[asterisk-bugs] [Asterisk 0013136]: [patch] sip peer qualified failed, asterisk lock.

Asterisk Bug Tracker noreply at bugs.digium.com
Mon Feb 9 17:16:03 CST 2009


A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=13136 
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Reported By:                pabelanger
Assigned To:                
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Project:                    Asterisk
Issue ID:                   13136
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     confirmed
Target Version:             1.6.0.5
Asterisk Version:           1.6.0 
Regression:                 No 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
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Date Submitted:             2008-07-23 02:32 CDT
Last Modified:              2009-02-09 17:16 CST
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Summary:                    [patch] sip peer qualified failed, asterisk lock.
Description: 
We just had asterisk lock on us tonight.  Best we can guess is because we
lost our SIP PEER (via qualify).

See output from 'show core locks'.
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---------------------------------------------------------------------- 
 (0099768) notthematrix (reporter) - 2009-02-09 17:16
 http://bugs.digium.com/view.php?id=13136#c99768 
---------------------------------------------------------------------- 
Dont know if it is the same issue but I somethimes get this problem when I
get a lockup asterisk completly locks up all channels when I doe a sip show
channels 
all channals are filled and it can not make new callas anymore
We are using a ht-502 and ht-503 from grandstream and using it in TLS mode
with the SRTP branch.
I could reproduce this bug when I disconect power from my ht-503 and bring
it to my 
niebours reconecting it over there connection both via NAT with nat
enabled.
It also seems to happen when a provider changes ip on the fly. 
I suspect this tyo happen with an austrailian friend who is connected via
bigpond adsl while my asterisk is in europe.
It looks that if I disable session timers
by doing 
session-timers=refuse
session-expires=110
session-minse=90
session-refresher=uas
The problem is gone but I only tryed it ones with a ht-502 again doing the
running to nighbour trick.
I cant use the patch provided because of SRTP (srtp branch)
But If it is the same bug it looks like tls does not disconect porperly 

sip show peers

Power disconnect.
123456789/123456789    (Unspecified)    D   N      0        UNKNOWN

while this is a normal peer in rest when not yet loged in after restart

Normal situation
987654321              (Unspecified)    D   N      5061     UNKNOWN

strange thing is that it defaults to port 5061 while I setted an other
default port 443

If this is the same bug I hope this can help since it is a very anoing bug
and locks my asterisk too manny times. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-02-09 17:16 notthematrix   Note Added: 0099768                          
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