sip.c:2428
https://issues.asterisk.org/view.php?id=4 0x00140dc0 in transmit_invite
(p=0xb67168d8, sipmethod=3, sdp=0,
init=2) at chan_sip.c:7683
https://issues.asterisk.org/view.php?id=5 0x0017288e in sip_poke_peer
(peer=0xb46c2f98) at chan_sip.c:17144
https://issues.asterisk.org/view.php?id=6 0x001460ff in sip_poke_peer_s
(data=0xb46c2f98) at chan_sip.c:8476
https://issues.asterisk.org/view.php?id=7 0x0810a9b7 in ast_sched_runq
(con=0x96b8a58) at sched.c:363
https://issues.asterisk.org/view.php?id=8 0x001714bb in do_monitor (data=0x0) at
chan_sip.c:16988
https://issues.asterisk.org/view.php?id=9 0x0811bc2d in dummy_start
(data=0x96ba168) at utils.c:856
https://issues.asterisk.org/view.php?id=10 0x004265cc in start_thread () from
/lib/tls/libpthread.so.0
https://issues.asterisk.org/view.php?id=11 0x0037ef0e in clone () from
/lib/tls/libc.so.6 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2010-03-22 10:42 zerohalo       Note Added: 0119650                          
======================================================================




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