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

Thomas Peters TPeters at mcts.org
Tue Jan 15 08:53:29 CST 2019


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  
Desk: 414.343.1720 | Helpdesk: x3400 or  helpdesk at mcts.org
Milwaukee County Transit System 

1942 N 17th Street | Milwaukee, WI  53205
Check us out on Facebook & Twitter 

-----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


-- 
_____________________________________________________________________
-- 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


More information about the asterisk-users mailing list