[asterisk-bugs] [Asterisk 0013689]: Asterisk 1.4.21.2 losing all ability to make calls

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=13689 
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Reported By:                dlogiudice
Assigned To:                
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Project:                    Asterisk
Issue ID:                   13689
Category:                   Core/Channels
Reproducibility:            have not tried
Severity:                   minor
Priority:                   normal
Status:                     acknowledged
Asterisk Version:           1.4.21.2 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Disclaimer on File?:        N/A 
Request Review:              
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Date Submitted:             2008-10-14 10:34 CDT
Last Modified:              2008-10-14 23:10 CDT
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Summary:                    Asterisk 1.4.21.2 losing all ability to make calls
Description: 
Randomly, in the two installations of this version of asterisk, we are
having the same issue reported in ticket ID 0010917.

Randomly but mostly in the in the mornings, the pbxs refuses to make any
calls, internal or thru the trunks, and when you check the
/var/log/asterisk/full you get the message about an active shutdown in
progress:

[Oct 14 09:49:39] NOTICE[2683] chan_sip.c: Unable to create/find SIP
channel for this INVITE
[Oct 14 09:49:46] WARNING[2683] channel.c: Channel allocation failed:
Refusing due to active shutdown
[Oct 14 09:49:46] WARNING[2683] chan_sip.c: Unable to allocate AST channel
structure for SIP channel
[Oct 14 09:49:46] NOTICE[2683] chan_sip.c: Unable to create/find SIP
channel for this INVITE
[Oct 14 09:49:51] WARNING[2683] channel.c: Channel allocation failed:
Refusing due to active shutdown
[Oct 14 09:49:51] WARNING[2683] chan_sip.c: Unable to allocate AST channel
structure for SIP channel
[Oct 14 09:49:51] NOTICE[2683] chan_sip.c: Unable to create/find SIP
channel for this INVITE
[Oct 14 09:49:58] WARNING[2683] channel.c: Channel allocation failed:
Refusing due to active shutdown

this is the runing version of the software:
Asterisk Source Version  : 1.4.21.2
Zaptel Source Version    : 1.4.12.1
Addons Source Version    : 1.4.7
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---------------------------------------------------------------------- 
 (0093669) blitzrage (administrator) - 2008-10-14 23:10
 http://bugs.digium.com/view.php?id=13689#c93669 
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Does this only happen with SIP channels? If so, maybe it has something to
do with chan_sip... can you enable history and debugging in sip.conf, and
then the logging that is generated? Perhaps the SIP history can give us
something useful. 

Issue History 
Date Modified    Username       Field                    Change               
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2008-10-14 23:10 blitzrage      Note Added: 0093669                          
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