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Summary:ASTERISK-23998: channel.c: Exceptionally long voice queue length queuing to Local....
Reporter:Kampolis Ioannis (ikampolis)Labels:
Date Opened:2014-07-07 15:02:40Date Closed:2014-09-10 09:08:43
Priority:MajorRegression?
Status:Closed/CompleteComponents:Applications/app_queue Channels/chan_local Channels/chan_sip/Transfers
Versions:11.10.0 Frequency of
Occurrence		Constant
Related
Issues:
Environment:CENTOS 5.10 both 32bit and 64bit uname -a : 2.6.18-371.9.1.el5Attachments:( 0) console
( 1) console_debug.tgz
( 2) core.show.locks
( 3) etc-asterisk.tgz
( 4) extensions_additional.conf
( 5) queues.conf
( 6) sip_additional.conf
( 7) stressdialer.tgz
Description:Two Queues:
Queue 703 has one member 3304.
Queue 702 has three members ()
Once 3304 forwards (not using the asterisk feature code) all his calls to 702 a warning is issued channel.c: Exceptionally long voice queue length queuing to Local making the system unresponsive.
This causes load more than 9 to a monster dual-cpu DELL R520 in a production environment with more than 20k calls per day. The issue is resolved once the call is terminated or with a restart of the pbx.
Comments:By: Matt Jordan (mjordan) 2014-07-08 07:49:35.899-0500

Thank you for taking the time to report this bug and helping to make Asterisk better. Unfortunately, we cannot work on this bug because your description did not include enough information. You may find it helpful to read the Asterisk Issue Guidelines http://www.asterisk.org/developers/bug-guidelines. We would be grateful if you would then provide a more complete description of the problem. At a minimum, we need:

1. the specific steps or actions you took that caused you to encounter the problem,
2. the behavior you expected, and
3. the behavior you actually encountered (in as much detail as possible).

This likely includes output from the console with debug level logging, a SIP trace (if this is SIP related), and configuration information such as dialplan (e.g. extensions.conf), your queues configuration (queues.conf) and channel configuration (e.g. sip.conf). Thanks!


By: Kampolis Ioannis (ikampolis) 2014-07-08 11:27:54.267-0500

I am attaching debug information.
The dialplan (extensions_additional.conf) is generated by a clean free-pbx installation.
The problem is caused when a call reaches queue 703.
The only member (3304@703) has a permanent forward to queue 702.
Queue 702 has three members (sip phones- three is a random number).

By: Kampolis Ioannis (ikampolis) 2014-07-08 11:30:25.213-0500

Hi, thanks for the prompt response, I updated with debug information. As mentioned before the dialplan was generated by a clean freepbx install. If this is not suitable please do tell me to try to narrow down the dialplan.
I also included a core show locks output and a part of the sip configuration (even though I think it is not sip related). This is a configuration from a VM- not the actual production pbx for obvious reasons.

Best regards,

Ioannis
By: Kampolis Ioannis (ikampolis) 2014-07-09 14:34:42.096-0500

Just to clarify the pbx behaviour is almost as expected.
The call arrives in queue 703, then is redirected to 702 but unfortunatelly then the error occurs and asterisk becomes unresponsive
By: Matt Jordan (mjordan) 2014-07-10 09:20:51.148-0500

Your log file is not a debug log file.

Please read the instructions on the wiki on how to properly generate a debug log file, and attach it to this issue once it is generated.

https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information
By: Kampolis Ioannis (ikampolis) 2014-07-10 10:21:42.884-0500

Console + debug info
By: Kampolis Ioannis (ikampolis) 2014-07-10 10:22:37.503-0500

Added the console debug
By: S Harvanek (sharvanek) 2014-07-15 14:16:24.201-0500

I can confirm the same behavior on 11.11.0 , FreePBX , CentOS-6 x86_64.
By: Rusty Newton (rnewton) 2014-07-24 16:50:08.724-0500

There is a lot going on. We need to be able to reproduce this to investigate. I'm unable to reproduce this without a lot of guesswork.

A few options to move forward:

1. Provide a complete freepbx backup that we can restore and then, with minimal effort reproduce the issue.

or

2. Provide step by step instructions on how to set up the scenario in a clean FreePBX install

3. Provide a set of Asterisk configuration files that we can drop in, set up a phone or two with and reproduce the issue. Include instructions of course..
By: Kampolis Ioannis (ikampolis) 2014-08-07 13:46:00.710-0500

Sorry for the late reply - I am on holidays.
I had a colleague - who will probably reply back if you need anything else for the next 7-10 days - prepare me a VM. We used Elastix 2.4 in this VM.

I attach a full .tgz of /etc/asterisk.

Queue 702 has three members: ext 3301,3302,3311.
Queue 703 has one member for this example 3304.
All extensions are sip extensions.
One the SIP phone 3304 I use unconditional call forward to queue 702
For the call forward I did not use the feature code (*72), I just enabled it on the phone.
Then I use a simple code (stressdialer.tgz) to start sending calls to 703 by creating call files. Depending on the CPU it takes different number of simultaneous calls for the problem to appear.
In this case (VM with 2 cores - 2/4 of a Core 2 Quad Q8400)  10 simultaneous queue calls were required for the problem to appear.




By: Rusty Newton (rnewton) 2014-08-25 15:58:24.542-0500

I'm unable to use your /etc/asterisk to drop in and reproduce the issue. There appears to be other files required for Elastix/FreePBX to load your configuration, likely databases and database configuration. The result is dialplan, queues and other items don't appear to be configured at all when I use your configuration files.

You'll need to provide a simplified set of Asterisk configuration files that *do not* require FreePBX or Elastix to run (the point, to simplify the configuration). Or else if you are unable to do that, then provide *specific* instructions on how to configure the scenario that reproduces the issue within the FreePBX GUI on a default, fresh install. You could write out the steps, or screenshot the configuration in your GUI.
By: Rusty Newton (rnewton) 2014-09-10 09:08:31.418-0500

Suspended due to lack of activity. Please request a bug marshal in #asterisk-bugs on the IRC network irc.freenode.net to reopen the issue should you have the additional information requested.  Further information can be found at http://www.asterisk.org/developers/bug-guidelines


By: Alexandre Keller (alexandrekeller) 2014-09-17 17:20:15.674-0500

Hi there.

Any news on that?

Thanks in advance.
By: Bobby Hakimi (bobbymc) 2015-10-14 14:23:33.905-0500

im having the same issue and asterisk hangs after