[asterisk-users] Issue with Asterisk 13, multiple CDR per queue and arbitrary upper limit

[Michaël Gaudette]

Hi,



I`ve recently upgraded a server from 1.8 to Asterisk 13.  While everything
is under control, I have one issue with the way CDRs are kept for queues.
And I don`t mean “I don`t like it”. I mean it crashes the server.



I realize there are multiple CDRs per queue call – one per ring/per phone,
basically.  The issue is that whenever the number of CDRs “to  be
recorded” for a call exceeds 5000, Asterisk becomes unresponsive for a few
minute. I get this message in the console:

“taskprocessor_push: The 'subm:cdr_engine-0003' task processor queue
reached 5000 scheduled tasks again.”



This scenario is trivial to reproduce: a queue, with simultaneous ring, 20
phones, all unreachable, 1 second between attempts.  After 250 (5000
divided by 20) seconds of waiting asterisk partially breaks down.



This seems to be because while multiple CDR`s are written per queue call,
it`s only done at the  end of the call, so CDRs accumulate in
memory/cacher/whatever and break some limit.



So, my question is:  is there any way to force the CDR`s to be written as
the queue app is working it`s magic, instead of at the very end of the
call? Or anyway to work around this limit? Or any fix for this?







Mike



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[asterisk-users] Asterisk crash when playing a WAV file to G722 SIP

[Richard Kenner]

I recently upgraded to Asterisk 14.3.0.  When playing a SIP file to a
G722 SIP channel (via chan_sip), I get a crash with the following
traceback.  This is reproducable:

#0  0x0036fdc30265 in raise () from /lib64/libc.so.6
#1  0x0036fdc31d10 in abort () from /lib64/libc.so.6
#2  0x0036fdc69beb in __libc_message () from /lib64/libc.so.6
#3  0x0036fdc7174f in _int_free () from /lib64/libc.so.6
#4  0x0036fdc75a4b in free () from /lib64/libc.so.6
#5  0x0050e19e in ast_frame_free (frame=0x5c35, cache=1) at frame.c:171
#6  0x00502bac in ast_readaudio_callback (s=0x6a5df88) at file.c:921
#7  0x00502d19 in ast_fsread_audio (data=0x5c35) at file.c:952
#8  0x004bb3df in __ast_read (chan=0x7ba68f8, dropaudio=0)
at channel.c:3848
#9  0x00504e51 in waitstream_core (c=0x7ba68f8, 
breakon=0x2b9630672bdb "", forward=0x5e56f8 "", reverse=0x5e56f8 "", 
skip_ms=0, audiofd=-1, cmdfd=-1, context=0x0, cb=0) at file.c:1602
#10 0x005053bf in ast_waitstream (c=0x5c35, 
breakon=0x5fca ) at file.c:1754
#11 0x2b963067272e in playback_exec (chan=0x7ba68f8, 

Does this "ring a bell" to anyone?  It looks like frame chainin has
gotten corrupted somehow, but this should be a straightforward case.


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Re: [asterisk-users] 100% CPU after upgrade.

[Matt Fredrickson]

One thing you didn't mention was what version you previously upgraded
from...  Also, more information about the system in general would
help.  (Endpoints, is it realtime or flat file configured, if
realtime, what type of database, what channel drivers (SIP or PJSIP,
and others).

Matthew Fredrickson

On Fri, Mar 31, 2017 at 12:08 PM, Mike Diehl  wrote:
> Hi all,
>
> I've upgraded to Asterisk 13.14.0 and now I'm seeing that Asterisk is using 
> 100% CPU.
>
> I have one AMI agent connected that is acting rationally.  I've got a hand 
> full of SIP (RT) registrations.  There is no other call activity.
>
> I've tried to unload various modules; nothing resolved the issue.
>
> Any suggestions?
>
> --
> Mike Diehl
>
>
>
>
> --
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>
> Check out the new Asterisk community forum at: https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
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>http://lists.digium.com/mailman/listinfo/asterisk-users



-- 
Matthew Fredrickson
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA

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[asterisk-users] 100% CPU after upgrade.

[Mike Diehl]

Hi all,

I've upgraded to Asterisk 13.14.0 and now I'm seeing that Asterisk is using 
100% CPU.  

I have one AMI agent connected that is acting rationally.  I've got a hand full 
of SIP (RT) registrations.  There is no other call activity.

I've tried to unload various modules; nothing resolved the issue.

Any suggestions?

-- 
Mike Diehl




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Re: [asterisk-users] Alphabet character in destination number (CDR)

[J Montoya or A J Stiles]

On Thursday 30 Mar 2017, Ikka Tirtawidjaja wrote:
> Dear all,
> 
> I have PBX with asterisk 13.x
> 
> a couple of IPPhone that connect to that asterisk PBX send an alphanumeric
> dialed phone number.
> 
> for example, in my CDR table, field DST, it show dialed phone number like
> - 0C81318304632C  (it should be 081318304632)
> - 08D11157112 (it should be 0811157112).
> 
> Why it's happening ? and how can I prevent it to happen ?

A, B, C and D are actually valid DTMF digits  (they belong in a column to the 
right of 3, 6, 9 and # respectively, and have the "high" frequency 1633 Hz).  
TTBOMK they were never actually used for anything in practice, they just keep 
kicking around like a vestigial organ  (compare how computer software for the 
UK financial industry still includes code to deal with mediaeval pounds, 
shillings and pence).  

Is it possible for a 1633 Hz tone, loud enough to swamp the "high" frequency 
of a dialled digit, to be finding its way somehow into the microphone of the 
affected phone and confusing it when digits are dialled?


-- 
JM

Note:  Originating address only accepts e-mail from list!  If replying off-
list, change address to asterisk1list at earthshod dot co dot uk .

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