[asterisk-bugs] [Asterisk 0011972]: tls transport often causes asterisk to lock

[noreply at bugs.digium.com]

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=11972 
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Reported By:                pj
Assigned To:                jamesgolovich
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Project:                    Asterisk
Issue ID:                   11972
Category:                   Channels/chan_sip/General
Reproducibility:            random
Severity:                   block
Priority:                   normal
Status:                     assigned
Asterisk Version:           SVN 
SVN Branch (only for SVN checkouts, not tarball releases):  trunk 
SVN Revision (number only!): 103313 
Disclaimer on File?:        N/A 
Request Review:              
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Date Submitted:             02-11-2008 11:04 CST
Last Modified:              03-05-2008 02:18 CST
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Summary:                    tls transport often causes asterisk to lock
Description: 
when asterisk is locked, no call or registration is processed,
this happened several times today and yesterday (tried several trunk
revisions)
currently haven't found thing, what causes the locking situation, 
it seems, that locking problem start appearing, after more peers start
using tls transport
my tls only peers are: asterisk trunk, asterisk beta2 and eyebeam
sofphone
+several udp clients.



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---------------------------------------------------------------------- 
 pj - 03-05-08 02:18  
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branch http://svn.digium.com/svn/asterisk/team/group/issue_11972
is not working good, I got two crashes after few minutes,
now I'm running SVN-trunk-r105677 + bug11972-8.diff.txt, fine for now.

some msg from console (branch issue_11972):

[Mar  5 09:10:43]     -- Called bill-gw/xxxx
[Mar  5 09:10:43] ERROR[21429]: chan_sip.c:18041 handle_request_do: We
could NOT get the channel lock for SIP/bill-gw-08260420!
[Mar  5 09:10:43] ERROR[21429]: chan_sip.c:18042 handle_request_do: SIP
transaction failed: 016b7d6d7fcf33d515fa5a2d7e57b308 at xxxxxxx
[Mar  5 09:10:53] WARNING[21429]: chan_sip.c:15209 handle_response: Remote
host can't match request CANCEL to call
'016b7d6d7fcf33d515fa5a2d7e57b308 at xxxxx'. Giving up.



[Mar  5 09:11:02]     -- Executing [xxxx at zamestnanci:3]
Dial("SIP/xxx-0825c010", "SIP/bill-gw/xxx") in new stack
[Mar  5 09:11:02]   == Using SIP RTP CoS mark 5
[Mar  5 09:11:02] ERROR[21429]: astobj2.c:115 INTERNAL_OBJ: bad magic
number 0x101 for 0x8248cc8
*CLI>
Disconnected from Asterisk server


(gdb) bt full
http://bugs.digium.com/view.php?id=0  0xb7c6c4c5 in _start () at rtld.c:788
        preloadlist = Cannot access memory at address 0xb7c86b89 

Issue History 
Date Modified   Username       Field                    Change               
====================================================================== 
03-05-08 02:18  pj             Note Added: 0083408                          
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