[asterisk-users] ERROR during high volume MoH dialplan

Fri Sep 1 15:46:53 CDT 2017

I had that problem before - I believe "task processor queue reached 500
scheduled tasks" crashing means your CDR records (queue) are being written
as the call ends, and if you had many thousands of entries being written
to disk it crashes asterisk (each ring to one phone is an entry, so it
goes up fast - for example 10 busy phones, with a between-ring delay of 1
second means every second there are 10 entries being put in memory)

I was using a MySQL CDR, but I had left the "CSV" type of CDR on. I
removed/disabled the CSV CDR module, kept on the SQL CDR only and things
have been working fine ever since.

Mike

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Joseph Smith
Sent: September 1, 2017 16:41
To: asterisk-users at lists.digium.com
Subject: Re: [asterisk-users] ERROR during high volume MoH dialplan

Thanks for the suggestion Tony,

I installed each codec for MoH, core sounds, and extra sound packages.
Unfortunately the tests produce the same results. 

[Sep  1 20:36:45] ERROR[10081][C-00007fe5]: frame.c:343 ast_frdup: FRACK!,
Failed assertion Excessive refcount 100000 reached on ao2 object 0x20380b0
(

continuously for a while followed by a 

[Sep  1 20:36:46] WARNING[7761][C-0000770d]: taskprocessor.c:888
taskprocessor_push: The 'subp:PJSIP/sipp-00000020' task processor queue
reached 500 scheduled tasks.

Then this time Asterisk actually crashed. :(

  _____  

From: asterisk-users-bounces at lists.digium.com
<asterisk-users-bounces at lists.digium.com> on behalf of Tony Mountifield
<tony at softins.co.uk>
Sent: Friday, September 1, 2017 11:01 AM
To: asterisk-users at lists.digium.com
Subject: Re: [asterisk-users] ERROR during high volume MoH dialplan 

In article
<CY4PR2201MB14643C2177C953FA27AC9E2BA8920 at CY4PR2201MB1464.namprd22.prod.ou
tlook.com>,
Joseph Smith <warlock1999 at hotmail.com> wrote:
> 
> Thanks for the feedback.
> 
> I do agree with having multiple smaller servers.  When I was first
approached with this task I mentioned as much. 
> However, the current desire is to work with already existing hardware.
That is out of my hands at the moment unless it
> just can't be done.  I will explore Freeswitch a bit soon to compare it
as well.
> 
> 
> I am struggling to find what the bottle neck is in this scenario.  Does
anyone have any advice on what that could be or
> on steps to discover it?   Do you think that tasks are pooling up
because of transcoding?  If so would it help to change
> the codec that is being used?  I am not sure about the MoH but the audio
files I am using are gsm.

You will find it less taxing on the server if you have MoH files and
sounds files
available in all the possible native formats. Then Asterisk can use the
appropriate
one for the channel without transcoding.

On my systems I have MoH and sounds installed in wav, ulaw, alaw, gsm and
g729.

They will also sound better than transcoding from the gsm versions.

Cheers
Tony
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