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Summary:ASTERISK-12888: Asterisk 1.4.21.2 losing all ability to make calls
Reporter:Diego Lo Giduice (dlogiudice)Labels:
Date Opened:2008-10-14 10:34:04Date Closed:2009-02-02 14:27:19.000-0600
Priority:MinorRegression?No
Status:Closed/CompleteComponents:Core/Channels
Versions:Frequency of
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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

****** ADDITIONAL INFORMATION ******

It only happens with version 1.4.21.2; we have a few installations of previous version without a problem.
Comments:By: Leif Madsen (lmadsen) 2008-10-14 10:53:50

As that bug you mentioned asked... can you explain how to reproduce the issue? What are you doing to cause the system to think it should perform a shutdown? Do you run a restart script or something at night?
By: Diego Lo Giduice (dlogiudice) 2008-10-14 11:05:03

nothing it's being done to the system, it happens it the two different installations we have with asterisk 1.4.21.2, (we have 12 other installations...).
It happens mostly in the mornings when the people arrive to make calls.
Nothing it's beeing done actively to reproduce the error.
The only diference we see, is that started happening last week (no updates were done...) in one machine, and today in the other; maybe, we can send you the "full"  log of both machines from the last few days for you to compare....
By: Leif Madsen (lmadsen) 2008-10-14 12:59:59

Well the thing is that the only time I've seen that issue is when a "restart when convenient" is done. Any additional information you can attach to this issue could prove to be useful.
By: Diego Lo Giduice (dlogiudice) 2008-10-14 16:30:48

No manual operation it's being performed to make this happen...., it just happens by itself.
If you want we can provide SSH access to the boxes for you to check the logs/configs....
By: Leif Madsen (lmadsen) 2008-10-14 23:10:58

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.
By: Diego Lo Giduice (dlogiudice) 2008-10-15 09:01:02

Well, since we cannot use the phone (all of them are sip...), we don't know if it is related to sip only because we cannot access the zaptel trunks.
I would we glad to post all the logging info; Please could you tell me the lines/commands that I need to put in the sip.conf to enable logging??
thank you.
By: Leif Madsen (lmadsen) 2008-10-15 09:17:52

;--------------------------- SIP DEBUGGING ---------------------------------------------------
;sipdebug = yes                 ; Turn on SIP debugging by default, from
                               ; the moment the channel loads this configuration
;recordhistory=yes              ; Record SIP history by default
                               ; (see sip history / sip no history)
;dumphistory=yes                ; Dump SIP history at end of SIP dialogue
                               ; SIP history is output to the DEBUG logging channel
By: Diego Lo Giduice (dlogiudice) 2008-10-15 10:49:21

Ok, I will put the lines, but since this is a production pbx, I had to fix the problem putting a "amportal start"line in crontab every 10 minutis, to start the asterisk server, since this is only solution I came across with; So, I don't know if much is it goingt to be shown in future logs.
If you want I can post the previous log files from the pbx.
By: Leif Madsen (lmadsen) 2008-10-15 11:33:10

You can but not sure how much more it'll show than what you've already done.

If you can reproduce this on a test system, that would be the ideal. Otherwise, we may just need to close this bug as suspended until you can reproduce the issue on a test system so we can request debugging information.

Thanks!
By: Leif Madsen (lmadsen) 2009-02-02 14:27:19.000-0600

Closing this issue as we're unable to move it forward without the requested debugging information. Please feel free to have a bug marshal re-open the issue should you have the requested information. You can find a bug marshal in #asterisk-bugs on the Freenode IRC network at irc.freenode.net. Thanks!