Re: [asterisk-users] Various extensions ring once and go to voicemail

2019-01-15 Thread Stefan Viljoen 
Subject: Re: [asterisk-users] Various extensions ring once and goto
voicemail - Thomas Peters

>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.

Ouch. Sounds like you're maybe sitting with a hybrid package install setup and 
a partial source based install - did you setup this box yourself?

It appears that the binaries of asterisk were compiled, then the source was 
deleted or the binaries that comprise your instance were installed from a 
package...

You can probably build only timerfd, but it does imply running menuconfig (I 
think) and for that you need a properly configured Asterisk source tree, of the 
correct version you want.

>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?

AFAIK it is a default, but as default, again AFAIK, res_timing_dahdi.so won't 
get loaded, the pthread timer or timerfd will be used. Since you don't even 
have the timerfd module, you are running by implication on timerfd.

>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. 

Ok... well scratch that theory then.

As I said earlier, we have had very strange misbehaviour with Asterisk in 
virtually hosted environments, and after bitter experience resolved to run it 
only in real physical boxes as it seems to perform best there and be the most 
stable and reliable.

All I might suggest is getting the latest asterisk source in the 1.8 series 
(1.8.32.2 if I'm not mistaken - we ran it for years) and compile it from 
scratch. But do not install it, e. g. if you do make install it will overwrite 
your current setup irretrivably.

Rather, compile it, and then make actual physical copies of your current 
asterisk binary (/usr/sbin/asterisk, I think) and of your 
/usr/lib/asterisk/modules folder, -then- make install it. Start it up and see 
if it works better. If it is a success, great. If not, simply copy back your 
copied asterisk binary and copy back all the files in 
/usr/lib/asterisk/modules, and restart your old version which at least is 
working partially.

Again, no guarantees, the fact that you apparently already have a disjointed 
setup (at least three asterisk versions?) might mitigate against this - your 
milage may vary and doing what I describe might also destroy your entire 
current setup.

The reader must beware. It sounds as if you will need to recompile Asterisk 
from a known clean source to begin troubleshooting it anyway?

Kind regards,

Stefan


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

2019-01-15 Thread Eric Wieling 

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 |  tpet...@mcts.org
Desk: 414.343.1720 | Helpdesk: x3400 or  helpd...@mcts.org
Milwaukee County Transit System

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

-Original Message-
From: asterisk-users  On Behalf Of 
Stefan Viljoen
Sent: Tuesday, January 15, 2019 12:05 AM
To: asterisk-users@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.soDAHDI 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

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

2019-01-15 Thread Thomas Peters 
Actually, I was wrong about that. We no longer use OVM. It's actually Citrix 
Xencenter 7.6

Thomas M. Peters | Sr. Systems Administrator |  tpet...@mcts.org  
Desk: 414.343.1720 | Helpdesk: x3400 or  helpd...@mcts.org
Milwaukee County Transit System 

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

-Original Message-
From: asterisk-users  On Behalf Of 
Doug Lytle
Sent: Tuesday, January 15, 2019 9:08 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion 

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

>>> Carlos and Stefan (and other who have helped):

Thomas,

You stated that your virtual environment was Oracle, would that equate to 
VirtualBox?

Doug

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

2019-01-15 Thread Doug Lytle 
>>> Carlos and Stefan (and other who have helped):

Thomas,

You stated that your virtual environment was Oracle, would that equate to 
VirtualBox?

Doug

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

2019-01-15 Thread Thomas Peters 
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 |  tpet...@mcts.org  
Desk: 414.343.1720 | Helpdesk: x3400 or  helpd...@mcts.org
Milwaukee County Transit System 

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

-Original Message-
From: asterisk-users  On Behalf Of 
Stefan Viljoen
Sent: Tuesday, January 15, 2019 12:05 AM
To: asterisk-users@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.soDAHDI 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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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
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Check out the new Asterisk community forum at: https://community.asterisk.org/

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  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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Re: [asterisk-users] Various extensions ring once and go to voicemail - Thomas Peters

2019-01-14 Thread Stefan Viljoen 
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.soDAHDI 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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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

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Re: [asterisk-users] Various extensions ring once and go to voicemail

2019-01-14 Thread Carlos Chavez 

On 1/14/19 4:30 PM, Thomas Peters wrote:


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 M. Peters | Sr. Systems Administrator | tpet...@mcts.org 
<mailto:tpet...@mcts.org>
Desk: 414.343.1720 | Helpdesk: x3400 or helpd...@mcts.org 
<mailto:helpd...@mcts.org>


_Milwaukee County Transit System <http://www.ridemcts.com/>___

1942 N 17th Street | Milwaukee, WI  53205

Check us out on Facebook <https://www.facebook.com/mcts> & Twitter 
<https://twitter.com/RideMCTS>


*From:*asterisk-users  *On 
Behalf Of *Carlos Chavez

*Sent:* Monday, January 14, 2019 4:08 PM
*To:* asterisk-users@lists.digium.com
*Subject:* Re: [asterisk-users] Various extensions ring once and go to 
voicemail


On 1/14/19 4:02 PM, Duncan Turnbull wrote:

Sent from my iPad


On 15/01/2019, at 10:34 AM, Thomas Peters mailto:tpet...@mcts.org>> wrote:

Duncan:

You may have it right—I took one phone and set the ring time
to 60 seconds. I now get about 4 rings on that one.

I wonder how I can change the timing source.

In one version (and I can’t recall which) asterisk moved to an
internal timing system, to avoid the hardware need.

There should be quite a lot of discussion of it in the archives or
perhaps voipinfo

I don’t know if you can slow the VM processor speed? I am guessing
it is counting something much faster than it used to

Cheers Duncan

*CLI> module show like timing
Module Description  Use Count Status  
Support Level
res_timing_dahdi.so    DAHDI Timing 
Interface   0  Running core
res_timing_pthread.so  pthread Timing 
Interface 0  Running  extended
res_timing_timerfd.so  Timerfd Timing 
Interface 1  Running  core

3 modules loaded

    This will show you what module Asterisk is using for timing.  You 
can try doing a noload on the two you do not need.


--
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez
+52 (55)8116-9161


If your Asterisk server is a VM it should not be using DAHDI as a 
timing source.  The res_timing_timerfd.so module would probably be the best 
candidate.
Make sure your /etc/asterisk/modules.conf loads the timerfd module (and make 
sure it was built or installed by your package).

--
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez
+52 (55)8116-9161

-- 
_
-- 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

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Re: [asterisk-users] Various extensions ring once and go to voicemail

2019-01-14 Thread 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.soDAHDI Timing Interface   4
2 modules loaded

So what would you suggest? (And thanks in advance.)


Thomas M. Peters | Sr. Systems Administrator |  
tpet...@mcts.org<mailto:tpet...@mcts.org>
Desk: 414.343.1720 | Helpdesk: x3400 or  
helpd...@mcts.org<mailto:helpd...@mcts.org>
Milwaukee County Transit System <http://www.ridemcts.com/>

1942 N 17th Street | Milwaukee, WI  53205
Check us out on Facebook<https://www.facebook.com/mcts> & Twitter 
<https://twitter.com/RideMCTS>

From: asterisk-users  On Behalf Of 
Carlos Chavez
Sent: Monday, January 14, 2019 4:08 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Various extensions ring once and go to voicemail


On 1/14/19 4:02 PM, Duncan Turnbull wrote:

Sent from my iPad

On 15/01/2019, at 10:34 AM, Thomas Peters 
mailto:tpet...@mcts.org>> wrote:
Duncan:

You may have it right—I took one phone and set the ring time to 60 seconds. I 
now get about 4 rings on that one.

I wonder how I can change the timing source.

In one version (and I can’t recall which) asterisk moved to an internal timing 
system, to avoid the hardware need.

There should be quite a lot of discussion of it in the archives or perhaps 
voipinfo

I don’t know if you can slow the VM processor speed? I am guessing it is 
counting something much faster than it used to

Cheers Duncan



*CLI> module show like timing
Module Description  Use 
Count  Status  Support Level
res_timing_dahdi.soDAHDI Timing Interface   0   
   Running  core
res_timing_pthread.so  pthread Timing Interface 0   
   Running  extended
res_timing_timerfd.so  Timerfd Timing Interface 1   
   Running  core
3 modules loaded

This will show you what module Asterisk is using for timing.  You can try 
doing a noload on the two you do not need.



--

Telecomunicaciones Abiertas de México S.A. de C.V.

Carlos Chávez

+52 (55)8116-9161
-- 
_
-- 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
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Re: [asterisk-users] Various extensions ring once and go to voicemail

2019-01-14 Thread Carlos Chavez 

On 1/14/19 4:02 PM, Duncan Turnbull wrote:




Sent from my iPad

On 15/01/2019, at 10:34 AM, Thomas Peters > wrote:



Duncan:

You may have it right—I took one phone and set the ring time to 60 
seconds. I now get about 4 rings on that one.


I wonder how I can change the timing source.



In one version (and I can’t recall which) asterisk moved to an 
internal timing system, to avoid the hardware need.


There should be quite a lot of discussion of it in the archives or 
perhaps voipinfo


I don’t know if you can slow the VM processor speed? I am guessing it 
is counting something much faster than it used to


Cheers Duncan



*CLI> module show like timing
Module Description  Use Count  Status 
Support Level
res_timing_dahdi.so    DAHDI Timing Interface   
0  Running  core
res_timing_pthread.so  pthread Timing Interface 
0  Running  extended
res_timing_timerfd.so  Timerfd Timing Interface 
1  Running  core

3 modules loaded

    This will show you what module Asterisk is using for timing. You 
can try doing a noload on the two you do not need.



--
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez
+52 (55)8116-9161

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

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  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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Re: [asterisk-users] Various extensions ring once and go to voicemail

2019-01-14 Thread Duncan Turnbull 


Sent from my iPad

> On 15/01/2019, at 10:34 AM, Thomas Peters  wrote:
> 
> Duncan:
>  
> You may have it right—I took one phone and set the ring time to 60 seconds. I 
> now get about 4 rings on that one.
>  
> I wonder how I can change the timing source.

In one version (and I can’t recall which) asterisk moved to an internal timing 
system, to avoid the hardware need.

There should be quite a lot of discussion of it in the archives or perhaps 
voipinfo

I don’t know if you can slow the VM processor speed? I am guessing it is 
counting something much faster than it used to

Cheers Duncan



>  
> Thomas M. Peters | Sr. Systems Administrator |  tpet...@mcts.org  
> Desk: 414.343.1720 | Helpdesk: x3400 or  helpd...@mcts.org
> Milwaukee County Transit System
>  
> 1942 N 17th Street | Milwaukee, WI  53205
> Check us out on Facebook & Twitter
>  
> From: asterisk-users  On Behalf Of 
> Duncan
> Sent: Monday, January 14, 2019 2:29 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion 
> 
> Subject: Re: [asterisk-users] Various extensions ring once and go to voicemail
>  
>  
> 
> On Tue, Jan 15, 2019 at 7:42 AM, Thomas Peters  wrote:
> 
> We have an old Asterisk 1.8.7.0 system desperately need to keep alive for 
> another 6 months or so. We had all kinds of hardware problems, so we 
> virtualized it.
>  
> Thats a while back, I think it tended to use zaptel or dahdi hardware as a 
> timer, you may need to find a timing source as perhaps the clock in the VM is 
> just going too fast
>  
> 
> 
> Now, random extensions ring once and go straight to voicemail.
>  
> I went to one of the affected extensions and changed the ring time from the 
> default (20) to 26. Still one ring. I changed it to 30. Now I get two rings. 
> Other extensions ring once or twice.  After some time has gone by since this 
> was first reported, all phones in my random sample ring only twice.
>  
> As I trace a call to that extension through the log, I notice it setting the 
> ring timer properly (I think) and then I see
> app_dial.c – SIP/1234- is ringing
> Then eventually
> app_dial.c: -- Nobody picked up in 3 ms
>  
> But according to the timestamps, the time from the one event to the other is 
> ten seconds!
>  
> Here’s another example- call starts:
> [2019-01-14 08:17:33] VERBOSE[13311] pbx.c: -- Executing 
> [3327@cc-long-distance:1] ExecIf("SIP/4704-1265", "0?Set(__RINGTIMER=0)") 
> in new stack
> . . .
> [2019-01-14 08:17:33] VERBOSE[13311] app_dial.c: -- SIP/3327-1266 is 
> ringing
> . . .
> [2019-01-14 08:17:41] VERBOSE[13311] app_dial.c: -- Nobody picked up in 
> 2 ms
> So again, the elapsed time is nowhere near 20 seconds.
>  
> Another: This time the ring time has been set to 30 seconds (and I still get 
> only 2 rings)
> [2019-01-14 08:41:54] VERBOSE[16008] pbx.c: -- Executing 
> [3327@cc-long-distance:1] ExecIf("SIP/4704-1304", 
> "1?Set(__RINGTIMER=30)") in new stack
> . . .
> [2019-01-14 08:41:54] VERBOSE[16008] pbx.c: -- Executing 
> [s@macro-exten-vm:5] Set("SIP/4704-1304", "RT=30") in new stack
> . . .
> [2019-01-14 08:41:54] VERBOSE[16008] pbx.c: -- Executing 
> [s@macro-dial-one:30] Set("SIP/4704-1304", "D_OPTIONS=trWw") in new stack
> . . .
> [2019-01-14 08:41:54] VERBOSE[16008] app_dial.c: -- 
> SIP/3327-1305 is ringing
> . . .
> [2019-01-14 08:42:05] VERBOSE[16008] app_dial.c: -- 
> Nobody picked up in 3 ms
>  
> So, after 9 seconds, it says “Nobody picked up after 3 ms”???
>  
> Is this some weirdness of Oracle VMs? Anybody have any advice for me?
>  
>  
> Additional information:
> FreePBX version 2.9.0.7
> PBX in a Flash Version 1.2 Daemon Status
> 
> * Asterisk  * ONLINE  * Dahdi * ONLINE  * MySQL  * ONLINE  *
> * SSH   * ONLINE  * Apache* ONLINE  * Iptables   * OFFLINE *
> * Fail2ban  * OFFLINE * IP Connect* ONLINE  * Ip6tables  * OFFLINE *
> * BlueTooth * ONLINE  * Hidd  * ONLINE  * NTPD   * ONLINE  *
> * Sendmail  * ONLINE  * Samba * OFFLINE * Webmin * LOADING *
> * Ethernet0 * ONLINE  * Ethernet1 * ONLINE  * Wlan0  *   N/A   *
> 
> * Running Asterisk Version : Asterisk 1.8.7.0
> * Asterisk Source Version  : 1.8.7.0
> * Dahdi Source Version : 2.5.0.1+2.5.0.1
> * Libpri Source Version: 1.4.1

Re: [asterisk-users] Various extensions ring once and go to voicemail

2019-01-14 Thread Thomas Peters 
Duncan:

You may have it right-I took one phone and set the ring time to 60 seconds. I 
now get about 4 rings on that one.

I wonder how I can change the timing source.

Thomas M. Peters | Sr. Systems Administrator |  
tpet...@mcts.org<mailto:tpet...@mcts.org>
Desk: 414.343.1720 | Helpdesk: x3400 or  
helpd...@mcts.org<mailto:helpd...@mcts.org>
Milwaukee County Transit System <http://www.ridemcts.com/>

1942 N 17th Street | Milwaukee, WI  53205
Check us out on Facebook<https://www.facebook.com/mcts> & Twitter 
<https://twitter.com/RideMCTS>

From: asterisk-users  On Behalf Of 
Duncan
Sent: Monday, January 14, 2019 2:29 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion 

Subject: Re: [asterisk-users] Various extensions ring once and go to voicemail



On Tue, Jan 15, 2019 at 7:42 AM, Thomas Peters 
mailto:tpet...@mcts.org>> wrote:

We have an old Asterisk 1.8.7.0 system desperately need to keep alive for 
another 6 months or so. We had all kinds of hardware problems, so we 
virtualized it.

Thats a while back, I think it tended to use zaptel or dahdi hardware as a 
timer, you may need to find a timing source as perhaps the clock in the VM is 
just going too fast



Now, random extensions ring once and go straight to voicemail.

I went to one of the affected extensions and changed the ring time from the 
default (20) to 26. Still one ring. I changed it to 30. Now I get two rings. 
Other extensions ring once or twice.  After some time has gone by since this 
was first reported, all phones in my random sample ring only twice.

As I trace a call to that extension through the log, I notice it setting the 
ring timer properly (I think) and then I see
app_dial.c - SIP/1234- is ringing
Then eventually
app_dial.c: -- Nobody picked up in 3 ms

But according to the timestamps, the time from the one event to the other is 
ten seconds!

Here's another example- call starts:
[2019-01-14 08:17:33] VERBOSE[13311] pbx.c: -- Executing 
[3327@cc-long-distance:1] ExecIf("SIP/4704-1265", "0?Set(__RINGTIMER=0)") 
in new stack
. . .
[2019-01-14 08:17:33] VERBOSE[13311] app_dial.c: -- SIP/3327-1266 is 
ringing
. . .
[2019-01-14 08:17:41] VERBOSE[13311] app_dial.c: -- Nobody picked up in 
2 ms
So again, the elapsed time is nowhere near 20 seconds.

Another: This time the ring time has been set to 30 seconds (and I still get 
only 2 rings)
[2019-01-14 08:41:54] VERBOSE[16008] pbx.c: -- Executing 
[3327@cc-long-distance:1] ExecIf("SIP/4704-1304", "1?Set(__RINGTIMER=30)") 
in new stack
. . .
[2019-01-14 08:41:54] VERBOSE[16008] pbx.c: -- Executing 
[s@macro-exten-vm:5] Set("SIP/4704-1304", "RT=30") in new stack
. . .
[2019-01-14 08:41:54] VERBOSE[16008] pbx.c: -- Executing 
[s@macro-dial-one:30] Set("SIP/4704-1304", "D_OPTIONS=trWw") in new stack
. . .
[2019-01-14 08:41:54] VERBOSE[16008] app_dial.c: -- 
SIP/3327-1305 is ringing
. . .
[2019-01-14 08:42:05] VERBOSE[16008] app_dial.c: -- Nobody 
picked up in 3 ms

So, after 9 seconds, it says "Nobody picked up after 3 ms"???

Is this some weirdness of Oracle VMs? Anybody have any advice for me?


Additional information:
FreePBX version 2.9.0.7
PBX in a Flash Version 1.2 Daemon Status

* Asterisk  * ONLINE  * Dahdi * ONLINE  * MySQL  * ONLINE  *
* SSH   * ONLINE  * Apache* ONLINE  * Iptables   * OFFLINE *
* Fail2ban  * OFFLINE * IP Connect* ONLINE  * Ip6tables  * OFFLINE *
* BlueTooth * ONLINE  * Hidd  * ONLINE  * NTPD   * ONLINE  *
* Sendmail  * ONLINE  * Samba * OFFLINE * Webmin * LOADING *
* Ethernet0 * ONLINE  * Ethernet1 * ONLINE  * Wlan0  *   N/A   *

* Running Asterisk Version : Asterisk 1.8.7.0
* Asterisk Source Version  : 1.8.7.0
* Dahdi Source Version : 2.5.0.1+2.5.0.1
* Libpri Source Version: 1.4.12
* Addons Source Version: 1.4.7

Voipserver on 10.10.141.251 - eth0
Red Hat Enterprise Linux Server release 4.5 (Tikanga) :32 Bit Kernel: 
2.6.18-92.1.6.el5



Thomas M. Peters | Sr. Systems Administrator |  
tpet...@mcts.org<mailto:tpet...@mcts.org>
Desk: 414.343.1720 | Helpdesk: x3400 or  
helpd...@mcts.org<mailto:helpd...@mcts.org>
Milwaukee County Transit System <http://www.ridemcts.com/>

1942 N 17th Street | Milwaukee, WI  53205
Check us out on Facebook<https://www.facebook.com/mcts> & Twitter 
<https://twitter.com/RideMCTS>

-- 
_
-- Bandwidth a

Re: [asterisk-users] Various extensions ring once and go to voicemail

2019-01-14 Thread Duncan 



On Tue, Jan 15, 2019 at 7:42 AM, Thomas Peters  wrote:
We have an old Asterisk 1.8.7.0 system desperately need to keep alive 
for another 6 months or so. We had all kinds of hardware problems, so 
we virtualized it.


Thats a while back, I think it tended to use zaptel or dahdi hardware 
as a timer, you may need to find a timing source as perhaps the clock 
in the VM is just going too fast




Now, random extensions ring once and go straight to voicemail.

I went to one of the affected extensions and changed the ring time 
from the default (20) to 26. Still one ring. I changed it to 30. Now 
I get two rings. Other extensions ring once or twice.  After some 
time has gone by since this was first reported, all phones in my 
random sample ring only twice.


As I trace a call to that extension through the log, I notice it 
setting the ring timer properly (I think) and then I see

app_dial.c – SIP/1234- is ringing
Then eventually
app_dial.c: -- Nobody picked up in 3 ms

But according to the timestamps, the time from the one event to the 
other is ten seconds!


Here’s another example- call starts:
[2019-01-14 08:17:33] VERBOSE[13311] pbx.c: -- Executing 
[3327@cc-long-distance:1] ExecIf("SIP/4704-1265", 
"0?Set(__RINGTIMER=0)") in new stack

. . .
[2019-01-14 08:17:33] VERBOSE[13311] app_dial.c: -- 
SIP/3327-1266 is ringing

. . .
[2019-01-14 08:17:41] VERBOSE[13311] app_dial.c: -- Nobody picked 
up in 2 ms

So again, the elapsed time is nowhere near 20 seconds.

Another: This time the ring time has been set to 30 seconds (and I 
still get only 2 rings)
[2019-01-14 08:41:54] VERBOSE[16008] pbx.c: -- Executing 
[3327@cc-long-distance:1] ExecIf("SIP/4704-1304", 
"1?Set(__RINGTIMER=30)") in new stack

. . .
[2019-01-14 08:41:54] VERBOSE[16008] pbx.c: -- 
Executing [s@macro-exten-vm:5] Set("SIP/4704-1304", "RT=30") in 
new stack

. . .
[2019-01-14 08:41:54] VERBOSE[16008] pbx.c: -- 
Executing [s@macro-dial-one:30] Set("SIP/4704-1304", 
"D_OPTIONS=trWw") in new stack

. . .
[2019-01-14 08:41:54] VERBOSE[16008] app_dial.c: 
-- SIP/3327-1305 is ringing

. . .
[2019-01-14 08:42:05] VERBOSE[16008] app_dial.c: 
-- Nobody picked up in 3 ms


So, after 9 seconds, it says “Nobody picked up after 3 ms”???

Is this some weirdness of Oracle VMs? Anybody have any advice for me?


Additional information:
FreePBX version 2.9.0.7
PBX in a Flash Version 1.2 Daemon Status

* Asterisk  * ONLINE  * Dahdi * ONLINE  * MySQL  * ONLINE  *
* SSH   * ONLINE  * Apache* ONLINE  * Iptables   * OFFLINE *
* Fail2ban  * OFFLINE * IP Connect* ONLINE  * Ip6tables  * OFFLINE *
* BlueTooth * ONLINE  * Hidd  * ONLINE  * NTPD   * ONLINE  *
* Sendmail  * ONLINE  * Samba * OFFLINE * Webmin * LOADING *
* Ethernet0 * ONLINE  * Ethernet1 * ONLINE  * Wlan0  *   N/A   *

* Running Asterisk Version : Asterisk 1.8.7.0
* Asterisk Source Version  : 1.8.7.0
* Dahdi Source Version : 2.5.0.1+2.5.0.1
* Libpri Source Version: 1.4.12
* Addons Source Version: 1.4.7

Voipserver on 10.10.141.251 - eth0
Red Hat Enterprise Linux Server release 4.5 (Tikanga) :32 Bit Kernel: 
2.6.18-92.1.6.el5




Thomas M. Peters | Sr. Systems Administrator |  tpet...@mcts.org
Desk: 414.343.1720 | Helpdesk: x3400 or  helpd...@mcts.org
Milwaukee County Transit System

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

-- 
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