Task processor queue reached 500 scheduled tasks again

Hi,

users cannot make calls (connection is not possible call back later)

asterisk console shows:

WARNING[29270][C-0000021a] taskprocessor.c: The ‘stasis/m:cache_pattern:1/channel:all-00000139’ task processor queue reached 500 scheduled tasks again.

WARNING[21411][C-0000014f] taskprocessor.c: The ‘stasis/m:channel:all-0000013a’ task processor queue reached 500 scheduled tasks again.

core show taskprocessors:
https://pastebin.freepbx.org/view/256aa76e

we have about 100 extensions and 4 trunks with about 100 channels in total

FreePBX 16.0.10.50
Asterisk Version 16.20.0
System Version 12.7.8-2201-3.sng7

please help to explain and resolve

This isn’t related to the number of extensions and trunks, but rather to the amount of dialplan and similar activity (calls that are just handling speech shouldn’t be loading the internal queues). Are you using custom dialplan? Do you have exceptionally high call set up rates? Is your machine virtual (including cloud) or real? Is your (virtual) machine underpowered? Is there anything that would delay database or file access?

The warning indicates that internal processing isn’t being cleared fast enough. The call failures are because the response to this is to try and reduce the load by refusing new calls, until the backlog is cleared.

im using default configuration of FreePBX (only set extensions trunk and routes via GUI)
FreePBX distro under Hyper-V VM (server 2022) on SSD M2, 8 GB RAM memory, 25% of I7-11700
htop shows 20-30% loaded
i have disabled CEL reporting but did not help

@Vengeful_Blade In all my experiences, it has been an issue with CPU and RAM.

First, I would remove unnecessary modules: for example isymphony if you do not use it (be sure to double check the daemons running, in the past I had to stop the service and prevent from starting on reboot with chkconfig off).

Are you doing anything like codec compression, or converting wav files to mp3 (call recordings or voicemail to email)? Any scripts running to input into the database with each call?

That said, it could be a problem with the version of asterisk (not as likely as the above). So if you have already done all the above, have you opened a post on the asterisk forum? They will most likely require a backtrace.
https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace

I have no any scrypts on system, just pure freepbx, only create extensions, incoming and oungoing routes and trunks

ERROR[24000][C-000007e8]: res_pjsip_header_funcs.c:622 func_read_header: This function requires a PJSIP channel.
[2022-01-27 11:49:09] WARNING[24006][C-000007e9]: func_strings.c:1442 function_eval: EVAL requires an argument: EVAL()
[2022-01-27 11:49:09] ERROR[16336]: res_pjsip_header_funcs.c:547 remove_header: No headers had been previously added to this session.

this flood can affect on this problem ?

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