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
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-7fe5]: frame.c:343 ast_frdup: FRACK!,
Failed assertion Excessive refcount 10 reached on ao2 object
In article
,
Joseph Smith 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 mom
On Friday 01 September 2017 at 16:48:17, Dovid Bender wrote:
> On Fri, Sep 1, 2017 at 9:13 AM, Joshua Colp wrote:
> > On Fri, Sep 1, 2017, at 09:01 AM, Dave Topping wrote:
> > > http:/www.theregister.co.uk/2017/09/01/asterisk_admin_patch/
> As Josh mentioned this is an issue with RTP and the SDP
On Fri, Sep 1, 2017 at 9:13 AM, Joshua Colp wrote:
> On Fri, Sep 1, 2017, at 09:01 AM, Dave Topping wrote:
> > http:/www.theregister.co.uk/2017/09/01/asterisk_admin_patch/
>
> This specific issue exists in a lot of different implementations and
> devices. Unfortunately there's nothing within SDP
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
On Fri, Sep 1, 2017, at 09:01 AM, Dave Topping wrote:
> http:/www.theregister.co.uk/2017/09/01/asterisk_admin_patch/
This specific issue exists in a lot of different implementations and
devices. Unfortunately there's nothing within SDP that guarantees or
provides what the source of media should be
http:/www.theregister.co.uk/2017/09/01/asterisk_admin_patch/
--
Dave Topping
e: i...@dntopping.uk
t: 03445 888 888
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Check out the new Asterisk community
Probably the best option is to create your own voicemail app using ARI.
Regards,
Marcelo H. Terres
IM: mhter...@jabber.mundoopensource.com.br
https://www.mundoopensource.com.br
https://twitter.com/mhterres
https://linkedin.com/in/marceloterres
On 1 September 2017 at 10:50, Tim Turpin wrote:
Thank you.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of J Montoya or A J
Stiles
Sent: Friday, September 01, 2017 3:47 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-us
On Friday 01 Sep 2017, Tim Turpin wrote:
> Is there a way that I can modify the source code for the voicemail
> application? I need to change some of the options in the user’s interface
> to make it work like an existing system that I’m replacing.
$ vi /usr/src/asterisk-*/apps/app_voicemail.c
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