[asterisk-users] Various extensions ring once and go to voicemail - Thomas Peters

Tue Jan 15 09:29:45 CST 2019

 From https://wiki.asterisk.org/wiki/display/AST/Timing+Interfaces:

res_timing_dahdi uses timing mechanisms provided by DAHDI. This method 
of timing was previously the only means by which Asterisk could receive 
timing. It has the benefit of being efficient, and if a system is 
already going to use DAHDI hardware, then it makes good sense to use 
this timing source. If, however, there is no need for DAHDI other than 
as a timing source, this timing source may seem unattractive. For users 
who are upgrading from Asterisk 1.4 and are used to the ztdummy timing 
interface, res_timing_dahdi provides the interface to DAHDI via the 
dahdi kernel module.

res_timing_timerfd uses a timing mechanism provided directly by the 
Linux kernel. This timing interface is only available on Linux systems 
using a kernel version at least 2.6.25 and a glibc version at least 2.8. 
This interface has the benefit of being very efficient, but at the time 
this is being written, it is a relatively new feature on Linux, meaning 
that its availability is not widespread.

On 01/15/2019 09:53 AM, Thomas Peters wrote:
> Carlos and Stefan (and other who have helped):
> 
> I DON'T HAVE the res_timing_timerfd.so file. Can I build it? Recompiling Asterisk is unrealistic in my position but I wonder if I can build the one module. Here's what I do have:
> 
> apbx:~ $ locate *res_timing_timerfd*
> /usr/src/asterisk-1.8.23.1/res/.res_timing_timerfd.makeopts
> /usr/src/asterisk-1.8.23.1/res/.res_timing_timerfd.moduleinfo
> /usr/src/asterisk-1.8.23.1/res/res_timing_timerfd.c
> /usr/src/asterisk-1.8.7.0/res/.res_timing_timerfd.makeopts
> /usr/src/asterisk-1.8.7.0/res/.res_timing_timerfd.moduleinfo
> /usr/src/asterisk-1.8.7.0/res/res_timing_timerfd.c
> 
> Why I have 1.8.23 and 1.8.7 I don't know. Asterisk on this system is version 1.8.7.0.
> 
> NEXT QUESTION: There are NO timing modules listed in /etc/asterisk/modules.conf at all. The only ones that are explicitly loaded are format_wav format_pcm format_mp3 and res_musiconhold. And there are "preload" directives for pbx_config.so and chan_local.so.
> 
> Is res_timing_dahdi.so getting loaded somewhere else? Or is it a default of some kind?
> 
> SYSTEM TIME OF DAY CLOCK which someone asked about, seems accurate. I did
> watch -n1 date
> and watched the time tick up, perfectly synchronized to my mobile phone. It might be off by a second or so, I'd have a hard time knowing for sure. NTPD is running, but not working for some reason. I fixed it (ownership of ntp.conf wrong) so now ntpq -pn returns a server ID.
> 
> 
> 
> Thomas M. Peters | Sr. Systems Administrator |  tpeters at mcts.org
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> 
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> 
> -----Original Message-----
> From: asterisk-users <asterisk-users-bounces at lists.digium.com> On Behalf Of Stefan Viljoen
> Sent: Tuesday, January 15, 2019 12:05 AM
> To: asterisk-users at lists.digium.com
> Subject: Re: [asterisk-users] Various extensions ring once and go to voicemail - Thomas Peters
> 
> Here’s what I get:
> 
> apbx*CLI> module show like timing
> Module                         Description                              Use Count
> res_timing_pthread.so          pthread Timing Interface                 0
> res_timing_dahdi.so            DAHDI Timing Interface                   4
> 2 modules loaded
> 
> So what would you suggest? (And thanks in advance.)
> 
> Thomas
> 
> I've had some good experience with
> 
> res_timing_dahdi
> 
> both when we ourselves were still on 1.8 and now with us on Asterisk 13 as well.
> 
> To force usage of a certain timer, specify in your modules.conf, e. g. to force use of DAHDI timing only, I did the following in my modules.conf:
> 
> .
> .
> .
> load => res_timing_dahdi.so
> noload => res_timing_pthread.so
> noload => res_timing_timerfd.so
> 
> That said, we have had some weird issues trying to run Asterisk in virtual machines - all our instances (16 of them) are physical machines.
> 
> We did a deployment at Azure in a Centos 7 "stock Azure" VM awhile ago and it suddenly lost the capability to encode .gsm audio files. All .gsm files the virtualised Asterisk 13 instances produced were all corrupt and no player would want to play the .gsm files. Neither could SOX convert them to anything. So we had to switch over to .wav, and then use a mixmonitor hook and manually convert the .wav files back to .gsm in SOX after each recording was written by Asterisk in .wav format. There were no errors logged, Asterisk just mysteriously lost the capacity to encode .gsm files when running on the Azure VM instance we had.
> 
> So quite probably the virtual environment / hypervisor you're using is part of the issue and switching timing modules around won't solve anything...
> 
> Have you checked that the system time is sane, and that one second on a stop watch externally to the VM instance, equates to one second inside it?
> 
> Because the symptoms described could indicate that the clock in the VM is just running too fast - or that some timing implementation detail inside Asterisk itself is running too fast.
> 
> Regards
> 
> Stefan
> 
> 

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