[asterisk-users] Subject: Re: ODBC locks warning in CLI - Asterisk
Stefan Viljoen
viljoens at verishare.co.za
Fri Nov 25 00:21:47 CST 2016
Hi Jonathan
Thx for the reply.
Yup, have tried them, may just be our incompetence and inexperience with
Asterisk, but cannot get either of them to work right with our particular
setup.
Due to legacy issues we run very different dialplans at 17 different sites,
and some in-house custom software for Asterisk, and from testing 13 and 14
it appears each and every one of the sites will need custom rebuilding and
redesigning to work right with the newer versions. We also use different
hardware (DAHDI wise) at each site, different, -very- old PRI cards
manufactured by different companies, etc.
Plus, been monitoring the group closely for about two years now, the
problems and bugs apparent with 13 and 14 (some of which were solved,
granted) are spine chilling - if we run into some of the issues I've seen
around, our business will collapse.
PJSIP especially appears to be an absolutely horrendous nightmare -
extremely complex and difficult to configure for the type of situations we
have where 1.8.32.3 has been doing fine for years, over several tens of
millions of calls.
But just my two cents, I could be completely wrong - if I can put the below
issue to bed definitively, the people I report to will probably stay on
standard 1.8.32.3 till it can no longer be compiled in a
whenever-contemporary Linux / libc / gcc environment...
>It might be worth pointing out that 1.8x was released 6 years ago, went
into security fix only over 2 years ago, and reached "end of life/no further
fixes" over a year ago.
>11.x went into "security fix only" last month - 13 and 14 are the current
versions - can you try with them?
On 23 November 2016 at 12:52, Stefan Viljoen <viljoens at verishare.co.za>
wrote:
> Hi all
>
> I get this warning in the Asterisk CLI about once every ten minutes or so:
>
> [Nov 23 14:47:36] WARNING[2544]: res_odbc.c:647
> ast_odbc_prepare_and_execute: SQL Execute returned an error -1: HY000:
> [MySQL][ODBC 5.1 Driver][mysqld-5.1.73]Deadlock found when trying to
> get lock; try restarting transaction (105) [Nov 23 14:47:36]
> WARNING[2544]: res_odbc.c:659
> ast_odbc_prepare_and_execute: SQL Execute error -1! Verifying
> connection to cdr [asterisk-cdr]...
> [Nov 23 14:47:36] WARNING[2544]: res_odbc.c:763 ast_odbc_sanity_check:
> Connection is down attempting to reconnect...
> [Nov 23 14:47:36] NOTICE[2544]: res_odbc.c:1541 odbc_obj_connect:
> Connecting cdr [Nov 23 14:47:36] NOTICE[2544]: res_odbc.c:1573
> odbc_obj_connect: res_odbc:
> Connected to cdr [asterisk-cdr]
>
> Does this imply that I'm missing the ODBC CELs and / or CDRs that were
> trying to write to MySQL over ODBC when the above occurred?
>
> Or will the ODBC module in Asterisk (or ODBC itself?) recover
> gracefully and re-emit the CEL or CDR insert that hit the lock and
> were therefore NOT written to MySQL?
>
> Thanks,
>
> Stefan
>
>
>
>
>
>
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
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>
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>
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Message: 3
Date: Wed, 23 Nov 2016 20:58:52 +0200
From: christopher kamutumwa <chriskamutumwa at gmail.com>
To: asterisk-users <asterisk-users at lists.digium.com>
Subject: [asterisk-users] Asterisk Installation
Message-ID:
<CADdH5aPyh4ZS_St-iONUtPd7_w4GQsfKWP=oLZM=KqkAT4oZyA at mail.gmail.com>
Content-Type: text/plain; charset="utf-8"
Goodday users
Am quite new to asterisk and trying to configure it with an fxo and fxs
digium card. also i need a gui interface implemented. I have a centos 6.8
server any tutorial i could use for install and configuration? would
appreciate.
Thanks
Chris
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Message: 4
Date: Wed, 23 Nov 2016 14:02:18 -0500
From: D'Arcy Cain <darcy at vex.net>
To: asterisk-users at lists.digium.com
Subject: Re: [asterisk-users] Touch tone stutter
Message-ID: <fa482650-add2-bf36-5479-915bff1ba19d at vex.net>
Content-Type: text/plain; charset=windows-1252; format=flowed
On 2016-11-22 07:49 PM, Pete Mundy wrote:
>
> One direction that may be worth exploring further is his ATA's config (or
perhaps swapping it for a different model). Eg adjusting echo cancellation
or line impedance settings.
I have to be careful here as I auto-provison these devices and changes would
propogate to every user. Echo cancellation is off. Do you think it should
be on?
> Is the ATA he is using the same as the ATA you use?
No but it is the same as other users who do not have the problem. I use a
SIP phone and a Cisco ATA.
> Failure to correctly recognise and decode DTMF is just one of many
> reasons why I never use them (ATAs). Like faxing over VoIP, they're
> just too much trouble :(
I understand but some use cases just need it.
> Genuine IP phones are pretty good value these days. Could you drop one of
those on-site as a temporary measure to prove that it's phone and/or ATA
related?
He does want to have an extension so that won't work.
> Ps, you might also want to consider joining VoiceOps (if you're not
> already subscribed) and posting there.
> https://puck.nether.net/mailman/listinfo/voiceops
I have subscribed. Thanks.
--
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:darcy at Vex.Net
VoIP: sip:darcy at Vex.Net
------------------------------
Message: 5
Date: Thu, 24 Nov 2016 00:40:18 +0530
From: Arun Kumar <arunvsadnikov at gmail.com>
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Subject: Re: [asterisk-users] Asterisk Installation
Message-ID:
<CANYuqXdQwoQtv6w_ihb-qt-GLs4_FpbYnejh_R7RN8B_pposyA at mail.gmail.com>
Content-Type: text/plain; charset="utf-8"
Hey Chris,
Starts from here,
https://wiki.asterisk.org/wiki/display/AST/Getting+Started or try Asterisk
Complete guide in pdf format. If you are looking for something graphical,
go for elastix or freepbx.
Thanks
~Arun
On Thu, Nov 24, 2016 at 12:28 AM, christopher kamutumwa <
chriskamutumwa at gmail.com> wrote:
> Goodday users
>
> Am quite new to asterisk and trying to configure it with an fxo and fxs
> digium card. also i need a gui interface implemented. I have a centos 6.8
> server any tutorial i could use for install and configuration? would
> appreciate.
>
> Thanks
>
> Chris
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> 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
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
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Message: 6
Date: Wed, 23 Nov 2016 17:41:19 -0500
From: Matt Riddell <lists at venturevoip.com>
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Subject: [asterisk-users] Subscribe to events via ARI from node.js
without sending to Stasis
Message-ID: <92B3CAC9-F2E0-4CAC-B0B1-59BDC62D2F0B at venturevoip.com>
Content-Type: text/plain; charset="us-ascii"
Hi,
I'm writing a node.js backend to pass events via a websocket to a CRM.
Basically what I want to do is notice when things happen (i.e. new channel,
new bridge etc) without sending the channels to the Stasis app.
The channels I'm interested in are agents who are in a queue only because
they are in a realtime MySQL database for the queue_member_table.
There doesn't appear to be a way to monitor general Asterisk events like you
can in the Asterisk manager without polling for channel statuses or sending
the channels to the Stasis app and recreating the logic of the Queue
application.
Is this a correct assumption?
--
Cheers,
Matt Riddell
_______________________________________________
http://www.venturevoip.com/news.php (Daily Asterisk News)
http://www.venturevoip.com/pabx_on_disk.php (PABX on a Disk)
http://www.venturevoip.com/exchange.php (Full ITSP Solution)
http://www.venturevoip.com/cc.php (Call Centre Solutions)
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Message: 7
Date: Thu, 24 Nov 2016 10:49:41 +0100
From: Juergen Sauer <juergen.sauer at automatix.de>
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Subject: [asterisk-users] unsbubscribe
Message-ID: <ed4b0b9a-49f9-cb6b-9f1b-99fc49615d88 at automatix.de>
Content-Type: text/plain; charset=utf-8
unsbubscribe
mit freundlichen Gr??en
J?rgen Sauer
--
J?rgen Sauer - automatiX GmbH,
+49-4209-4699, juergen.sauer at automatix.de
Gesch?ftsf?hrer: J?rgen Sauer,
Gerichtstand: Amtsgericht Walsrode ? HRB 120986
Ust-Id: DE191468481 ? St.Nr.: 36/211/08000
GPG Public Key zur Signaturpr?fung:
http://www.automatix.de/juergen_sauer_publickey.gpg
------------------------------
Message: 8
Date: Thu, 24 Nov 2016 17:20:54 +0000
From: A J Stiles <asterisk_list at earthshod.co.uk>
To: asterisk-users at lists.digium.com
Subject: [asterisk-users] Triggering an AGI script when a queued call
is answered
Message-ID: <201611241720.54479.asterisk_list at earthshod.co.uk>
Content-Type: Text/Plain; charset="us-ascii"
Many years ago, I used to have an AGI script that fired on an incoming call,
did some database lookups and ended up raising a notification on the screen
of
the person whose phone was ringing, with the details looked up from the
incoming caller ID.
All that fell by the wayside when Debian Squeeze introduced KDE4 and the
notification system I had created stopped working. And some time after
that,
we introduced queues instead of everyone having their own direct inbound
number .....
Now, some tie-wearer is dribbling on me to bring back the old system.
I am confident that I could write something that will work with the new
cross-
desktop notification model (and in any case, that is a matter for Elsewhere
On
The Internet). However, I am going to need to hook it into Asterisk
somehow.
What I think I need is for an event to fire when someone answers a queued
call;
then I can run an AGI script, or execute a script using the System()
command.
Within my script, I need the variable ${CALLERID(num)} to look up the
caller's
details from their number, and the answering extension to decide where to
send
the notification.
Is there a way of specifying in the dialplan or queue configuration that I
want
to execute a script when an agent answers?
So far, all I can think of is joining local channels into the queue instead
of
the actual phones, so I get to run a bit of dialplan where I can kick off
the
AGI script and then Dial() the actual extension; but that could get terribly
unwieldy if not done extremely carefully.
(Of course, the manager in question also insists for me to implement all
this
without a moment's downtime. Kids, this is what happens when your brain is
deprived of oxygen .....)
--
AJS
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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