Re: [asterisk-users] ERROR during high volume MoH dialplan

2017-09-01 Thread Mike
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

Re: [asterisk-users] ERROR during high volume MoH dialplan

2017-09-01 Thread Joseph Smith
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

Re: [asterisk-users] ERROR during high volume MoH dialplan

2017-09-01 Thread Tony Mountifield
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. >

Re: [asterisk-users] Asterisk bugs make a right mess of RTP

2017-09-01 Thread Antony Stone
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

Re: [asterisk-users] Asterisk bugs make a right mess of RTP

2017-09-01 Thread Dovid Bender
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

Re: [asterisk-users] ERROR during high volume MoH dialplan

2017-09-01 Thread Joseph Smith
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

Re: [asterisk-users] Asterisk bugs make a right mess of RTP

2017-09-01 Thread Joshua Colp
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

[asterisk-users] Asterisk bugs make a right mess of RTP

2017-09-01 Thread Dave Topping
http:/www.theregister.co.uk/2017/09/01/asterisk_admin_patch/ -- Dave Topping e: i...@dntopping.uk t: 03445 888 888 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community

Re: [asterisk-users] Asterisk Voicemail changes

2017-09-01 Thread Marcelo Terres
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

Re: [asterisk-users] Asterisk Voicemail changes

2017-09-01 Thread Tim Turpin
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:

Re: [asterisk-users] Asterisk Voicemail changes

2017-09-01 Thread J Montoya or A J Stiles
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