Re: [asterisk-users] 13.22.0 - HTTP session count exceeded 100 sessions - instance unusable

[Stefan Viljoen]

Hi Joshua

No back-off, but I am caching the last 5000 results and and first hitting the 
cache to see if a recent command already provided the information I'm seeking 
for a particular request.

I'll see if I can do some simulation and see if I'm effectively DDOSing the 
local HTTP interface.

I'll have to see if I maybe have a resource leak in my code that makes the HTTP 
request, which is set  to "Connection: close" already (and not "Connection: 
keep-alive" at HTTP request header level) hang around and keep the HTTP 
connection open to Asterisk. 

Does the HTTP implementation in Asterisk heed this "Connection:" HTTP header?

--
Stefan


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Re: [asterisk-users] 13.22.0 - HTTP session count exceeded 100 sessions - instance unusable

[Stefan Viljoen]

Hi Joshua

No back-off, but I am caching the last 5000 result and and first hitting the 
cache to see if a recent command already provided the information I'm seeking 
for a particular request.




From: Joshua C. Colp  
Sent: Wednesday, 01 July 2020 15:47
To: viljo...@verishare.co.za; Asterisk Users Mailing List - Non-Commercial 
Discussion 
Subject: Re: [asterisk-users] 13.22.0 - HTTP session count exceeded 100 
sessions - instance unusable

On Wed, Jul 1, 2020 at 10:32 AM Stefan Viljoen 
<mailto:viljo...@verishare.co.za> wrote:
Hi Joshua

HTTP is used on in our setup on

http://127.0.0.1/mxml?

to send commands to the server, such as

http://127.0.0.1/mxml?action=login=myuser=thesecret

to log in and then

http://127.0.0.1/mxml?ActionID=123=BlindTransfer=Channel=local=123=1

etc. to control transfers, for example.

ARI is not being used.

WebRTC is not being used.

I have in the meantime modified httpd.conf and halved the session_inactivity 
from 30 000 to 20 000 and session_keep_alive from 15 000 to 10 000, and 
increased the sessionlimit to 1000 (instead of 100) to see if shorter sessions 
expiring more frequently and just plain more available slots for sessions will 
help?

So far so good, done about 55 000 new calls that way the past 32 minutes.

Or am I barking up the wrong tree?

It's possible it will help - but that still doesn't explain what exactly 
happened. Does your code have a back off strategy if HTTP requests fail? If not 
and they have an immediate retry with loop, then that may have effectively 
DDoSed things. It would also be good to explain why 100 connections were in use 
- did old connections not get closed? Was your load sufficient to just meet 
that amount? 

-- 
Joshua C. Colp
Asterisk Technical Lead
Sangoma Technologies
Check us out at http://www.sangoma.com and http://www.asterisk.org


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Re: [asterisk-users] 13.22.0 - HTTP session count exceeded 100 sessions - instance unusable

[Stefan Viljoen]

Hi Joshua

HTTP is used on in our setup on

127.0.0.1/mxml?

to send commands to the server, such as

http://127.0.0.1/mxml?action=login=myuser=thesecret

to log in and then

http://127.0.0.1/mxml?ActionID=123=BlindTransfer=Channel=local=123=1

etc. to control transfers, for example.

ARI is not being used.

WebRTC is not being used.

I have in the meantime modified httpd.conf and halved the session_inactivity 
from 30 000 to 20 000 and session_keep_alive from 15 000 to 10 000, and 
increased the sessionlimit to 1000 (instead of 100) to see if shorter sessions 
expiring more frequently and just plain more available slots for sessions will 
help?

So far so good, done about 55 000 new calls that way the past 32 minutes.

Or am I barking up the wrong tree?

Thanks!

--
Stefan


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[asterisk-users] 13.22.0 - HTTP session count exceeded 100 sessions - instance unusable

[Stefan Viljoen]

Hi all

I'm running an Asterisk 13.22.0 instance on Centos 7 - I7-8700 12 core HT
with 16GB of RAM.

The server maintains a total active call count of approx 285 calls with 440
channels at any one time. The totals never go below 200 calls concurrently
active.

For the kernel, top reports load levels at around 5.0 to 6.0 constantly.

I'm having issues where at random intervals, the CLI is filled with
thousands of these lines

[Jul  1 14:32:24] WARNING[11530]: http.c:1942 httpd_helper_thread: HTTP
session count exceeded 100 sessions.

This appears to be some form of race condition, the Asterisk instance goes
completely unresponsive, all RTP stops and all calls get hung up the moment
these appear on the CLI. All registered SIP phones then deregister off the
server withing one to two minutes.

At the time this takes place, kernel load spikes to about 9.0 and then falls
off to under 1 while asterisk endlessly outputs the above warning in the
Asterisk CLI.

You can still log into the CLI but no command is executed, you also cannot
get the Asterisk to stop with HUP / kill -1 and I have to kill -9 the
Asterisk PID at the end of the day.

I'm also, just prior the above taking place, getting tens of thousands of
these warnings in the CLI:

[Jul  1 14:30:39] WARNING[16819][C-04bf]: taskprocessor.c:895
taskprocessor_push: The 'stasis-core-control' task processor queue reached
500 scheduled tasks again.

I'm using AMI at several hundred lines of traffic per second to control the
Asterisk instance, spawn new calls, get queue and agent statuses, etc. -
standard AMI stuff.

Where can I start to look for how to fix this "session count exceeded 100
sessions" error?

Sometimes, it doesn't come up for weeks (loads remain much the same) and
then on one day it will happen four to six times in quick succession, load
broadly the same.

Any ideas?

Thanks!

Stefan


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[asterisk-users] Asterisk 13.22.0 unstable on Azure Centos 7 & cannot encode .gsm files

[Stefan Viljoen]

Hi all,

 

I'm running a Centos 7 instance in Azure with Asterisk 13. The Centos VM has
24GB of RAM and identifies the CPU as Intel(R) Xeon(R) CPU E5-2660 0 @
2.20GHz.

 

This is a virtual copy of a physical Centos 7 machine which has 16GB of RAM
and the physical CPU identifies itself as the same in /proc/cpuinfo.

 

I'm running up to 150 to 200 callers without problems on the physical
machine, usually have about 320 to 370 channels open simultaneously.

 

The physical box encodes to .gsm for recordings.

 

The virtual Centos 7 box keeps crashing with the same dialplan, at around
100 callers  (200+- channels).

 

Symptoms are that these will appear in the CLI on the virtual Asterisk
instance:

 

May 11 11:54:32 asterisk: [May 11 11:54:32]
#033[1;31mWARNING#033[0m[8830][C-0d02]:
#033[1;37mtaskprocessor.c#033[0m:#033[1;37m895#033[0m
#033[1;37mtaskprocessor_push#033[0m: The
'subp:ast_channel_topic_all-064d' task processor queue reached 500
scheduled tasks again.

May 11 11:54:32 asterisk: [May 11 11:54:32]
#033[1;31mWARNING#033[0m[8830][C-0d02]:
#033[1;37mtaskprocessor.c#033[0m:#033[1;37m895#033[0m
#033[1;37mtaskprocessor_push#033[0m: The
'subp:ast_channel_topic_all-0683' task processor queue reached 500
scheduled tasks.

May 11 11:54:32 asterisk: [May 11 11:54:32]
#033[1;31mWARNING#033[0m[8830][C-0d02]:
#033[1;37mtaskprocessor.c#033[0m:#033[1;37m895#033[0m
#033[1;37mtaskprocessor_push#033[0m: The
'subp:ast_channel_topic_all-0673' task processor queue reached 500
scheduled tasks again.

May 11 11:54:32 asterisk: [May 11 11:54:32]
#033[1;31mWARNING#033[0m[8773][C-0cfc]:
#033[1;37mtaskprocessor.c#033[0m:#033[1;37m895#033[0m
#033[1;37mtaskprocessor_push#033[0m: The
'subp:ast_channel_topic_all-0517' task processor queue reached 500
scheduled tasks again.

May 11 11:54:32 asterisk: [May 11 11:54:32]
#033[1;31mWARNING#033[0m[8755][C-0cf9]:
#033[1;37mtaskprocessor.c#033[0m:#033[1;37m895#033[0m
#033[1;37mtaskprocessor_push#033[0m: The
'subp:ast_channel_topic_all-0645' task processor queue reached 500
scheduled tasks again.

 

Mixed with 100s of lines of these: 

May 11 11:48:31 asterisk: [May 11 11:48:31]
#033[1;31mWARNING#033[0m[115959][C-0937]:
#033[1;37mast_expr2.fl#033[0m:#033[1;37m470#033[0m
#033[1;37mast_yyerror#033[0m: ast_yyerror():  syntax error: syntax error,
unexpected '', expecting $end; Input:

 

Then there will be a few 100 of these:

 

[May 11 12:10:33] WARNING[63930]: pbx.c:4548 increase_call_count: Maximum
loadavg limit of 50.00 load exceeded by 'Local/3155@local-2d4e;2'
(currently 50.19)!

 

And then the virtual Asterisk instance will go into a state where no call
hangs up and no new calls can be made via the AMI, calls to the AMI just
being indefinitely suspended and not receiving a response.

 

Neither can the virtual asterisk be shut down using a "core stop now" or
"core stop gracefully", and it doesn't respond to kill -1, I have to kill -9
the PID.

 

The virtual equivalent Centos 7 Asterisk 13.22.0 machine with 8GB more RAM
than the physical cannot nearly handle even half of what the real-metal
equivalent machine handles, without any of the above CLI errors and
warnings.

 

Also, the virtual asterisk encodes corrupt .gsm files, none it produces are
readable or playable by other Asterisk instances and not by VLC.

 

Is there anything I can do about the crashes - except get off Azure? I've
bypassed the corrupt .gsms by encoding to .wav and hooking into the
recording hook that ther mixmonitor app offers.

 

-ANY- ideas?

 

Thanks,



 

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Re: [asterisk-users] Asterisk 13.22.0 under very high load conditions - freezes in H exten and blocks new calls

[Stefan Viljoen]

Hi Dovid

No NFS connections are in use. Directly mounted EXT4 SSD (emulated, this is in 
a virtual environment).

Yes, I'm using ODBC to connect to the MySQL instance that records my CDRs.

Thanks for the reply!

Stefan

---
From: Dovid Bender  
Sent: Tuesday, 21 April 2020 16:53
To: viljo...@verishare.co.za; Asterisk Users Mailing List - Non-Commercial 
Discussion 
Subject: Re: [asterisk-users] Asterisk 13.22.0 under very high load conditions 
- freezes in H exten and blocks new calls

Are you using NFS? Any ODBC connections?

On Tue, Apr 21, 2020 at 10:23 AM Stefan Viljoen 
<mailto:viljo...@verishare.co.za> wrote:
Hi all

I'm running an Asterisk on an Intel XEON E5-2660 virtual with Centos 7 -
32GB RAM. 
When I approach about 320 channels, I -sometimes- get thousands of these
messages suddenly streamed in the CLI / Asterisk log:

WARNING[60753][C-00022cb9] channel.c: Exceptionally long voice queue length
queuing to Local/xx@local-0002dbea;2
WARNING[71993][C-00022dcc] channel.c: Exceptionally long voice queue length
queuing to Local/xx@local-0002dd10;2
WARNING[72848][C-00022dd1] channel.c: Exceptionally long voice queue length
queuing to Local/@local-0002dd30;1

Top reports system load as 30.0, but the box is still very responsive over
SSH. Asterisk is shown as consuming 600% CPU when NOT in this state (but
close to it) and doesn't consume more as this state comes into effect.

Top also shows that HT CPU instances are idle 40% of their time.

What then happens when these messages appear is Asterisk stops accepting
originates on the AMI, and that no CDRs write through to the CDR DB anymore
- MySQL itself is idle and has no locks or running inserts indicated in
"show processlist" in MySQL CLI.

The Asterisk instance will then effectively freeze, never hanging up
terrminated calls, and never allowing new cals to start, keeping around 320
channels open, forever.

Thing is I can sometimes peak at over 350 channels, then come back down as
answers, hangups, etc. take place - no problem. But -sustaining- more than
320 channels over a two or three minutes seems to reliably cause the above
to happen.

Asterisk has to be kill -9'ed to get it restarted, it ignores kill -1 /
sighup / cli "core stop now"

CLI does work and is still repsonsive in this state, until I do a "core stop
now", which doesn't stop the instance even after 20 to 30 minutes.

Asterisk max load is set to 40 and max calls to 350.

Any idea what I can look at? Any tips for running specifically 13 under high
load?

Thanks

Stefan





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[asterisk-users] Asterisk 13.22.0 under very high load conditions - freezes in H exten and blocks new calls

[Stefan Viljoen]

Hi all

I'm running an Asterisk on an Intel XEON E5-2660 virtual with Centos 7 -
32GB RAM. 
When I approach about 320 channels, I -sometimes- get thousands of these
messages suddenly streamed in the CLI / Asterisk log:

WARNING[60753][C-00022cb9] channel.c: Exceptionally long voice queue length
queuing to Local/xx@local-0002dbea;2
WARNING[71993][C-00022dcc] channel.c: Exceptionally long voice queue length
queuing to Local/xx@local-0002dd10;2
WARNING[72848][C-00022dd1] channel.c: Exceptionally long voice queue length
queuing to Local/@local-0002dd30;1

Top reports system load as 30.0, but the box is still very responsive over
SSH. Asterisk is shown as consuming 600% CPU when NOT in this state (but
close to it) and doesn't consume more as this state comes into effect.

Top also shows that HT CPU instances are idle 40% of their time.

What then happens when these messages appear is Asterisk stops accepting
originates on the AMI, and that no CDRs write through to the CDR DB anymore
- MySQL itself is idle and has no locks or running inserts indicated in
"show processlist" in MySQL CLI.

The Asterisk instance will then effectively freeze, never hanging up
terrminated calls, and never allowing new cals to start, keeping around 320
channels open, forever.

Thing is I can sometimes peak at over 350 channels, then come back down as
answers, hangups, etc. take place - no problem. But -sustaining- more than
320 channels over a two or three minutes seems to reliably cause the above
to happen.

Asterisk has to be kill -9'ed to get it restarted, it ignores kill -1 /
sighup / cli "core stop now"

CLI does work and is still repsonsive in this state, until I do a "core stop
now", which doesn't stop the instance even after 20 to 30 minutes.

Asterisk max load is set to 40 and max calls to 350.

Any idea what I can look at? Any tips for running specifically 13 under high
load?

Thanks

Stefan





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[asterisk-users] What are "non critical" invites?

[Stefan Viljoen]

Hi All

I'm getting tens of thousands of these messages ever hour in the Asterisk
CLI for Asterisk 13.22.0:

[Apr 20 15:59:46] WARNING[45462]: chan_sip.c:4127 retrans_pkt: Timeout on
192420-502043860-301870737 on non-critical invite transaction.
[Apr 20 15:59:46] WARNING[45462]: chan_sip.c:4127 retrans_pkt: Timeout on
301794058-652332923-1834701069 on non-critical invite transaction.
[Apr 20 15:59:47] WARNING[45462]: chan_sip.c:4127 retrans_pkt: Timeout on
393288224-685421081-2030967079 on non-critical invite transaction.
[Apr 20 15:59:48] WARNING[45462]: chan_sip.c:4127 retrans_pkt: Timeout on
1724665510-1751083232-677093666 on non-critical invite transaction.
[Apr 20 15:59:49] WARNING[45462]: chan_sip.c:4127 retrans_pkt: Timeout on
1958221882-69806522-85947083 on non-critical invite transaction.
[Apr 20 15:59:49] WARNING[45462]: chan_sip.c:4127 retrans_pkt: Timeout on
1013186441-666073546-1030482332 on non-critical invite transaction.

If an invite is non-critical, why do it then?

What is the purpose of a "non-critical" invite.

My server mostly works, with occassional channels that get stuck and need to
be hung up manually, but otherwise is working fine.

I'm trying to identify what a non-critrical invite is, and if it is
non-critical, can it be turned off somehow?

Since it is explictily shown as "non-critical" - it means it can be
dispensed with, as it is non-critical.

Clearly, it IS critical, since it exists, but why the strange semantic of
calling it "non critical" - why do something that is non-critical and
therefore not necessary.

Thanks!

Stefan



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Re: [asterisk-users] Predictive call - agent talking to a customer, then suddenly talking to another customer

[Stefan Viljoen]

Hi, do you have NAT between Asterisk and agent phones? 

S pozdravem
Tomáš Holý

Hi Tomas

Thanks for replying.

Yes, the phones are in one location in a LAN and are then NATed to enable them 
to contact the Asterisk which is hosted in the cloud.

A typical sip.conf phone configuration on the remote server for the site is

[general]
session-timers=refuse
disallow=all
allow=g729:20
allow=ulaw
allow=alaw
fromuser=xxx
useragent=xxx
callcounter=yes
alwaysauthreject=yes
allowguest=no
jbnable=yes
jbforce=no
jbimpl=adaptive
jblog=no
jbmaxsize=200
jbresyncthreshold=1000
externaddr=xx.xx.xx.xx
localnet=xx.xx.xx.xx/255.255.255.0

[]
type=peer
user=
secret=
host=dynamic
disallow=all
allow=g729
allow=ulaw
allow=alaw
dtmfmode=rfc2833
context=
call-limit=1
limitonpeers=yes
callgroup=1
pickupgroup=1
deny=0.0.0.0/0.0.0.0
permit=xxx.xxx.xxx.xxx/255.255.255.255
nat=force_rport,comedia

Not sure if that helps much.

Thanks for the reply!

> 13. 2. 2020 v 19:06, Stefan Viljoen :
> 
> 
> Hi all
>  
> Asterisk 13 instance - I’ve got a situation in an agent queue that an agent 
> will be talking to one person, then suddenly the same agent will be talking 
> to another person who was talking to another agent.
>  
> The calls do not switch around between the two agents, the “losing” agent 
> will just suddenly have silence in his handset and the other agent will now 
> be talking to “his” customer.
>  
> The original customer is simply cut off instantly the moment this happens.
>  
> This happens randomly. I have yet to collect log output and capture the CLI, 
> etc. but anybody ever heard of this happening?
>  
> It is as if agent channels get randomly reassigned / lose their audio channel 
> from outside the Asterisk - one channel is disconneted that a customer was on 
> and certain agents are suddenly talking to someone else and lose the original 
> caller they were busy with.
>  
> Where can I even being to look?
>  
> Thx!
> 


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[asterisk-users] Predictive call - agent talking to a customer, then suddenly talking to another customer

[Stefan Viljoen]

Hi all

 

Asterisk 13 instance - I've got a situation in an agent queue that an agent
will be talking to one person, then suddenly the same agent will be talking
to another person who was talking to another agent.

 

The calls do not switch around between the two agents, the "losing" agent
will just suddenly have silence in his handset and the other agent will now
be talking to "his" customer.

 

The original customer is simply cut off instantly the moment this happens.

 

This happens randomly. I have yet to collect log output and capture the CLI,
etc. but anybody ever heard of this happening?

 

It is as if agent channels get randomly reassigned / lose their audio
channel from outside the Asterisk - one channel is disconneted that a
customer was on and certain agents are suddenly talking to someone else and
lose the original caller they were busy with.

 

Where can I even being to look?

 

Thx!



 

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Re: [asterisk-users] Asterisk 13 on Microsoft Azure Centos 7 instance cannot encode gsm via MixMonitor

[Stefan Viljoen]

Hi Patrick

Wow ok thanks. I was not aware of this.

As far as I can determine the Azure VM we're using was set up on the "stock" 
Centos 7 option Azure offers. So you're correct, it then won't be an official 
Centos 7 image.

The one we are running on our bare-metal hosts IS installed from an official 
Centos 7 ISO burned to DVD... then yum updated fully once the base install is 
done.

This changes things a bit. THANKS!

I'll try and investigate further. Still wonder what the "RogueWave" or 
"OpenLogic" repos then have different that -only- GSM encoding doesn't work in 
Ast 13.22.0.

Thanks for the help.

Kind regards,

-Original Message-
From: Patrick Laimbock  
Sent: Friday, 13 September 2019 13:37
To: viljo...@verishare.co.za; asterisk-users@lists.digium.com
Subject: RE: [asterisk-users] Asterisk 13 on Microsoft Azure Centos 7 instance 
cannot encode gsm via MixMonitor

Hi Stefan,

> Hi all
> 
> I maintain the above - it was set up by an external party with whom relations 
> have now been severed by my employer.
> 
> Quite early after the deployment it became evident that all .gsm audio files 
> produced on this virtual instance at Azure via MixMonitor are corrupt.
[snip]

Is the CentOS 7 installation/image the same across your bare-metal hosts and 
the one on azure? AFAIK there is still no official CentOS 7 image provided by 
the CentOS Project on the azure marketplace. Instead it's created by a third 
party [1]. So there may be differences that could cause issues. On your azure 
host, check the repo files in /etc/yum.repos.d/. If the mirrorlist/basurl 
points to openlogic or roguewave than it's a third-party image. IIRC Amazon and 
GCP have official CentOS 7 images provided by the CentOS Project. Maybe try one 
of those to see if the issue persists? Alternatively create your own CentOS 7 
VM from the official CentOS 7 repositories using kickstart and try that on 
azure.

Best, Patrick

[1] 
https://azuremarketplace.microsoft.com/en-us/marketplace/apps/RogueWave.CentOS76?tab=Overview


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[asterisk-users] Asterisk 13 on Microsoft Azure Centos 7 instance cannot encode gsm via MixMonitor

[Stefan Viljoen]

Hi all

I maintain the above - it was set up by an external party with whom relations 
have now been severed by my employer.

Quite early after the deployment it became evident that all .gsm audio files 
produced on this virtual instance at Azure via MixMonitor are corrupt. If you 
play back the file in an audio player the audio almost sounds reversed and is 
severely distorted to the extent of being completely unintelligble. You can 
still convert the file to .wav with SOX, and if you look at the waveform in 
something like audacity the amplitude all over is near maximum and visually 
there IS no waveform to see... e. g. apparently the .gsm file itself is valid, 
but the GSM audio data itself is effectively gibbrerish.

But anyway:

The Asterisk config files and dialplan are identical to 17 other sites where 
Asterisk 13 is used in the same way as at Azure.

All the other sites are physical bare-metal hardware, and they work fine on the 
exact same Centos 7 based setup.

No errors are emitted at any verbosity level (in the Asterisk CLI or the log 
files) when the Azure Centos 7 Linux instance is writing corrupt .gsm files.

The problem most definitely is the fact that Asterisk runs in a  Centos 7 
instance on a virtual machine in the Azure cloud. 

Move the exact same config with the same updated Centos 7 kernel to a physical 
box, and the corrupt .gsm file problem disappears.

What can possibly be the problem that deploying Asterisk 13 in an Azure 
instance / VM breaks .gsm encoding?

(I have managed to do a workaround by encoding conversations via MixMonitor to 
.wav, and then using the MixMonitor hook to convert the .wav back to .gsm with 
SOX - which works fine.)

Why can SOX on the same Azure instance happily encode .wav to .gsm, but 
Asterisk "live" via MixMonitor produced corrupt .gsm files, with no errors 
emitted during encoding?

How can one fix this?

Thanks

Stefan


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Re: [asterisk-users] Odd one-way audio problem (Mike Diehl)

[Stefan Viljoen]

Hi Mike

Oh yes, this:

---
Note that the call ID is much longer than in the column display, e. g. a visual 
call ID in the sip show channelstats display may be

31f867c50ce

but the full call ID is then

32d867a55cfb563b7f59da01de84d...@xxx.xxx.xxx.xxx:5060

The full ID can be obtained by typing the partial ID as displayed in the list 
in the Asterisk CLI and then hitting TAB on the keyboard.
---

will only work if you go

sip show channel 31f867c50ce

and THEN press TAB on the keyboard.

Apologies.

Stefan


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Re: [asterisk-users] Odd one-way audio problem (Mike Diehl

[Stefan Viljoen]

Hi Mike

No rtp.conf in /etc/asterisk?!

This might be part of your problem, AFAIK rtp.conf is part of any standard 
installation might it be that you have a corrupted Asterisk install and / 
or -other- missing conf files as well?

Might this not also be part of your problem?

I installed Asterisk 13 by downloading the 13.22.0 version from Digium as a 
source .tar.gz and compiling and installig it myself, by hand. Not sure if 
yours comes from a distro repo or something similar which has been modified or 
is not a standard install.

That might also be something to try, e. g. see if you can compile a "clean" 
instance, from canon community version source, from Digium itself directly, and 
see if that changes anything.

Note that the 1 and 2 ranges I show below was mandated by my voice 
service provider, yours may differ! For example, I run 17 instances of Asterisk 
in South Africa with Centracom (my voice service provider) and in South Africa 
I have to use 1 - 2.

However, we also run a branch in Colorado in the United States with Twilio LLC 
as voice service provider in the USA, and they -require- 1 - 65535 to be 
used in rtp.conf... or you get exactly what you describe, e. g. random calls 
have no audio in either of the two directions.

For reference, your /etc/asterisk/rtp.conf file has a set format, e. g. it has 
to look like this (to parse, and work, I assume):

/etc/asterisk/rtp.conf
---
;
; RTP Configuration
;
[general]
;
; RTP start and RTP end configure start and end addresses
;
; Defaults are rtpstart=5000 and rtpend=31000
;
rtpstart=1
rtpend=2
;
; Whether to enable or disable UDP checksums on RTP traffic
;
;rtpchecksums=no
;
; The amount of time a DTMF digit with no 'end' marker should be
; allowed to continue (in 'samples', 1/8000 of a second)
;
;dtmftimeout=3000
; rtcpinterval = 5000   ; Milliseconds between rtcp reports
;(min 500, max 6, default 5000)
---

As you can see, defaults (if no rtp.conf specified) is 5000 and 31000 which is 
probably what you have been using up to this point, since you state that there 
was no rtp.conf on your system.

Also, you might need to do a "core reload" to make any changes to rtp.conf 
effective. A sip reload only re-parses sip.conf and reloads the SIP 
substructure and channel drivers, AFAIK it a sip reload doesn't affect RTP and 
doesn't parse and apply rtp.conf.

As regards the problems, are you sure that (for example) the audio codecs used 
are fully specified on, for example, the phones on either side?

With Yealink T-21 phones for example (and older ATCOM 820's and some older 
Polycoms) it is possible to turn off some codecs, and turn others on - I once 
had this kind of intermittent voice issue but only on a certain extension, and 
it turned out that the particular Yealink's codec mix was set too narrow (it 
would only accept ulaw and alaw, not G729) and this was precluding that phone 
from receiving incoming calls from our service provider (which only offer G729) 
while in-office calls to and from it worked fine as the other phones in the 
office were allowing running ulaw and alaw, -and- G729 so that any phone could 
take an incoming call (in G729) and have audio. Not sure of the dynamics (as 
Asterisk AFAIK should handle this situation transparently, transcoding as 
needed AFAIK) but it was definitively fixed by allowing G729 on the particular 
Yealink phone's web interface.

The only remaining issue may be NATing, e. g. for a particular network path to 
/ from a particular client, the intervening firewalls and / or routers might 
not allow the required UDP ports to be opened to allow RTP flow. You can set 
what you like in rtp.conf, but all the intervening network segments you use to 
get to the phone (and your VOIP trunk provider / service provider) -also- need 
to allow all the necessary UDP ports to be opened so RTP can flow back and 
forth. This might again be why it sometimes works, e. g. sometimes RTP ports 
are negotiated on port numbers that are too low / too high for a particular 
network segment or network edge device like a router or hardware / software 
firewall, and then there's no audio. The RTP port number is too low or too high 
for the intervening routers / firewalls, even though Asterisk is perfectly fine 
with that RTP port number, the intervening network(s) is incapable of carrying 
the RTP flow as the port number is invalid according to a router / firewall 
configuration on a device between you and the phone.

If a customer is in a state of no audio, what does

sip show channelstats

in the Asterisk CLI, show?

E. g. the "sip show channelstats" display shows RTCP (RTP control protocol) 
statistics, e. g. how bad the jitter and packet loss for a particular call is 
on a certain extension for the incoming and outgoing legs of the call.

In a situation with no audio, can you check on this to see if (for example) 
-any- RTP packets flow at all? E. g. a particular readout 

Re: [asterisk-users] Odd one-way audio problem (Mike Diehl)

[Stefan Viljoen]

Hi Mike

In rtp.conf, what are the port ranges you specify?

I had almost exactly the same problem not too long ago. People will phone, and 
sometimes it will work, sometimes not - one way audio would happen, then start 
working, then stop working.

The problem turned out to be that the port specification for RTP traffic in 
/etc/asterisk/rtp.conf was too wide.

It was set to

rtpstart=1
rtpend=65535

(apparently by a previous maintainer / technician who worked on the system.)

The high port number was too high, and only after I investigated in detail with 
our trunk provider, were they able to determine that somtimes the Asterisk on 
my side was negotiating too high port numbers for RTP with their system.

I changed rtp.conf to read

rtpstart=1
rtpend=2

and all the random one-way audio problems have been gone for more than two 
months. This client now has had thousads of successful calls so far after this 
change was made.

I also had the issue where MOST calls in their office was fine (with rtp.conf 
at 1 to 65535) though some would still fail, I'm guessing that was due to 
NATing not being done in the office (e. g. a wider "range" of RTP ports worked) 
vs. when they connected to their provider's SIP trunk on the internet to 
negotiate calls where it was ignoring the higher ports ("too high" ports) or 
their local firewall wasn't allowing some high ports to be opened that were 
"too high".

Restricting the RTP port range between 1 and 2 in this case solved 
their problem definitively and forever.

E. g. something similar given that you start that "most of the time" things 
worked fine - which is exactly the symptom I had with this client.

Just a thought...

Regards

Stefan

---

Hi all,

I have a user who is reporting one-way audio, but only when a call is made to 
or from particular PSTN (cell) numbers.

Their phones are behind a NAT router and my server is on the open Internet.  

Calls within their office sound fine.  Calls to/from most numbers sound fine.

When they took their phones home, those same phone numbers still had problems.

So, I don't think it's their network.  I've taken pcaps of both legs of example 
calls.  On the provider-side, I see 2-way audio.  On the client-side, I only 
hear one side.

Most of the time, though, their phones work correctly.

Any ideas where to look to fix this?

Thanks in advance.




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Re: [asterisk-users] Asterisk users survey

[Stefan Viljoen]

Hi Joshua

Does the survey imply that there are big changes coming for Asterisk?

E. g. features or facilities will be dropped / deprecated from the open source 
version in  new releases, big changes to existing facilities / protocols, what 
is supported officialy by Digium for the official version, and what not, etc.

Given that you are now owned by Sangoma & all and there are different people in 
control now than before.

Can you confirm / deny?

Thanks,

> We are in the same situation as Jean Denis, running 1.4 to 16 version as 
> integrator/service provider/user/...
> 
> Difficult to replay the survey for each scenarios ;)

The most active deployment, or a few deployments, is fine.

-- 
Joshua C. Colp
Digium - A Sangoma Company | Senior Software Developer


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Re: [asterisk-users] Problem with the DB() function (Ira)

[Stefan Viljoen]

>So the new install is coming along. I hooked up the new box for a couple of 
>hours and got a bunch more problems worked out. And yet some still remain. I 
>have this subroutine I call occasionally:
>.
>.
>.
>Also, when I installed asterisk it did not set itself up to start when the 
>machine boots. Is there something else I need to do?  It's the most recent 
>Fedora.

If you're on the latest Fedora you're probably using systemd for init and 
startup / control of background functions.

On Centos7, to get Asterisk to start up with the system on boot I do the 
following as root:

# cd /etc/systemd/system

# vim asterisk.service

- In this file, then put

---

[Unit]
Description=Asterisk
After=network.target
After=network-online.target
After=startup.service
After=mysql.service
Wants=mysql.service

[Service]
Type=idle
User=root
Group=root
ExecStart=/usr/sbin/asterisk -f
ExecStop=/usr/sbin/asterisk -rx 'core stop now'
ExecReload=/usr/sbin/asterisk -rx 'core reload'
TimeoutSec=300

LimitCORE=infinity
LimitNOFILE=16384
Restart=always
RestartSec=5

[Install]
WantedBy=multi-user.target 

---

# systemctl enable asterisk.service

Now, on reboot, systemd should start up asterisk for you.

Hope this helps.

Stefan


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Re: [asterisk-users] Asterisk 1.8.7.0 connectivity to Avaya SM (Thomas Peters)

[Stefan Viljoen]

E. g. You can also then try in /etc/hosts to put

10.10.0.103 mcts.org

e. g. if any local reverse lookup is done "10.10.0.103" resolves to "mcts.org" 
and the converse for normal DNS lookups.

You can even try in /etc/asterisk/asterisk.conf to make the "systemname" line be

systemname=10.10.0.103

This is also why if you have 

systemname=mcts.org

in /etc/asterisk/asterisk.conf a DNS lookup on "mcts.org" on the linux box 
itself will give "10.10.0.103" or "127.0.0.1" depending on what is in 
/etc/hosts, e. g. to try and force your Asterisk instance if it needs / tries 
to ID itself when sending SIP messages it will identify itself as "10.10.0.103" 
in one part of the sip message, while you also set its domain name to be 
"mcts.org" hopefully forcing it to put that to the right of the "@" in 

10.10.0@mcts.org<mailto:10.10.0@mcts.org>

Also, does the Avaya device behave at all like a Linux / WIndows machine e. g. 
maybe it needs to do R-DNS on itself as well, e. g. if the Avaya people on 
their side can edit their version of the Linux "/etc/hosts" and put in there

10.10.0.103 mcts.org

so that if THEY do a R-DNS lookup on "mcts.org" (if your Asterisk co-operates 
and passes this in to them in SIP, see above & previous message) and they do 
reverse dns lookup on it they get "10.10.0.103" in

10.10.0@mcts.org<mailto:10.10.0@mcts.org>

All conjecture and guesswork though, but maybe something makes a light-bulb 
light up...

Not probably relevant, but I had once to do this with Asterisk 1.8.32.3 in much 
the same situation, e. g. we had a 2nd Linux box running ViciDial (bascially 
also Asterisk) that would sometimes dial through to "our" Asterisk server to 
get outgoing lines.

I put together this in "our" Asterisk server's sip.conf:

[9090] ;This is the SIP identity VICIDial uses to dial out when it wants to
type=peer
user=9090
host=10.1.51.174
disallow=all
allow=g729
allow=alaw
allow=ulaw
context=local
call-limit=20
limitonpeer=yes
canreinvite=yes
qualify=no
deny=0.0.0.0/0.0.0.0
permit=10.1.51.174/255.0.0.0

This caused "our" Asterisk to treat "their" Asterisk as just another SIP 
device, e. g. no difference at all between it and a normal VOIP phone on the 
network as far as "our" Asterisk was concerned. "Our" Asterisk could not 
distinguish if it was getting a call from a phone, or from this ViciDial server.

Hope some of this helps.

Regards

Stefan

-Original Message-
From: Stefan Viljoen  
Sent: Thursday, 28 February 2019 08:18
To: 'asterisk-users@lists.digium.com' 
Subject: Asterisk 1.8.7.0 connectivity to Avaya SM (Thomas Peters)
 
Hi Thomas

What is your IVR box's domain name in Linux?

With a hostname of, for example, "mcts.org" do you have a line like this in 
/etc/hosts:

127.0.0.1   mcts.org

in your /etc/hosts?

Additionally, in /etc/asterisk/asterisk.conf, is there a line

systemname =

that is -uncommented- and contains the string you need to the right of the @ e. 
g.

mcts.org

?

The immediate above just a shot in the dark but could affect how your IVR box 
identifies itself.

If the Linux system host name of the IVR machine is not "mcts.org" maybe you 
can change the machine's name via (for Centos 7 at least - if you have 
NetworManager running):

nmcli general hostname mcts.org
systemctl restart systemd-hostnamed

and then reboot the machine which implies also restarting asterisk.

If I remember right this will cause the Asterisk instance to identify itself as 

"mcts.org" sip:mcts@mcts.org:0;tag=...
^   ^
||
From asterisk.conf |
 From nmcli general hostname mcts.org

?

Might still not solve your precise issue but might be a step in the right 
direction??

Stefan
---
>We have a brand-new Avaya phone system in place and we will be cutting over to 
>it in late March 2019.

> Presently:

 > *   We have an Asterisk 13.3.2 server with no phones registered to it, 
 > acting as a PSTN gateway. Calls come into it and get distributed to other 
 > Asterisk boxes with phones.
 > *   If a call comes in from the provider marked as having been dialed as 
 > xxx-xxx-6711 (those are digits, not a pattern) it gets routed to the IVR box
 > *   The IVR box runs Asterisk 1.8.7.0 and a custom IVR.

>Where we have to get to:

 > *   The new Avaya Session Manager has to have a working SIP trunk to the IVR 
 > so it can pass calls that come into xxx-xxx-6711 to it.

>What the problem is:

>  *   I don't fully understand what's going on here, neither how it works now, 
> nor what I need to do to make Avaya's SM happy.
  



-- 
___

[asterisk-users] Asterisk 1.8.7.0 connectivity to Avaya SM (Thomas Peters)

[Stefan Viljoen]

Hi Thomas

What is your IVR box's domain name in Linux?

With a hostname of, for example, "mcts.org" do you have a line like this in 
/etc/hosts:

127.0.0.1   mcts.org

in your /etc/hosts?

Additionally, in /etc/asterisk/asterisk.conf, is there a line

systemname =

that is -uncommented- and contains the string you need to the right of the @ e. 
g.

mcts.org

?

The immediate above just a shot in the dark but could affect how your IVR box 
identifies itself.

If the Linux system host name of the IVR machine is not "mcts.org" maybe you 
can change the machine's name via (for Centos 7 at least - if you have 
NetworManager running):

nmcli general hostname mcts.org
systemctl restart systemd-hostnamed

and then reboot the machine which implies also restarting asterisk.

If I remember right this will cause the Asterisk instance to identify itself as 

"mcts.org" sip:mcts@mcts.org:0;tag=...
^   ^
||
From asterisk.conf |
 From nmcli general hostname mcts.org

?

Might still not solve your precise issue but might be a step in the right 
direction??

Stefan
---
>We have a brand-new Avaya phone system in place and we will be cutting over to 
>it in late March 2019.

> Presently:

 > *   We have an Asterisk 13.3.2 server with no phones registered to it, 
 > acting as a PSTN gateway. Calls come into it and get distributed to other 
 > Asterisk boxes with phones.
 > *   If a call comes in from the provider marked as having been dialed as 
 > xxx-xxx-6711 (those are digits, not a pattern) it gets routed to the IVR box
 > *   The IVR box runs Asterisk 1.8.7.0 and a custom IVR.

>Where we have to get to:

 > *   The new Avaya Session Manager has to have a working SIP trunk to the IVR 
 > so it can pass calls that come into xxx-xxx-6711 to it.

>What the problem is:

>  *   I don't fully understand what's going on here, neither how it works now, 
> nor what I need to do to make Avaya's SM happy.
  



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Re: [asterisk-users] trying to upgrade asterisk and Debian -- not working

[Stefan Viljoen]

Hi John

I've jumped around between several Asterisk versions on Centos 7 by doing

1. Copying the asterisk binary in 

/usr/sbin/asterisk

to

/usr/sbin/asterisk.1.8.11.0

(or whatever version number)

2. Tar'ing up the /usr/lib/asterisk/modules directory's contents

# cd /usr/lib/asterisk/modules

# tar cvf asterisk.1.8.11.0.tar *

3. Deleting all the old / different version's module object files, e. g. still 
in /usr/lib/asterisk/modules

rm -rf *.so

4. Untarring and ungzipping the new / different version I downloaded manually 
into my

/usr/src

folder, e. g.

cd /usr/src

gunzip asterisk-1.8.32.3.tar.gz

tar xvf asterisk-1.8.32.3.tar

5. Then the usual

# cd asterisk-1.8.32.3
# ./configure
# make menuconfig

then set options then

# make
# make install
# make samples
# make progdocs

Ok, so now I'm on 1.8.32.3 and if I go

# asterisk

I will start up the "new" 1.8.32.3 version.

To revert to Asterisk 1.8.11.0 (or whatever) I then stop asterisk, and then I 
simply rename the current 

/usr/sbin/asterisk to 

asterisk.18.32.3

and the 

/usr/sbin/asterisk.1.8.11.0

that is still there to

/usr/sbin/asterisk

thereby making that binary active again.

Then I tar up all the .so files in 

/usr/lib/asterisk/modules

into

asterisk.1.8.32.3.tar

and delete all .so files in /usr/lib/asterisk/modules, then extract

tar xvf asterisk.1.8.11.0.tar

to revert to the older version's modules.

I can then go

# asterisk

and now I'm back to running

Asterisk 1.8.11.0

I've done this with three or four different Asterisk versions in the past, 
switching between them for testing.

A bit more manual procedure that you've done, but it does work to try out 
similar Asterisk versions / subversions for testing?

Regards,

Message: 2
Date: Thu, 24 Jan 2019 02:49:52 -0500
From: John Covici 
To: viljo...@verishare.co.za, Asterisk Users Mailing List -
Non-Commercial Discussion 
Subject: Re: [asterisk-users] trying to upgrade asterisk and Debian --
not working (John Covici)
Message-ID: 
Content-Type: text/plain; charset=US-ASCII

I checked out 13.15.0, ./configure, make delete all modules, followed by make 
install.

On Thu, 24 Jan 2019 01:17:32 -0500,
Stefan Viljoen wrote:
> 
> What procedure did you follow to revert back to the old version?


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Re: [asterisk-users] trying to upgrade asterisk and Debian -- not working (John Covici)

[Stefan Viljoen]

What procedure did you follow to revert back to the old version?

It sounds like your binary has been revereted, but the modules it needs to load 
are still the 13.24.0-rc1 modules...

---
Hi.  I am trying to upgrade my asterisk from 13.15 to the latest of asterisk 13 
which seems to be 13.24.0-rc1.  At the same time I want to go from Debian 8 to 
DEbian 9 to get a more recent operating system and applications.

I ran in to the following problems when trying to do this.

When trying to use asterisk 13.24.0-rc1, I ran into some strange problems with 
some of my custom scripts.

It seems the following statement immediately disconnects the user exten => 
s,n,Read(digit,,1,,1,20) ; read a digit

In the log after that statement it says user disconnected.  I have an agi which 
speaks some text before the read and that agi does not even say anything, 
although it does complete.

Now, if I try to go back to 13.15.0, it does not work at all because it keeps 
telling in my log that modules support is not available, so no modules get 
loaded.

Any ideas on thispuzzle would be appreciated.


--
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

 John Covici wb2una
 cov...@ccs.covici.com



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Re: [asterisk-users] Cannot originate to extension unless /etc/hosts is edited constantly? [Tony Mountfield]

[Stefan Viljoen]

Hi Tony

Ok, got this solved.



I discovered my AMI message was corrupt due to a bug in our third party dialer 
app we wrote ourselves...!

E. g. this worked on Asterisk 1.8:

ActionID=12edad43-e817-427b-aa21-31a9659f86e1
=Originate
=SIP/local/3035@local
=
=local
=1
=3035
=recordinglisten
=12edad43-e817-427b-aa21-31a9659f86e1
=12edad43-e817-427b-aa21-31a9659f86e1B
=CallLimit=3600,recfile=/var/spool/asterisk/monitor/archive/1901/15/201901151654r1g4679,altfile=/var/spool/asterisk/monitor/archive/1901/15/201901151654r1g4679
=true

But doesn't on Asterisk 13 - for a very good reason:

=SIP/local/3035@local

was acceptable to Asterisk 1.8, but NOT to Asterisk 13 - and I kind of agree.

It was a bug in our third party dialer app that instead of passing to Asterisk 
13

=SIP/3035@local

was passing

=SIP/local/3035@local

which is just WRONG.

This was why 13 was replying

[Jan 15 18:19:23] ERROR[10519]: netsock2.c:305 ast_sockaddr_resolve: 
getaddrinfo("local", "(null)", ...): Name or service not known
[Jan 15 18:19:23] WARNING[10519]: chan_sip.c:6316 create_addr: No such host: 
local

which makes perfect sense...!

So I fixed the incorrect variable reference in our app that was generating the 
wrong 

SIP/local/3035@local

channel, and recording playback started working correctly and no more of the 
above error messages.

So all not Asterisk fault, though 13 does appear to interpret the channel name 
differently as 

SIP/local/3035@local

DOES work on 1.8.32.3 (which it shouldn't, but it does.)

Thanks for taking the time to reply.



Regards

Stefan

---
Date: Tue, 15 Jan 2019 17:32:40 + (UTC)
From: t...@softins.co.uk (Tony Mountifield)
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Cannot originate to extension unless
/etc/hosts is edited constantly?
Message-ID: 

In article <018201d4acef$898a4b10$9c9ee130$@verishare.co.za>,
Stefan Viljoen  wrote:
> Hi Guys
> 
> I've run into a weird problem on Asterisk 13. Again something that worked 
> fine on 1.8 but is now broken on Asterisk 13.
> 
> I have an extension 3015. I'm trying to originate a recording playback call 
> on it via AMI by sending
> 
> Action: Originate
> ActionID: test
> Channel: SIP/3015
> Exten: 
> Context: local
> Priority: 1
> CallerID: 3015
> Account: recordinglisten
> ChannelID: abc
> OtherChannelID: def
> Variable: 
> CallLimit=3600,recfile=/var/spool/asterisk/monitor/1807/25/2507180836591192526,altfile=/var/spool/asterisk/monitor/archive/1807/25/2507180836591192526
> Async: true
> 
> My dialplan code:
> 
> exten=>,1,Answer()
> exten=>,n,NoOp(Requesting File ${recfile})
> exten=>,n,Set(${__recfile}=${recfile})
> exten=>,n,Set(${__altfile}=${altfile})
> exten=>,n,NoOp(Rec file set to ${recfile})
> exten=>,n,NoOp(Alt file set to ${altfile})
> exten=>,n,NoOp(Requesting Alt File ${altfile})
> exten=>,n,Set(__numbertarget=)
> exten=>,n(play),ControlPlayback(${recfile},2,6,4,8,5,9)
> exten=>,n(play2),ControlPlayback(${altfile},2,6,4,8,5,9)
> exten=>,n,hangup()
> 
> However, on sending the above to the AMI I get in the console of Asterisk 13:
> 
> [Jan 15 18:19:23] ERROR[10519]: netsock2.c:305 ast_sockaddr_resolve: 
> getaddrinfo("local", "(null)", ...): Name or service not known
> [Jan 15 18:19:23] WARNING[10519]: chan_sip.c:6316 create_addr: No such host: 
> local
> 
> and no recording playback takes place.
> 
> I've found that by editing /etc/hosts and adding
> 
> 172.56.4.11  local
> 
> where 172.56.4.11 is the phone SIP/3015's IP address, it works perfectly and 
> the recording starts playing once 3015 is answered.

What does your sip.conf look like? (without comments, e.g. grep -v '^;' 
sip.conf | grep -v '^$')

Particularly, do you have a separate section for each phone, e.g. [3015] ?

> On Asterisk 1.8 the above worked without having to constantly edit /etc/hosts 
> and having to constantly map and remap the one phone that is then capable of 
> playing back a recording.
> 
> This appears to be some kind of DNS / name resolution issue exclusive to 
> Asterisk 13 (for me) - how can I fix this, e. g. NOT get
> 
> [Jan 15 18:19:23] ERROR[10519]: netsock2.c:305 ast_sockaddr_resolve: 
> getaddrinfo("local", "(null)", ...): Name or service not known
> [Jan 15 18:19:23] WARNING[10519]: chan_sip.c:6316 create_addr: No such host: 
> local
> 
> and constantly having to edit /etc/hosts to get one of my phones (the one 
> tied to "local") to be able to play back a recording on Asterisk 13?
> 
> (This obviously is fatal anyway as I got lots of phones on which I want to 
> playback recordings and editing /etc/hosts for each phone is impossible if 

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

[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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[asterisk-users] Cannot originate to extension unless /etc/hosts is edited constantly?

[Stefan Viljoen]

Hi Guys

I've run into a weird problem on Asterisk 13. Again something that worked fine 
on 1.8 but is now broken on Asterisk 13.

I have an extension 3015. I'm trying to originate a recording playback call on 
it via AMI by sending

Action: Originate
ActionID: test
Channel: SIP/3015
Exten: 
Context: local
Priority: 1
CallerID: 3015
Account: recordinglisten
ChannelID: abc
OtherChannelID: def
Variable: 
CallLimit=3600,recfile=/var/spool/asterisk/monitor/1807/25/2507180836591192526,altfile=/var/spool/asterisk/monitor/archive/1807/25/2507180836591192526
Async: true

My dialplan code:

exten=>,1,Answer()
exten=>,n,NoOp(Requesting File ${recfile})
exten=>,n,Set(${__recfile}=${recfile})
exten=>,n,Set(${__altfile}=${altfile})
exten=>,n,NoOp(Rec file set to ${recfile})
exten=>,n,NoOp(Alt file set to ${altfile})
exten=>,n,NoOp(Requesting Alt File ${altfile})
exten=>,n,Set(__numbertarget=)
exten=>,n(play),ControlPlayback(${recfile},2,6,4,8,5,9)
exten=>,n(play2),ControlPlayback(${altfile},2,6,4,8,5,9)
exten=>,n,hangup()

However, on sending the above to the AMI I get in the console of Asterisk 13:

[Jan 15 18:19:23] ERROR[10519]: netsock2.c:305 ast_sockaddr_resolve: 
getaddrinfo("local", "(null)", ...): Name or service not known
[Jan 15 18:19:23] WARNING[10519]: chan_sip.c:6316 create_addr: No such host: 
local

and no recording playback takes place.

I've found that by editing /etc/hosts and adding

172.56.4.11  local

where 172.56.4.11 is the phone SIP/3015's IP address, it works perfectly and 
the recording starts playing once 3015 is answered.

On Asterisk 1.8 the above worked without having to constantly edit /etc/hosts 
and having to constantly map and remap the one phone that is then capable of 
playing back a recording.

This appears to be some kind of DNS / name resolution issue exclusive to 
Asterisk 13 (for me) - how can I fix this, e. g. NOT get

[Jan 15 18:19:23] ERROR[10519]: netsock2.c:305 ast_sockaddr_resolve: 
getaddrinfo("local", "(null)", ...): Name or service not known
[Jan 15 18:19:23] WARNING[10519]: chan_sip.c:6316 create_addr: No such host: 
local

and constantly having to edit /etc/hosts to get one of my phones (the one tied 
to "local") to be able to play back a recording on Asterisk 13?

(This obviously is fatal anyway as I got lots of phones on which I want to 
playback recordings and editing /etc/hosts for each phone is impossible if two 
phones want to listen to different recordings at the same time- /etc/hosts can 
only contain one "local").

How can I fix this?

Thanks

Stefan


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

[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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Re: [asterisk-users] Outbound caller ID ignored

[Stefan Viljoen]

Hi Mitch

I don't think you can expect it to work on the analog lines at all out of the 
box.

What kind of hardware are you using to connect to the analog lines?

We used Xorcom Astribanks at a stage, but could never (ever) figure out the 
signalling needed to set the caller ID at our local public phone company 
servicing our POTS lines (Telkom, South Africa). It was useless anyway, as they 
didn't support it and their software couldn't apply it. See below.

It is a two-pronged problem, e. g. your hardware and drivers need to be able to 
signal the caller-id to the carrier / public telephone company, and, if you CAN 
communicate this to the local telephone company, they need to be able to 
interpret this and set it on calls they process for you.

In our case (if I remember right, we've long since dumped all analog lines) 
there were double problems there as well, e. g. the POTS network here ignored 
our signalling (and their hardware / software mix doesn't support it anyway, so 
even if they interpreted and understood the protocol and number being sent, 
they couldn't apply it) and some legislation that prohibited "CID masquerading" 
for calls made through the government-owned public telephone network.

Our current SIP trunk provider cannot do it either, their systems simply do not 
support it and all numbers going out are always fixed to the constant trunk 
outgoing number. E. g. we also spent several days of dev trying to get it to 
work, eventually checking with them and being informed it is impossible anyway 
and they do not support changing the number from traffic we pass them.

However, they can easily spin up extra logical SIP trunks at miniscule cost 
(several US cents in local currency equivalent), so if we need a custom number 
for callees to dial back for a temporary sales campaign or whatever, we simply 
request it and that trunks fixed, constant number will appear on callee phones. 
We then simply route the correct internal dialout calls to the temporary 
virtual trunk by whatever logic we want.

Hope this helps.

Kind regards,
--
Message: 2
Date: Sun, 13 Jan 2019 09:45:27 -0600
From: Mitch Claborn 
Subject: [asterisk-users] Outbound caller ID ignored

Setting the outbound caller ID works fine on our PRI (T1) lines, but does not 
work on our local POTS lines. No errors in the logs, the new caller ID just 
seems to be ignored. Should I expect it to work on the analog lines?

Dial plan:
same =>n,Set(CALLERID(all)=111222)
same =>n(dialit),Dial(DAHDI/50/1222333,30)

Channels:
signalling=fxs_ks
callerid=asreceived
group=20,21
context=from-pstn
faxdetect=incoming
faxdetect_timeout=0
faxbuffers => 12,half
channel => 49-53


-- 

Mitch


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Re: [asterisk-users] Switched from Asterisk 1.8 to 13 - CDR ringtime now always zero (Joel)

[Stefan Viljoen]

>Hi,

>On the other side.. There is a specific note regarding CDR behavior changes 
>from asterisk 12 onwards. So going from 1.8 to 13 means it applies to you. 

>Have a look at:

>https://wiki.asterisk.org/wiki/display/AST/Upgrading+to+Asterisk+12 

>And 

>https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+CDR+Specification

Hi Joel

I've read the above in detail, and studied them, but I cannot find anything 
that explains the CDR behaviour I'm seeing in Asterisk 13.

The only thing relevant is the following in the Upgrade Notes:

---
The duration, billsec, start, answer, and end times now reflect the times 
associated with the current CDR for the channel, as opposed to a cumulative 
measurement of all CDRs for that channel.
---

That is exactly where the problem lies, e. g. the above behaviour is exactly 
what is NOT happening...

E. g. instead of the origination channel created by the AMI call we make as 
previously described, having its CDR written and done with, its CDR data for 
the originating call leg apparently gets propagated into the resulting outgoing 
call leg of the call to the AMI originate command.

E. g. my origination channel's CDR is

Start: 2019-01-11 08:22:07
Answer: 2019-01-11 08:22:12
End: 2019-01-11 08:24:09

and the associated outgoin channel's CDR is

Start: 2019-01-11 08:22:12
Answer: 2019-01-11 08:22:12
End: 2019-01-11 08:24:09

e.g. it appears that the origination channel's pickup time of Answer: 
2019-01-11 08:22:12 gets propagated into the outgoing channel's answer time of 
Answer: 2019-01-11 08:22:12.

E. g. the above in the release notes is 180 degrees the opposite of how 
Asterisk 13 actually behaves in practice?

Despite the above release note, it DOES appear in fact that CDR measurement is 
cumulative... very definitely not distinct

Thanks for the reply!

Regards

Stefan



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Re: [asterisk-users] Switched from Asterisk 1.8 to 13 - CDR

[Stefan Viljoen]

>> Is it possible that the setup part of the call (between initiation and 
>> answer)
>> is recorded in a separate CDR?

>An excellent question. Unlike in the past versions calls can actually generate 
>multiple CDRs because CDRs now represent the flow of communication between 
>things.

>Providing the actual CDR records that were generated as well as console output 
>would allow better understanding of what is going on.

>--
>Joshua C. Colp
>Digium - A Sangoma Company | Senior Software Developer

Hi Joshua

That is exactly what is happening. 

I get a separate CDR for the originate of the call to the agent's extension, 
and then a separate CDR for the call that then goes out to the client.

The separate CDR for an originate event DOES appear to be correct, there is 
usually a second or two seconds worth of ringtime indicated, e. g. start and 
answer will vary (as outgoing calls did on 1.8) under 13:

Start  | Answer | End
2019-01-11 08:01:20 | 2019-01-11 08:01:22 | 2019-01-11 08:01:25

E. g. the agent originated by calling the AMI via our app at 2019-01-11 
08:01:20, he picked up the phone at 2019-01-11 08:01:22 (e. g. he let it ring 
for 2 seconds) etc.

So the "originate call" to the agent's local channel has correct difference 
between start and and start, but the actual call going out to the client from 
the agent has the "13 sickness" e. g. start and answer times are exactly the 
same.

Here are sample actual CDR records (apologies for the format):

id  clidsrc dst dcontextchannel dstchannel  lastapp 
lastdatastart   answer  end durationbillsec disposition 
amaflagsaccountcode uniqueiduserfield   linkedid
sequencecustomdata
8132380b-1569-11e9-9184-18d6c702b3d76140_ctd61406140local   
Local/6140@local-0006232d;2 SIP/6140-004e99bd   DialSIP/6140,120,tT 
2019-01-11 08:22:07 2019-01-11 08:22:12 2019-01-11 08:24:09 122 
116 ANSWERED3   201901110822sobtpb8 201901110822sobtpb8B
/var/spool/asterisk/monitor/1901/11/201901110822sobtpb8.gsm 
201901110822sobtpb8 9217937 

813671dc-1569-11e9-9184-18d6c702b3d7614061405554567781  local   
Local/6140@local-0006232d;1 SIP/centra-out-rng-004e99c5 Dial
SIP/centra-out-rng/5554567781,120,TL(390:6:3)   2019-01-11 
08:22:12 2019-01-11 08:22:12 2019-01-11 08:24:09 116 116 
ANSWERED3   201901110822sobtpb8 201901110822sobtpb8 
/var/spool/asterisk/monitor/1901/11/201901110822sobtpb8.gsm 
201901110822sobtpb8 9217936 

E. g. the first line is the originate to extension 6140. As you can see on the 
times, the agent waited 5 seconds after the phone started ringing internally, 
to pick up the originated call:

Start: 2019-01-11 08:22:07
Answer: 2019-01-11 08:22:12
End: 2019-01-11 08:24:09

As you can see the "origination call" CDR's end time exactly spans the "actual 
call" duration.

The second line is the actual call to the customer at 5554567781. As you can 
see on the times, apparently the client picked up instantly (which they did 
not, if I listend to the recording in 
/var/spool/asterisk/monitor/1901/11/201901110822sobtpb8.gsm):

Start: 2019-01-11 08:22:12
Answer: 2019-01-11 08:22:12
End: 2019-01-11 08:24:09

As you can see the "actual call" CDR's start and end times are exactly the 
same, even though the client picked up only after several seconds (8, in this 
case, if you listen to the recording with a stopwatch and time start of audio 
and actual pickup.)

E. g. it is impossible to now tell how long the agent waited for the client to 
pick up using the above model of originating calls, whereas it was working with 
1.8, 13 always emits the same times to the second for both start and answer in 
the outgoing leg of the call in the above scenario.

The "origination leg" of the call's total time spans the outgoing leg, but the 
start / answer times in the "origination leg" is the internal pickup of the 
agent of his phone after he click "Originate call" in our app which calls the 
AMI.

Again, on 1.8, both CDRs as above were created with the same dialplan code as 
communicated yesterday, but on 1.8 each of the two CDRs start and answer times 
differ, the one CDR emitted indicating how long the agent took to pickup the 
phone after it started ringing for his click of the Originate button, and the 
other CDR emitted indicating (in varying start and answer times) how long the 
client took to pickup the phone.

-

Console output for a typical originated call:

core set verbose 3
Console verbose was OFF and is now 3.
-- Called 6062@local
-- Executing [6062@local:1] Dial("Local/6062@local-0001984b;2", 
"SIP/6062,120,t") in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/6062
-- SIP/6062-000600f5 is ringing
-- Local/6062@local-0001984b;1 is 

Re: [asterisk-users] Switched from Asterisk 1.8 to 13 - CDR ringtime now always zero (Tony)

[Stefan Viljoen]

In article <009d01d4a7e0$af2e0a50$0d8a1ef0$@verishare.co.za>,
Stefan Viljoen  wrote:
>> Regarding this I've read the specs linked to in detail, but I can find 
>> no mention anywhere of any change that implies or states that no ring 
>> time will be recorded anymore in Asterisk 13 and that all times in start and 
>> answer columns will now be equal for all calls.
>> 
>> Can this be because I nowhere use the Answer() application in my dialplan 
>> when dialing out?

>You shouldn't Answer() the originating channel before calling Dial() on it.
>You should allow Dial() to propagate the answer, busy or other failure from 
>the destination channel back to the originating channel.

Hi Tony

Yes, that is exactly what I'm doing... no Answer() calls anywhere in the 
dialout parts of my dialplan, as detailed in my previous posts.

>Is it possible that the setup part of the call (between initiation and answer) 
>is recorded in a separate CDR?

Yes, that is correct, I get a separate CDR for the originate of the call to the 
agent's extension, and then a separate CDR for the call that then goes out to 
the client.

The separate CDR for an originate event DOES appear to be correct, there is 
usually a second or two seconds worth of ringtime indicated, e. g. start and 
answer will vary (as outgoing calls did on 1.8) under 13:

Start  | Answer | End
2019-01-11 08:01:20 | 2019-01-11 08:01:22 | 2019-01-11 08:01:25

E. g. the agent originated by calling the AMI via our app at 2019-01-11 
08:01:20, he picked up the phone at 2019-01-11 08:01:22 (e. g. he let it ring 
for 2 seconds) etc.

This is exactly what I need to have, but not on the origination calls, the 
outgoing calls under Asterisk 13 - e. g. that there is a difference between the 
Start and Answer times, which = ringtime.

For the origination calls, duration and billsec is correct as well, just as it 
was under 1.8 too.

It is just with 13 that the origination calls are STILL correct, as they were 
under 1.8, but with 13 the outgoing calls now indicate no ringtime.

Thanks for the reply!

Stefan


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Re: [asterisk-users] Switched from Asterisk 1.8 to 13 - CDR ringtime now always zero (Joshua C. Colp)

[Stefan Viljoen]

Message: 2
Date: Mon, 07 Jan 2019 06:07:54 -0500
From: "Joshua C. Colp" 

>> On Mon, Jan 7, 2019, at 3:04 AM, Stefan Viljoen wrote:
>> Hi guys
.
.
.
>> E. g. on 13, I see this (zero ringtime) for a call that I make to my 
>> cellphone to test, with my cellphone ringing for at least 10 seconds 
>> and ringing heard on the Yealink connected to the asterisk - e. g.
>> completely wrong:

>This is the way it is supposed to work[1], but it ultimately depends on your 
>dialplan. Are you using Local channels? Are you doing Answer in the dialplan? 
>What is the complete flow?

Hi Joshua

Thank you for the reply. I'll go read the spec in detail (which is probably 
what I should have done in the first place anyway.)

Yes, I am using local channels and generating the calls via AJAM by calling the 
Originate AMI / AJAM application. The "local" extension that is calling out is 
defined as 

local/@local 

where  is the extension number, e. g. 3509, 3175, or whatever.

So my AJAM Originate command is

ActionID=201901080814t4qn82v
Action=Originate
Channel=local/3916@local
Exten=0825588996
Context=local
Priority=1
CallerID=3916_ctd
Account=201901080814t4qn82v
ChannelID=201901080814t4qn82v
OtherChannelID=201901080814t4qn82vB
Variable=__CallLimit=3600
Async=true

I'm not calling the Answer application in the dialplan when dialing out.

Effectively the flow is the user clicks dial in our external application. This 
sends the above AJAM command to Asterisk, with parameters as specified. 
Asterisk originates the call on his extension as passed in the AJAM command, he 
picks up and the call then goes into the local context and gets routed. The 
STDOUT macro is called which does some prep (lots of 1.8 legacy stuff still in 
here) and this then calls VCCALLOUT, which contains the call to the dial() 
application.

VCCALLOUT calls

exten=>s,n(dialNoLimit),Dial(${chantouse}/${numtodial},60,TL(390:6:3))

${chantouse} will be something like SIP/sip-trunk-name
${numtodial} will be the target number, e. g. 27825588996

to dial out.

---

Detail:

My local context (simplified):

[local]

exten=>_082xxx,1,Macro(STDOUT,SIP/centra-out,27${EXTEN:1})

and the STDOUT macro, edited for brevity:

[macro-STDOUT]
;${ARG1} = channel
;${ARG2} = number
exten=>s,1,Macro(WAITCHANNEL)
exten=>s,n,Macro(WAITCDR)
exten=>s,n,Macro(VCRECORD,${MACRO_CONTEXT}X${CALLERID(num)}ACC${CHANNEL(accountcode)},${ARG2})
exten=>s,n(dodial),Macro(VCCALLOUT,${ARG1},${ARG2},${ARG3},${ARG4})
exten=>s,n,NoOp(Setting Userfield after call completion)
exten=>s,n,Set(CDR(userfield)=${MIXMONITOR_FILENAME})
exten=>s,dodial+101,Busy()
exten=>s,n,Hangup()
exten=>s,n,MacroExit

exten=>h,1,NoOp(Call hangup MACROSTDOUT)
exten=>h,n,UserEvent(RecordingToFile,Uniqueid: ${UNIQUEID},Channel: 
${Channel},Filename: ${MIXMONITOR_FILENAME})
exten=>h,n,NoOp(Recorded to ${MIXMONITOR_FILENAME})
exten=>h,n,Set(CDR(userfield)=${MIXMONITOR_FILENAME})
exten=>h,n,NoOp(Account code is ${CHANNEL(accountcode)})
exten=>h,n,NoOp(call link var is ${call_Link})
exten=>h,n,GotoIf($["${CHANNEL(accountcode)}" != ""]?done)
exten=>h,n(setacc),Set(CHANNEL(accountcode)=${call_Link})
exten=>h,n(done),noOp(Call Completed)

and the VCCALLOUT macro:

[macro-VCCALLOUT] ;macro to dial numbers
; ${ARG1} Channel To Use
; ${ARG2} Number To Dial
; ${ARG3} FailOver Channel
; ${ARG4} FailOverNumber
exten=>s,n(setchan),Set(chantouse=${ARG1})
exten=>s,n,Set(numtodial=${ARG2})
exten=>s,n(makecall),GotoIf($["${timeLimit}" = ""]?dialNoLimit:dialLimit)
exten=>s,n(dialNoLimit),Dial(${chantouse}/${numtodial},60,TL(390:6:3))
exten=>s,n,NoOp(Dial Status: ${DIALSTATUS})
exten=>s,n,GoTo(s-${DIALSTATUS},1)
exten=>s,n(dialLimit),Dial(${chantouse}/${numtodial},60,TL(390:6:3))
exten=>s,n,NoOp(Dial Status: ${DIALSTATUS})
exten=>s,n,GoTo(s-${DIALSTATUS},1)
exten=>s,dialNoLimit+101,Goto(s-${DIALSTATUS},1)
exten=>s,dialLimit+101,Goto(s-${DIALSTATUS},1)
exten=>s,n(endcall),busy()
exten=>s,n,NoOp(Call Completed - setting userfield to recording)
exten=>s,n,Set(CDR(userfield)=${MIXMONITOR_FILENAME})
exten=>s,n,Hangup()
exten=>s,n,MacroExit

exten=>s-NOANSWER,1,goto(s,endcall)
exten=>s-CANCEL,1,goto(s,endcall)
exten=>s-BUSY,1,goto(s,endcall)

exten=>h,1,NoOp(Call Hungup)
exten=>h,n,UserEvent(RecordingToFile,Uniqueid: ${UNIQUEID},Channel: 
${Channel},Filename: ${MIXMONITOR_FILENAME})
exten=>h,n,NoOp(Recorded to ${MIXMONITOR_FILENAME})
exten=>h,n,Set(CDR(userfield)=${MIXMONITOR_FILENAME})
exten=>h,n,NoOp(Account code is ${CHANNEL(accountcode)})
exten=>h,n,NoOp(call link var is ${call_Link})
exten=>h,n,GotoIf($["${CHANNEL(accountcode)}" != ""]?done)
exten=>h,n(setacc),Set(CHANNEL(accountcode)=${call_Link})
exten=>h,n(done),noOp(Call

Re: [asterisk-users] Switched from Asterisk 1.8 to 13 - CDR ringtime now always zero (Joshua C. Colp)

[Stefan Viljoen]

Regarding this I've read the specs linked to in detail, but I can find no 
mention anywhere of any change that implies or states that no ring time will be 
recorded anymore in Asterisk 13 and that all times in start and answer columns 
will now be equal for all calls.

Can this be because I nowhere use the Answer() application in my dialplan when 
dialing out?

-Original Message-
From: Stefan Viljoen  
Sent: Tuesday, 08 January 2019 08:49
To: 'asterisk-users@lists.digium.com' 
Subject: RE: Re: Switched from Asterisk 1.8 to 13 - CDR ringtime now always 
zero (Joshua C. Colp)

Message: 2
Date: Mon, 07 Jan 2019 06:07:54 -0500
From: "Joshua C. Colp" 

>> On Mon, Jan 7, 2019, at 3:04 AM, Stefan Viljoen wrote:
>> Hi guys
.
.
.
>> E. g. on 13, I see this (zero ringtime) for a call that I make to my 
>> cellphone to test, with my cellphone ringing for at least 10 seconds 
>> and ringing heard on the Yealink connected to the asterisk - e. g.
>> completely wrong:

>This is the way it is supposed to work[1], but it ultimately depends on your 
>dialplan. Are you using Local channels? Are you doing Answer in the dialplan? 
>What is the complete flow?

Hi Joshua

Thank you for the reply. I'll go read the spec in detail (which is probably 
what I should have done in the first place anyway.)

Yes, I am using local channels and generating the calls via AJAM by calling the 
Originate AMI / AJAM application. The "local" extension that is calling out is 
defined as 

local/@local 

where  is the extension number, e. g. 3509, 3175, or whatever.

So my AJAM Originate command is

ActionID=201901080814t4qn82v
Action=Originate
Channel=local/3916@local
Exten=0825588996
Context=local
Priority=1
CallerID=3916_ctd
Account=201901080814t4qn82v
ChannelID=201901080814t4qn82v
OtherChannelID=201901080814t4qn82vB
Variable=__CallLimit=3600
Async=true

I'm not calling the Answer application in the dialplan when dialing out.

Effectively the flow is the user clicks dial in our external application. This 
sends the above AJAM command to Asterisk, with parameters as specified. 
Asterisk originates the call on his extension as passed in the AJAM command, he 
picks up and the call then goes into the local context and gets routed. The 
STDOUT macro is called which does some prep (lots of 1.8 legacy stuff still in 
here) and this then calls VCCALLOUT, which contains the call to the dial() 
application.

VCCALLOUT calls

exten=>s,n(dialNoLimit),Dial(${chantouse}/${numtodial},60,TL(390:6:3))

${chantouse} will be something like SIP/sip-trunk-name ${numtodial} will be the 
target number, e. g. 27825588996

to dial out.

---

Detail:

My local context (simplified):

[local]

exten=>_082xxx,1,Macro(STDOUT,SIP/centra-out,27${EXTEN:1})

and the STDOUT macro, edited for brevity:

[macro-STDOUT]
;${ARG1} = channel
;${ARG2} = number
exten=>s,1,Macro(WAITCHANNEL)
exten=>s,n,Macro(WAITCDR)
exten=>s,n,Macro(VCRECORD,${MACRO_CONTEXT}X${CALLERID(num)}ACC${CHANNEL(accountcode)},${ARG2})
exten=>s,n(dodial),Macro(VCCALLOUT,${ARG1},${ARG2},${ARG3},${ARG4})
exten=>s,n,NoOp(Setting Userfield after call completion)
exten=>s,n,Set(CDR(userfield)=${MIXMONITOR_FILENAME})
exten=>s,dodial+101,Busy()
exten=>s,n,Hangup()
exten=>s,n,MacroExit

exten=>h,1,NoOp(Call hangup MACROSTDOUT)
exten=>h,n,UserEvent(RecordingToFile,Uniqueid: ${UNIQUEID},Channel: 
${Channel},Filename: ${MIXMONITOR_FILENAME}) exten=>h,n,NoOp(Recorded to 
${MIXMONITOR_FILENAME})
exten=>h,n,Set(CDR(userfield)=${MIXMONITOR_FILENAME})
exten=>h,n,NoOp(Account code is ${CHANNEL(accountcode)}) exten=>h,n,NoOp(call 
link var is ${call_Link}) exten=>h,n,GotoIf($["${CHANNEL(accountcode)}" != 
""]?done)
exten=>h,n(setacc),Set(CHANNEL(accountcode)=${call_Link})
exten=>h,n(done),noOp(Call Completed)

and the VCCALLOUT macro:

[macro-VCCALLOUT] ;macro to dial numbers ; ${ARG1} Channel To Use ; ${ARG2} 
Number To Dial ; ${ARG3} FailOver Channel ; ${ARG4} FailOverNumber
exten=>s,n(setchan),Set(chantouse=${ARG1})
exten=>s,n,Set(numtodial=${ARG2})
exten=>s,n(makecall),GotoIf($["${timeLimit}" = ""]?dialNoLimit:dialLimit)
exten=>s,n(dialNoLimit),Dial(${chantouse}/${numtodial},60,TL(390:6:3))
exten=>s,n,NoOp(Dial Status: ${DIALSTATUS})
exten=>s,n,GoTo(s-${DIALSTATUS},1)
exten=>s,n(dialLimit),Dial(${chantouse}/${numtodial},60,TL(390:6:3))
exten=>s,n,NoOp(Dial Status: ${DIALSTATUS})
exten=>s,n,GoTo(s-${DIALSTATUS},1)
exten=>s,dialNoLimit+101,Goto(s-${DIALSTATUS},1)
exten=>s,dialLimit+101,Goto(s-${DIALSTATUS},1)
exten=>s,n(endcall),busy()
exten=>s,n,NoOp(Call Completed - setting userfield to recording)
exten=>s,n,Set(CDR(userfield)=${MIXMONITOR_FILENAME})
exten=>s,n,Hangup()
exten=>s,n,MacroExit

exten=>s-NOANSWER,1,goto(s,endcall)
exten=>s-CANCEL

[asterisk-users] Switched from Asterisk 1.8 to 13 - CDR ringtime now always zero

[Stefan Viljoen]

Hi guys

A few months ago I upgraded most of my Asterisk servers to 13 from 1.8. I've 
still got about 25% of my servers on 1.8.

I've since noticed that ringtime on Asterisk 13 - the time difference between 
"start" and "answer" in the CDR record for any call, and between "duration" and 
"billsec" - has completely disappeared. E. g. the two times and two durations 
are now the same for all outgoing calls made on Asterisk 13.

On 1.8 the time difference between "start" and "answer" and "duration" and 
"billsec" was always my ring time - e. g. if I phone out to my cellphone from 
one of my 1.8 servers, the amount of seconds the call rings on my cell in my 
1.8 instances is the difference between "start" and "answer" in the 
1.8-generated CDR record, and the difference between "duration" and "billsec".

E. g. on 13, I see this (zero ringtime) for a call that I make to my cellphone 
to test, with my cellphone ringing for at least 10 seconds and ringing heard on 
the Yealink connected to the asterisk - e. g. completely wrong:

Start  | Answer | End   
 | dur.| billsec
2019-01-07 08:13:49 | 2019-01-07 08:13:49 | 2019-01-07 08:14:35 |   45 |
45

when it should have shown duration 45 billsec 30, 'cause I let it ring 10 
seconds before answering my cell / mobile.

On 1.8, I see this for a call that I make to my cellphone to test, with the 
phone ringing for 10 seconds and ringing heard on the Yealink connected to the 
asterisk - e. g. correct:

Start  | Answer | End   
 | dur. | billsec
2019-01-07 08:26:43 | 2019-01-07 08:26:53 | 2019-01-07 08:29:25 |  162 |
 152 

e. g. it shows clearly that the call was ringing for 10 seconds before I 
answered my cell / mobile.

My billing logic can thus tell that on Asterisk 1.8 there was 10 seconds of 
ringtime, but on Asterisk 13 it says that there was 0 seconds of ringtime - 
which in reality was also 10 seconds, but is no longer written as such into the 
CDR on Asterisk 13.

So on all my Asterisk 13 instances all my outgoing calls are "suddenly" 
immediately answered by whomever I'm calling, and clients' landlines / mobiles  
never ring at all, they just instantly answer... which is obviously incorrect.

How can I get Asterisk 13 to behave like Asterisk 1.8 as regards CDRs, e. g. 
still recording ring time and not just make all call start and call answer 
times, and all durations and billsec counts, exactly the same?

I'm using the same SIP trunk provider with 13 as with 1.8, and in all cases the 
same operating system (Centos 7, upgraded to date) with both my 1.8 and 13 
instances, and almost always the same physical hardware, same network, etc. The 
CDRs are logging locally on the Asterisk itself in both my 1.8 and 13.22.0 
instances, unixODBC connections to Percona 5.6 running locally on the same box.

So why is there no longer any ringing recorded in CDRs Asterisk 13 generates? 
It seems some element of logic changed in the CDR engine, but I cannot find any 
references to such... anybody got any help or pointers?

The holy grail would be to have Asterisk 13 record ringing time on its CDR 
records, e. g. have a variation between the time a call is shown as having 
started and the time it is shown as having answered, e. g. the time spent in 
ringing waiting for the callee to pick up.

So 1.8 CDR ringtime behaviour - but in 13.

Any ideas?

Thanks,

Stefan


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[asterisk-users] CURL to post application/json (David P)

[Stefan Viljoen]

>We tried to use the CURL fn to POST json, but it's sent as form data and
>there seems no support for changing the Content-Type header. We switched to
>invoking curl in the shell.

Hi David

If you've got if fixed that way, great.

Just thought I'd comment and share the scripts / manner we use this, using 
CURL, and mention one major gotcha I ran into that took me a few hours to solve 
related with using CURL in Centos 7 on Asterisk 1.8.32.3, and then switching to 
13.22.0.

This is what our dialplan logic looks like to get a UUID from a CURL call made 
in a BASH file to our back-end application's API.

Dialplan:

same=>n,Set(curlResult=${SHELL(/usr/src/bash/getUUID.sh)})
---

BASH file:

getUUID.sh
---
#!/bin/bash
function jsonval {
temp=`echo $jsonResult | sed 's/\//\//g' | sed 's/[{}]//g' | awk -v 
k="text" '{n=split($0,a,","); for (i=1; i<=n; i++) print a[i]}' | sed 
's/\"\:\"/\|/g' | sed 's/[\,]/ /g' | sed 's/\"//g'`
echo ${temp##*|}
}

baseUrl=http://127.0.0.1/api/getUUID

jsonResult=$(curl --connect-timeout 16 --max-time 32 -s -X GET $baseUrl)

uuid=`jsonval`

echo -n $uuid
---

This returns our UUID (from the Java app) into the dialplan variable curlResult 
which we then use further on in the dialplan.

The MAJOR gotcha in this above is the

>echo -n $uuid

line...

On Asterisk 1.8.32.3, this worked fine in the BASH script:

echo $uuid

However, on Asterisk 13.22.0, it did NOT work - the dialplan variable 
curlResult stayed empty on any calls to the BASH script vir SET() in Asterisk.

Turns out that 13.22.0 does NOT like line feed / new line (\n) in ANY strings 
that come back from BASH.

So I had to change the above line in the BASH file to

echo -n $uuid

so that the line-feed / newline was omitted from being returned to the Asterisk 
13 instance.

Just some two cents worth.

Regards

Stefan


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[asterisk-users] Spontaneous reboot due to MySQL lookups ? (Jonas Kellens)

[Stefan Viljoen]

Hi Jonas

We have more or less this behaviour with 1.8.32.3 when writing CDRs to ODBC on 
Percona 5.6 (MySQL drop-in replacement with some optimisations and extra 
features.)

In our case the system does NOT reboot, but Asterisk itself crashes with a 
segfault inside unixODBC itself.

We're running on Centos 7, so the first thing we tried was to upgrade the ODBC 
drivers to the most recent version, and also use the most recent MySQL ODBC 
connector.

This did not solve the problem, and the spontaneous shutdowns (and then 
restarts via Centos 7 systemd) of the 1.8.32.3 instances continued.

Eventually we traced this to contention on the CDR table itself in the 
associated MySQL / Percona instance. We had a stored procedure running on the 
CDR table at set intervals via a MySQL timed event that was converting the 
Asterisk CDR data into another table we use for our own purposes. This was 
effectively locking the CDR table in ways that wasn't allowing Asterisk 
1.8.32.3 to enter CDRs when it wanted to.

The practical effect in real-world terms was that people could make a call, but 
then NOT do another call on that same extension, for up to five or ten minutes 
at a time. This was because the Asterisk 1.8.32.3 instance was waiting to get 
access to the CDR table to write that extensions most-recent CDR record in, and 
was holding onto the extension to prevent a new call when it hadn't yet managed 
to write into the CDR table for the previous call.

Also, if pushed hard enough with concurrent calls, the Asterisk instance would 
get segfaulted due to a segfault in the ODBC drivers, due to this contention 
and waiting for access to the CDR table.

All I can recommend is to upgrade to, for example, Asterisk 13.22.0 which is 
relatively painless to upgrade to if you're on 1.8.32.3 or the 1.8 series (at 
least in our experience) - OR remove contention on the CDR table.

We eventually removed the extra SP we had running on the MySQL / Percona CDR 
table, as the contention problems persisted with 13.22.0, BUT IT DOES NOT CRASH 
LIKE 1.8.32.3 - while running against the same Percona 5.6 MySQL version, with 
the same level of MySQL DB contention taking place.

E. g. look for contention on the MySQL tables you use, or just SQL contention 
in general - generate enough contention (with tables locked when Asterisk tries 
to write to them) and it does seem that 1.8 series Asterisk instances will 
segfault in the ODBC driver, if the concurrent call count goes high enough.

Or (the better option) upgrade to at least Asterisk 13 (in our experience).

Even with the same contention on MySQL db's accessed with unixODBC 2.3.6, and 
our same table-access killer SP running, Asterisk 13.22.0 doesn't crash at all 
when writing with the same version unixODBC driver to the contentious tables in 
MySQL. (We still had the same problem with people only being capable of making 
one call and then waiting up to five minutes post-hangup before being able to 
make the next call, but we solved that by not running the SP on schedule 
anymore against the CDR and CEL tables.)

E.g. effectively we solved the crashes by upgrading from 1.8.32.3 to 13.22.0 - 
the rest is incidentals unique to our usage case.

Hope this helps.

Date: Thu, 4 Oct 2018 17:10:01 +0200
From: Jonas Kellens 
To: Asterisk Users Mailing List - Non-Commercial Discussion

Subject: [asterisk-users] Spontaneous reboot due to MySQL lookups ?
Message-ID: 
Content-Type: text/plain; charset="utf-8"; Format="flowed"

Hello

using Asterisk 1.8.32.

I notice that there is a spontaneous reboot of the Asterisk system from time to 
time.

When I look in the logs (verbose file) I noticed that every time this occurs 
it's at a moment that there is a MySQL action, be it a lookup or an 
insert/update/delete.

I must say I do have some MySQL queries that occur in my dialplan when a call 
comes in, to look up different actions to perform on this call.


An idea how to overcome this problem ? Seems a "performance" issue, no ?!

Is it better to have these MySQL queries to be done by an external 
script (like a php script that I call with the System()-command or a 
SHELL()-command) ?


Here are some examples from the verbose file.



[Aug 22 15:19:10] VERBOSE[2977] pbx.c: [Aug 22 15:19:10] -- 
Executing [s@sub-GetAlertInfo:3] MYSQL("SIP/SipAgenT01-317d", 
"Connect connid localhost myuser mypwd myDB") in new stack
[Aug 22 15:19:10] VERBOSE[2977] pbx.c: [Aug 22 15:19:10] -- 
Executing [s@sub-GetAlertInfo:5] MYSQL("SIP/SipAgenT01-317d", "Query 
resultid 1 SELECT uri, callinfo FROM distringtone WHERE onoff='1'") in 
new stack
[Aug 22 15:19:18] VERBOSE[3306] config.c: [Aug 22 15:19:18]   == Parsing 
'/etc/asterisk/logger.conf': [Aug 22 15:19:18] VERBOSE[3306] config.c: 
[Aug 22 15:19:18]   == Found
[Aug 22 15:19:18] VERBOSE[3306] config.c: [Aug 22 15:19:18]   == Parsing 
'/etc/asterisk/asterisk.conf': [Aug 22 15:19:18] VERBOSE[3306] config.c: 
[Aug 22 15:19:18]   == Found
[Aug 22 

Re: [asterisk-users] Asterisk 13.22.0 - No channel type registered for 'Agent' when queue rings - solved

[Stefan Viljoen]

Hi Guys

Found the solution for this...!

https://wiki.asterisk.org/wiki/display/AST/New+in+12#Newin12-channels_chan_a
gent

and

https://reviewboard.asterisk.org/r/2657/diff/1/

and

https://blogs.asterisk.org/2016/02/10/converting-from-chan_agent-to-app_agen
t_pool/

clarifies the situation. After Asterisk 12, chan_agent  was removed from
Asterisk.

It has been replaced with the AgentRequest application.

So I had to adapt my previously posted configs as follows to get queued
agents working and answering incoming calls to the queue.

My agents.conf stayed unchanged from my original post.

My queues.conf had to change from my original post to:

---
member=> Local/3997@internal,,3997,Agent:3997
member=> Local/3998@internal,,3998,Agent:3998
---

with the rest of the queue settings remaining the same.

The agent login extension had to change to:

---
exten=>1007,1,NoOp(Login agent from ${CALLERID(num)})
 same=>n,Set(MONITOR_OPTIONS=b)
 same=>n,UnpauseQueueMember(,Local/${CALLERID(num)}@internal)
 same=>n,AgentLogin(${CALLERID(num)},s)
---

In my [internal] context (where my phones live - including my test phones
3997 and 3998) I had to add extensions that would call the AgentRequest()
dialplan application on each of them if that extension was dialled. This is
effectively how the queue is answered:

---
[internal]
.
.
.
exten=>3997,1,AgentRequest(3997)
 same=>n,Hangup()

exten=>3998,1,AgentRequest(3998)
 same=>n,Hangup()
.
.
.
---

Then I set up a testing extension in [internal] called '48' that would
trigger the queue and let me test:

[internal]
.
.
.
exten=>48,1,NoOp(Sending test call to testq)
same=>n,Queue(testq,trhc)
.
.
.
---

So, my test procedure was:

1. Phone 1007 on phone 3997 (1007 - see my original post - is my agent login
extension that calls AgentLogin on the dialing extension.) This logs me into
the queue 'testq' as Agent 3997.

2. Phone 1007 on phone 3998. This logs me into the queue 'testq' as Agent
3998.

3. Phone 48 from any other phone, which calls the Queue application on the
testq, which is defined in queues.conf as consisting of agents / phones 3997
and 3998.

4. Due to my queue being set to " strategy = leastrecent" on repeated calls
to extension 48, either 3997 or 3998 would pick up calls made.

This is exactly the functionality desired and I now have a working model of
Agent queues in Asterisk 13 using the new (to me) app_agent_pool interface
via AgentRequest instead of trying to use "chan_agent" which no longer
exists post Asterisk 12.

All that is now needed is to GOTO from my incoming extension to local,48,1
or simply do Queue(testq,trhc) from my incoming extension directly. If I
have logged in agents in the testq queue, the most least recently dialled
agent will get connected to the caller.

Maybe this helps someone.

Kind regards,

Stefan


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[asterisk-users] Asterisk 13.22.0 - No channel type registered for 'Agent' when queue rings

[Stefan Viljoen]

Hi All

With the below config, I just keep gettings this in the Asterisk 13.22.0
CLI:

WARNING[15872][C-0051]: channel.c:6343 ast_request: No channel type
registered for 'Agent'

whenever a caller gets sent to that agent queue with logged in agents
waiting for calls on Asterisk 13.

3997 and 3998 are my two agents. They can call each other and receive calls
from other extensions. They can phone out of my PBX, no problems.

My agents.conf:

---
[agents](!)
autologoff=15
wrapuptime=5000
custom_beep=beep

[3997](agents)
fullname=>AGENT 1

[3998](agents)
fullname=>AGENT 2
---

My queues.conf:

---
[testq]
musicclass = default
strategy = leastrecent
timeout = 10
retry = 5
autofill = yes

member => Agent/3997,0,3997,hint:3997@default,no
member => Agent/3998,0,3998,hint:3998@default,no
---

Also, setting ringinuse on (change member line above to have "yes" at the
end) has no effect except to make the error "No channel type..." appear
twice in the 13.22.0 CLI per external ring heard by caller, instead of once
per external ring if ringinuse is set to "no" in the two lines above.

Logging in an agent in extensions.conf:

---
exten=>1007,1,NoOp(Login agent from ${CALLERID(num)})
 same=>n,Set(MONITOR_OPTIONS=b)
 same=>n,UnpauseQueueMember(,Agent/${CALLERID(num)})
 same=>n,AgentLogin(${CALLERID(num)},s)
---

Queue show with agents logged in:

---
localhost*CLI> queue show
testq has 0 calls (max unlimited) in 'leastrecent' strategy (0s holdtime, 0s
talktime), W:0, C:0, A:16, SL:0.0% within 0s
   Members:
  3997 (Agent/3997 from hint:3997@default) (ringinuse disabled) (In use)
has taken no calls yet
  3998 (Agent/3998 from hint:3998@default) (ringinuse disabled) (In use)
has taken no calls yet
   No Callers
---

Then, if the queue is called from outside:

---
localhost*CLI> queue show
testq has 1 calls (max unlimited) in 'leastrecent' strategy (0s holdtime, 0s
talktime), W:0, C:0, A:16, SL:0.0% within 0s
   Members:
  3997 (Agent/3997 from hint:3997@default) (ringinuse disabled) (In use)
has taken no calls yet
  3998 (Agent/3998 from hint:3998@default) (ringinuse disabled) (In use)
has taken no calls yet
   Callers:
  1. SIP/3916-0055 (wait: 0:08, prio: 0)
---

3916 is never answered, and neither 3997 or 3998 receive the call, both
continuing with MOH indefinitely.

Constantly on the CLI this comes up, timed with each ring of 3916 (the
"outside" phone):

---
localhost*CLI> queue show
testq has 1 calls (max unlimited) in 'leastrecent' strategy (0s holdtime, 0s
talktime), W:0, C:0, A:16, SL:0.0% within 0s
   Members:
  3997 (Agent/3997 from hint:3997@default) (ringinuse disabled) (In use)
has taken no calls yet
  3998 (Agent/3998 from hint:3998@default) (ringinuse disabled) (In use)
has taken no calls yet
   Callers:
  1. SIP/3916-0055 (wait: 0:08, prio: 0)
  
localhost*CLI>
[Aug  2 10:37:39] WARNING[16925][C-0056]: channel.c:6343 ast_request: No
channel type registered for 'Agent'
[Aug  2 10:37:44] WARNING[16925][C-0056]: channel.c:6343 ast_request: No
channel type registered for 'Agent'
[Aug  2 10:37:49] WARNING[16925][C-0056]: channel.c:6343 ast_request: No
channel type registered for 'Agent'


What am I doing wrong?

The expectation is that either 3997 or 3998 will receive the incoming call
placed by SIP/3916. As of now, 3916 rings forever, and 3997 and 3998 just
play MOH forever, with the above error emitted with each ring heard on 3916
which is calling into the queue looking for someone to pick up.

Thanks!


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Re: [asterisk-users] SHELL() function Asterisk 13 - can only accept one paramter in string?

[Stefan Viljoen]

Hi Ludovic

 

Thank you very much for replying!

 

Ok, I’ll give up on the dialplan apps then in total and try out AGI.

 

Thanks again,

 



 

From: Ludovic Gasc  
Sent: Friday, 27 July 2018 13:37
To: viljo...@verishare.co.za; Asterisk Users Mailing List - Non-Commercial 
Discussion 
Subject: Re: [asterisk-users] SHELL() function Asterisk 13 - can only accept 
one paramter in string?

 

Hi Stefan,

 

You might use AGI instead of SHELL, it's better because you can interact with 
the call via AGI and you don't have a risk of remote execution you might have 
if you add arguments that come from outside, for example via a SIP header.

 

Regards.


--

Ludovic Gasc (GMLudo)

 

 

Le ven. 27 juil. 2018 à 09:37, Stefan Viljoen mailto:viljo...@verishare.co.za> > a écrit :

Hi all

This is a followup on my post "Asterisk 13 - system() dialplan app cannot call 
bash scripts" from yesterday

I've given up trying to use system() to call BASH scripts with parameters from 
Asterisk 13. 

Turned out under Asterisk 13.22.0 System() DOES work, but only if you do NOT 
attempt to pass any parameters to the called script.

This works, and reliably calls the script:

same=>n,System(/bin/bash /usr/src/bash/setData.sh)

However, the moment you do this:

same=>n,System(/bin/bash /usr/src/bash/setData.sh ${CHANNEL(accountcode)})

you get

WARNING[30982][C-0238] app_system.c: Unable to execute 
'/usr/src/bash/setData.sh'

I therefore tried to use SHELL() to do what I was trying to do with SYSTEM().

This also does not work, as SHELL() apparently can only ever parse ONE asterisk 
parameter in the string sent to it. All the rest are sent as blank.

If I do this:

same=>n,Set(nothing=${SHELL(/usr/src/verdi/bash/verdiLogIncomingCall.sh NA 
201807270838t49hgzs SIP/centra-out-6d9a IN SIP/3027-6db1 
SIP/3027-6db1 ApiLogIncomingCall.java 1)})

The script sees, on dialplan execution:

[root@acasterisk bash]# cat passed_param.txt
http://127.0.0.1/api/logIncomingCall?account_reference=NA 
<http://127.0.0.1/api/logIncomingCall?account_reference=NA_reference=201807270838t49hgzs_channel_id=SIP/centra-out-6d9a=IN_endpoint=SIP/3027-6db1_id=SIP/3027-6db1_from=ApiLogIncomingCall.java_sent_to_server=1>
 
_reference=201807270838t49hgzs_channel_id=SIP/centra-out-6d9a=IN_endpoint=SIP/3027-6db1_id=SIP/3027-6db1_from=ApiLogIncomingCall.java_sent_to_server=1
[root@acasterisk bash]#

E. g. all params present - because no variable references need to be parsed.

If I use this:

[macro-verdianswer]
exten=>s,1,NoOp(Entering Verdi answer macro - picked up by ${CHANNEL})
same=>n,NoOp(Source Channel: ${sourceChannel})
same=>n,NoOp(Answering Channel: ${CHANNEL})
same=>n,NoOp(Lodging CDR accountcode: ${curIncAccCode} as an incoming call from 
${numbersource} with VerDi and answered by ${CHANNEL}...)
same=>n,Set(CHANNEL(accountcode)=${curIncAccCode})
same=>n,Set(nothing=${SHELL(/usr/src/verdi/bash/verdiLogIncomingCall.sh NA 
${curIncAccCode} ${sourceChannel} IN ${CHANNEL} ${numbersource} 
ApiLogIncomingCall.java 1)})
same=>n,MacroExit()

giving this on exection:

-- SIP/3002-70c2 answered SIP/centra-out-70bf
-- Executing [s@macro-verdianswer:1] NoOp("SIP/3002-70c2", "Entering 
Verdi answer macro - picked up by SIP/3002-70c2") in new stack
-- Executing [s@macro-verdianswer:2] NoOp("SIP/3002-70c2", "Source 
Channel: SIP/centra-out-70bf") in new stack
-- Executing [s@macro-verdianswer:3] NoOp("SIP/3002-70c2", "Answering 
Channel: SIP/3002-70c2") in new stack
-- Executing [s@macro-verdianswer:4] NoOp("SIP/3002-70c2", "Lodging CDR 
accountcode: 2018072709061hrriyu
--  as an incoming call from 27182941000 with VerDi and answered by 
SIP/3002-70c2...") in new stack
-- Executing [s@macro-verdianswer:7] Set("SIP/3002-70c2", 
"nothing=Incoming call NOT stored. Contact software support.
-- ") in new stack

e. g. my variables ARE populated, and if I NoOp them, they have values.

In this situation, the script called via SHELL() sees:

[root@acasterisk bash]# cat passed_param.txt
http://127.0.0.1/api/logIncomingCall?account_reference=NA 
<http://127.0.0.1/api/logIncomingCall?account_reference=NA_reference=2018072709061hrriyu_channel_id==_endpoint=_id=_from=_sent_to_server=>
 
_reference=2018072709061hrriyu_channel_id==_endpoint=_id=_from=_sent_to_server=

E. g. SHELL() apparently only ever parses the FIRST Asterisk variable passed 
into it as a string, and never parses subsequent variable references.

Can anybody confirm or suggest a solution?

I desperately need to be able to execute external BASH scripts and pass 
multiple parameters to them, somehow. Nothing that worked in 1.8 for this works 
in 13...

Thanks!

Stefan


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Re: [asterisk-users] asterisk-users Digest, Vol 167, Issue 17

[Stefan Viljoen]

Hi Daniel

Thanks for the reply!

Yes, turns out it was all my fault, I had a line feed character (0x0a a.k.a 
printf("\n")) in one of the Asterisk channel variables passed via system() / 
shell() to my target script.

It seems 13.22.0 (I'm using the same version as you) reacts to a line feed in 
the parameter string by not processing or parsing the string any further one 
the line feed is encountered.

I removed the line feed (by using "echo -n" instead of just "echo" in a related 
BASH script that was providing input to Asterisk also via SHELL) and the 
problem disappeared and system() (and shell() itself ) started working 
correctly.

So, the moral is just keep the data clean that is fed through SYSTEM() and 
SHELL() from having any line-feeds in them.

On Asterisk 1.8 (my previous version) line-feeds in an Asterisk channel 
variable dereferenced in a SYSTEM() or SHELL() call did NOT stop parsing of the 
rest of the string - so apparently a relevant difference between the two 
versions there.

Kind regards,
---
Message: 2
Date: Fri, 27 Jul 2018 10:24:56 +0200
From: Administrator TOOTAI 
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] SHELL() function Asterisk 13 - can only
accept one paramter in string?
Message-ID: <008fecea-e2c0-9230-7cb4-cd1b6990c...@tootai.net>
Content-Type: text/plain; charset=utf-8; format=flowed

Le 27/07/2018 à 09:36, Stefan Viljoen a écrit :
> Hi all
> 
> This is a followup on my post "Asterisk 13 - system() dialplan app 
> cannot call bash scripts" from yesterday
> 
> I've given up trying to use system() to call BASH scripts with parameters 
> from Asterisk 13.
> 
> Turned out under Asterisk 13.22.0 System() DOES work, but only if you do NOT 
> attempt to pass any parameters to the called script.

[...]

*CLI> core show version
Asterisk 13.22.0 built by root @ pabx on a x86_64 running Linux on
2018-07-14 13:36:49 UTC

This works for us

same = n,system(/bin/echo "To: ${CALLED_CHANNEL}   From: ${ORI_CALL}" | 
/usr/bin/mail -s "TOOTAiAudio - Congestion Gateway ${CONGESTION_GW} used" 
${AUDIO_ADMIN})

--
Daniel


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Re: [asterisk-users] Asterisk 13 - system() dialplan app cannot call bash scripts

[Stefan Viljoen]

Hi Guys

Just feedback on this particular thread, this issue is SOLVED.

The reason why SYSTEM() and SHELL() was not working for me was that I was
passing a linefeed character (\n, hex 0x0a) in one of the channel variables
in Asterisk that was then parsed in the call to SYSTEM() and SHELL().

It appears that a big difference between 1.8 (where this code was developed)
and Asterisk 10 / 11 / 12 / 13 et al is that if SYSTEM() or SHELL()
encounters a linefeed in the parsed parameters passed in the string they are
called on, they stop parsing and evaluating any other Asterisk variables
passed in to the right of the parameter string given to SYSTEM() and
SHELL().

I eliminated the spurious \n character from a UUID generation BASH script
and SYSTEM() and SHELL() now appear to be working correctly, parsing and
evaluating all Asterisk channel variables passed into them as parameters,
and then calling the external BASH script 100% correctly with these
parameters.

So the issue was "bad data" on my side, and the apparent fact that Asterisk
13.22.0 is sensitive to \n in parameter strings passed to the SHELL()
function and SYSTEM() dialplan application, and stops processing the string
if a \n is encountered in a call to SHELL() or SYSTEM().

Hope this helps someone.

Regards

Stefan


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Re: [asterisk-users] SHELL() function Asterisk 13 - can only accept one paramter in string?

[Stefan Viljoen]

With thanks to Ludovic Gasc

---

Turns out there is nothing wrong with the SYSTEM() or SHELL() dialplan 
functions in Asterisk 13.22.0.

After several hours of painstaking debugging, the problem turned out to be a 
linefeed (\n character, 0x0a hex) in the first parameter passed from Asterisk 
to the bash script in the call to the SYSTEM() or SHELL() applications.

Ludovic's suggestion to try AGI revealed the issue - it has nothing to do with 
AGI or SYSTEM() but rather my data for one parameter passed through system 
containing a linefeed (\n)

E. g. I was calling this:

same=>n,System(/usr/src/verdi/bash/verdiLogIncomingCall.sh NA ${curIncAccCode} 
${sourceChannel} IN ${CHANNEL} ${numbersource} ApiLogIncomingCall.java 1)

but

${curIncAccCode}

appeared to contain a UUID I generate in a JAVA application.

In fact it contained

uuidtexthere\n

instead of just

uuidtexthere

E. g. it had an appended \n (hex 0a) linefeed character appended.

My script that generated this UUID looked like this:

---
#!/bin/bash
function jsonval {
temp=`echo $jsonResult | sed 's/\//\//g' | sed 's/[{}]//g' | awk -v 
k="text" '{n=split($0,a,","); for (i=1; i<=n; i++) print a[i]}' | sed 
's/\"\:\"/\|/g' | sed 's/[\,]/ /g' | sed 's/\"//g'`
echo ${temp##*|}
}

baseUrl=http://127.0.0.1/api/getVerdiUUID

jsonResult=$(curl --connect-timeout 16 --max-time 32 -s -X GET $baseUrl)

uuid=`jsonval`

echo $uuid
---

So all that I actually needed to fix was to change

echo $uuid

to

echo -n $uuid

and all my problems were solved, SYSTEM() started working perfectly everywhere 
in the dialplan as the script wasn't emitting a trailing \n anymore.

So the germane difference between Asterisk 1.8 and Asterisk  13 / 12 / 11 / 10 
in this case was that SYSTEM() and SHELL() in the later versions are sensitive 
to \n being passed in inside any of the Asterisk variables passed into those 
applications' parameter string.

Maybe this helps somebody else.

Regards

Stefan


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[asterisk-users] SHELL() function Asterisk 13 - can only accept one paramter in string?

[Stefan Viljoen]

Hi all

This is a followup on my post "Asterisk 13 - system() dialplan app cannot call 
bash scripts" from yesterday

I've given up trying to use system() to call BASH scripts with parameters from 
Asterisk 13. 

Turned out under Asterisk 13.22.0 System() DOES work, but only if you do NOT 
attempt to pass any parameters to the called script.

This works, and reliably calls the script:

same=>n,System(/bin/bash /usr/src/bash/setData.sh)

However, the moment you do this:

same=>n,System(/bin/bash /usr/src/bash/setData.sh ${CHANNEL(accountcode)})

you get

WARNING[30982][C-0238] app_system.c: Unable to execute 
'/usr/src/bash/setData.sh'

I therefore tried to use SHELL() to do what I was trying to do with SYSTEM().

This also does not work, as SHELL() apparently can only ever parse ONE asterisk 
parameter in the string sent to it. All the rest are sent as blank.

If I do this:

same=>n,Set(nothing=${SHELL(/usr/src/verdi/bash/verdiLogIncomingCall.sh NA 
201807270838t49hgzs SIP/centra-out-6d9a IN SIP/3027-6db1 
SIP/3027-6db1 ApiLogIncomingCall.java 1)})

The script sees, on dialplan execution:

[root@acasterisk bash]# cat passed_param.txt
http://127.0.0.1/api/logIncomingCall?account_reference=NA_reference=201807270838t49hgzs_channel_id=SIP/centra-out-6d9a=IN_endpoint=SIP/3027-6db1_id=SIP/3027-6db1_from=ApiLogIncomingCall.java_sent_to_server=1
[root@acasterisk bash]#

E. g. all params present - because no variable references need to be parsed.

If I use this:

[macro-verdianswer]
exten=>s,1,NoOp(Entering Verdi answer macro - picked up by ${CHANNEL})
same=>n,NoOp(Source Channel: ${sourceChannel})
same=>n,NoOp(Answering Channel: ${CHANNEL})
same=>n,NoOp(Lodging CDR accountcode: ${curIncAccCode} as an incoming call from 
${numbersource} with VerDi and answered by ${CHANNEL}...)
same=>n,Set(CHANNEL(accountcode)=${curIncAccCode})
same=>n,Set(nothing=${SHELL(/usr/src/verdi/bash/verdiLogIncomingCall.sh NA 
${curIncAccCode} ${sourceChannel} IN ${CHANNEL} ${numbersource} 
ApiLogIncomingCall.java 1)})
same=>n,MacroExit()

giving this on exection:

-- SIP/3002-70c2 answered SIP/centra-out-70bf
-- Executing [s@macro-verdianswer:1] NoOp("SIP/3002-70c2", "Entering 
Verdi answer macro - picked up by SIP/3002-70c2") in new stack
-- Executing [s@macro-verdianswer:2] NoOp("SIP/3002-70c2", "Source 
Channel: SIP/centra-out-70bf") in new stack
-- Executing [s@macro-verdianswer:3] NoOp("SIP/3002-70c2", "Answering 
Channel: SIP/3002-70c2") in new stack
-- Executing [s@macro-verdianswer:4] NoOp("SIP/3002-70c2", "Lodging CDR 
accountcode: 2018072709061hrriyu
--  as an incoming call from 27182941000 with VerDi and answered by 
SIP/3002-70c2...") in new stack
-- Executing [s@macro-verdianswer:7] Set("SIP/3002-70c2", 
"nothing=Incoming call NOT stored. Contact software support.
-- ") in new stack

e. g. my variables ARE populated, and if I NoOp them, they have values.

In this situation, the script called via SHELL() sees:

[root@acasterisk bash]# cat passed_param.txt
http://127.0.0.1/api/logIncomingCall?account_reference=NA_reference=2018072709061hrriyu_channel_id==_endpoint=_id=_from=_sent_to_server=

E. g. SHELL() apparently only ever parses the FIRST Asterisk variable passed 
into it as a string, and never parses subsequent variable references.

Can anybody confirm or suggest a solution?

I desperately need to be able to execute external BASH scripts and pass 
multiple parameters to them, somehow. Nothing that worked in 1.8 for this works 
in 13...

Thanks!

Stefan


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[asterisk-users] Edit: Asterisk 13.22.0 - "stat" dialplan function clears channel vars?

[Stefan Viljoen]

Sorry, I see I have submitted a testing version of the dialplan fragment.

Actual extension  is:

;listen to recording
exten=>,1,Answer()
exten=>,n,NoOp(Requesting File ${recfile})
exten=>,n,NoOp(Rec file set to ${recfile}) exten=>,n,NoOp(Alt file set 
to ${altfile}) exten=>,n,NoOp(Requesting Alt File ${altfile})
exten=>,n,Set(__numbertarget=)
exten=>,n,Set(haveFile=${STAT(e,${recfile}.gsm)})
exten=>,n,NoOp(Original File Exist ${haveFile}) 
exten=>,n,GotoIf($["${haveFile}" = "1"]?play) exten=>,n,NoOp(rec File 
not found using alt file)
exten=>,n,Set(recfile=${altFile})
exten=>,n(play),ControlPlayback(${recfile},2,6,4,8,5,9)
exten=>,n,hangup()

for my previous post.

Thanks!


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[asterisk-users] Asterisk 13.22.0 - "stat" dialplan function clears channel vars?

[Stefan Viljoen]

Hi Guys

I have the following dialplan code that I use to play back recordings, the 
filename being provided in an originate statement to the AMI (AJAM) interface:

Action: Originate
ActionID: test
Channel: SIP/3015
Exten: 
Context: local
Priority: 1
CallerID: 3015
Account: recordinglisten
ChannelID: abc
OtherChannelID: def
Variable: 
CallLimit=3600,recfile=/var/spool/asterisk/monitor/1807/25/2507180836591192526,altfile=/var/spool/asterisk/monitor/archive/1807/25/2507180836591192526
Async: true

Extension  is:

;listen to recording
exten=>,1,Answer()
exten=>,n,NoOp(Requesting File ${recfile})
exten=>,n,Set(${__recfile}=${recfile})
exten=>,n,Set(${__altfile}=${altfile})
exten=>,n,NoOp(Rec file set to ${recfile})
exten=>,n,NoOp(Alt file set to ${altfile})
exten=>,n,NoOp(Requesting Alt File ${altfile})
exten=>,n,Set(__numbertarget=)
exten=>,n,Set(haveFile=${STAT(e,${recfile}.gsm)})
exten=>,n,NoOp(Original File Exist ${haveFile})
exten=>,n,GotoIf($["${haveFile}" = "1"]?play)
exten=>,n,NoOp(rec File not found using alt file)
exten=>,n,Set(recfile=${altFile})
exten=>,n(play),ControlPlayback(${recfile},2,6,4,8,5,9)
exten=>,n,hangup()

However, if the above originate is done, when this line is executed in the 
dialplan Asterisk 13.22.0

exten=>,n,Set(haveFile=${STAT(e,${recfile}.gsm)})

it clears the variables recfile and altfile - if you NoOp them out below this 
line they are blank.

Why would the "STAT" dialplan function in 13.22.0 apparently clear all 
variables set on the channel, instead of checking if the given file exists (as 
it did in previous Asterisk versions) and leaving the channel variables alone...

Additionally, not only "recfile" which is passed as a STAT parameter is clear, 
"altfile" is -also- cleared to blank.

Can anybody assist?

Thank you

Stefan


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[asterisk-users] Recompiling Ast results in a binary with differing SHA256 sums?

[Stefan Viljoen]

Hi Guys

If I recompile Asterisk (on a Centos 7 test box, Asterisk 1.8.32.3) multiple
times in a row, e. g. 

make clean;configure;make menuselect;make

I note that the asterisk binary in the /main folder in the source tree, has
a different SHA256 hash each time I recompile Asterisk using the above
commands.

I do not change anything on the system or in the menuselect configs for each
run.

But each time the checksum for the "asterisk" binary is different.

Why is that? Shouldn't a freshly compiled binary off the same source, with
no changes in the Asterisk menuselect, with nothing changed on the rest of
the system, result each time in an IDENTICAL binary, down to the last byte?

Why am I getting a completely different "asterisk" ELF binary each time I
recompile asterisk, according to checksum?

Can someone shed light...

Thanks,

Stefan






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[asterisk-users] Big leap - 1.8 to 15.4.0

[Stefan Viljoen]

Hi all

We have to upgrade soon from prehistoric Asterisk 1.8.32.0 to Asterisk 15.x.x 
(whatever minversion is current at the time.)

We are quite heavily invested into 1.8.32.0 at about 17 sites locally and 
internationally and have a LOT of custom software running our sites via AMI 
from various applications.

What are big gotchas to watch out for? Especially regarding PJSip as I've been 
following posts and forums and it seems theres massive changes there, nevermind 
just overall changes in jumping years of dev ahead on the Asterisk version...

What should I watch out for so that our sites don't collapse completely on 
telecoms functionality when we go to 15 from 1.8? Any resources I can refer to? 
No upgrade guide I can find cover -this- wide a leap in versions.

I assume virtually every site's dialplan will have to be rewritten from 
scratch, which in itself is already a massive undertaking.

Anybody with some advice?

Thanks,

Stefan


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Re: [asterisk-users] Passing parameter to Queue-called macro

[Stefan Viljoen]

Hi Marie

That's why I asked about thread safety for global vars, cause it was
appearing that global variables was going to be the only way to send data
between the two instances - e. g. between the running call and the
queue-triggered macro.

So the inherited (_ prepend) channel variables are exactly what I needed, e.
g. a way to definitively isolate the unique ID for a call from the unique
IDs for all other concurrent queued calls when the relevant macro is
triggered by users picking up calls in a very busy queue. And I need to pass
a variable in from the calling context into the macro being called.

So all's good, at least my solution will now start to work in the way I
envisioned it to.

Thanks!
-Original Message-
From: Marie Fischer <ma...@vtl.ee> 
Sent: Friday, 11 May 2018 15:28
To: viljo...@verishare.co.za; Asterisk Users Mailing List - Non-Commercial
Discussion <asterisk-users@lists.digium.com>
Subject: Re: [asterisk-users] Passing parameter to Queue-called macro

Hi Stefan,

glad you got it solved.
Just to clarify, those are not global, but channel variables you are using -
so they should be visible only to their respective channel (and child
channels with inheritance).

Global variables are defined in a [globals] section in extensions.conf.
(https://wiki.asterisk.org/wiki/display/AST/Global+Variables+Basics)

--
BR,

marie

On 11.05.2018, at 9:01, Stefan Viljoen <viljo...@verishare.co.za> wrote:

> Hi Marie
> 
> Thanks!
> 
> I was just worried about thread safety if I had to use a global variable,
e.
> g. it might be set to a value by one call (since I'm using the same 
> global for every incoming call to transfer the accountcode gotten from 
> my HTTP endpoint to the same macro, and there can be several calls 
> simultaneously all inserting HTTP-sourced values at more or less the 
> same instant) and then another call is in such a state that it then 
> reads this call's data - and never reads its logical "own" data. The 
> classic concurrently accessed single variable issue.
> 
> Anyway, I've managed to solve this by declaring a variable in the main 
> dialplan as inheritable and storing my back-office relevant GUID in 
> there, then referencing that variable without the pre-prended _ in the
macro:
> 
> E. g.
> 
> [verdianswer]
> exten=>s,n,NoOp(Lodging CDR accountcode: ${curIncAccCode} as an 
> incoming call from ${numbersource} with VerDi and answered by 
> ${MEMBERINTERFACE}...) exten=>s,n,MacroExit
> 
> [telkomin]
> .
> .
> .
> same=>n,Set(curlResult=${SHELL(/usr/src/verdi/bash/verdiIncGetUUID.sh)
> })
> same=>n,Set(_curIncAccCode=${curlResult})
> same=>n,Queue(stefantest,trhc,,,60,,verdianswer)
> 
> The above works just fine for doing what I want to do, e. g. pass a 
> parameter from an Asterisk dialplan context into a queue-triggered 
> "agent just answered in the queue" Asterisk macro.
> 
> Thanks for the reply!
> 
> Kind regards
> 
> Stefan
> -Original Message-
> From: Marie Fischer <ma...@vtl.ee>
> Sent: Thursday, 10 May 2018 15:08
> To: viljo...@verishare.co.za; Asterisk Users Mailing List - 
> Non-Commercial Discussion <asterisk-users@lists.digium.com>
> Subject: Re: [asterisk-users] Passing parameter to Queue-called macro
> 
> Hi,
> 
> maybe I am overlooking something, but channel variables should be 
> thread safe, shouldn't they?
> 
> I am using the following (sorry, in ael):
> 
> macro dial-queue (number) {
>   Set(_ORIG_UNIQUEID=${UNIQUEID});
>   Queue(${number},rCt,,,${timeout},,set-dst-agent);
>   ..
> }
> 
> // the "context macro-..." things is an ael-specific workaround to get 
> transfer working (macro sets context to 
> app_queue_gosub_virtual_context) context macro-set-dst-agent {
>   s => {
>   Noop(${ORIG_UNIQUEID});
>   (${ORIG_UNIQUEID},${MEMBERNAME});
>   }
> }
> 
> macro add-current-call-agent (id,num) {
>   Set(ODBC_ADD_CURRENT_AGENT(${id},${num})=1);
>   return;
> }
> 
> --
> 
> marie
> 
> On 08.05.2018, at 16:16, Stefan Viljoen <viljo...@verishare.co.za> wrote:
> 
>> Hi all
>> 
>> I need to pass a parameter in a thread-safe manner to the Queue 
>> pickup macro. This is to know when (and who) picked up an incoming 
>> call to a queue and log that to my back-office system with a CURL to 
>> a HTTP
> endpoint.
>> 
>> However, the Queue application does not appear to allow passing of 
>> parameters to the called queue pickup macro.
>> 
>> E. g. non-working code is:
>> 
>> [queuetest]
>> timeout = 60
>> retry = 2
>> member=>SIP/testnum
>> 
>> [macro-verdian

Re: [asterisk-users] Passing parameter to Queue-called macro

[Stefan Viljoen]

Hi Marie

Thanks!

I was just worried about thread safety if I had to use a global variable, e.
g. it might be set to a value by one call (since I'm using the same global
for every incoming call to transfer the accountcode gotten from my HTTP
endpoint to the same macro, and there can be several calls simultaneously
all inserting HTTP-sourced values at more or less the same instant) and then
another call is in such a state that it then reads this call's data - and
never reads its logical "own" data. The classic concurrently accessed single
variable issue.

Anyway, I've managed to solve this by declaring a variable in the main
dialplan as inheritable and storing my back-office relevant GUID in there,
then referencing that variable without the pre-prended _ in the macro:

E. g.

[verdianswer]
exten=>s,n,NoOp(Lodging CDR accountcode: ${curIncAccCode} as an incoming
call from ${numbersource} with VerDi and answered by ${MEMBERINTERFACE}...)
exten=>s,n,MacroExit

[telkomin]
.
.
.
same=>n,Set(curlResult=${SHELL(/usr/src/verdi/bash/verdiIncGetUUID.sh)})
same=>n,Set(_curIncAccCode=${curlResult})
same=>n,Queue(stefantest,trhc,,,60,,verdianswer)

The above works just fine for doing what I want to do, e. g. pass a
parameter from an Asterisk dialplan context into a queue-triggered "agent
just answered in the queue" Asterisk macro.

Thanks for the reply!

Kind regards

Stefan
-Original Message-
From: Marie Fischer <ma...@vtl.ee> 
Sent: Thursday, 10 May 2018 15:08
To: viljo...@verishare.co.za; Asterisk Users Mailing List - Non-Commercial
Discussion <asterisk-users@lists.digium.com>
Subject: Re: [asterisk-users] Passing parameter to Queue-called macro

Hi,

maybe I am overlooking something, but channel variables should be thread
safe, shouldn't they?

I am using the following (sorry, in ael):

macro dial-queue (number) {
Set(_ORIG_UNIQUEID=${UNIQUEID});
Queue(${number},rCt,,,${timeout},,set-dst-agent);
..
}

// the "context macro-..." things is an ael-specific workaround to get
transfer working (macro sets context to app_queue_gosub_virtual_context)
context macro-set-dst-agent {
s => {
Noop(${ORIG_UNIQUEID});
(${ORIG_UNIQUEID},${MEMBERNAME});
}
}

macro add-current-call-agent (id,num) {
Set(ODBC_ADD_CURRENT_AGENT(${id},${num})=1);
    return;
}

-- 

marie

On 08.05.2018, at 16:16, Stefan Viljoen <viljo...@verishare.co.za> wrote:

> Hi all
> 
> I need to pass a parameter in a thread-safe manner to the Queue pickup 
> macro. This is to know when (and who) picked up an incoming call to a 
> queue and log that to my back-office system with a CURL to a HTTP
endpoint.
> 
> However, the Queue application does not appear to allow passing of 
> parameters to the called queue pickup macro.
> 
> E. g. non-working code is:
> 
> [queuetest]
> timeout = 60
> retry = 2
> member=>SIP/testnum
> 
> [macro-verdianswer]
> exten=>s,1,NoOp(Entering Verdi answer macro)
> exten=>s,n,NoOp(Value: ${ARG1})
> exten=>s,n,MacroExit
> 
> [incomingcontext]
> 
> exten=>tstqueue,1,NoOp(Incoming call for VerDi)
> same=>n,Set(curlResult=${SHELL(/usr/src/verdi/bash/verdiIncGetUUID.sh)
> })
> same=>n,Set(curlResultLength=${LEN(${curlResult})})
> same=>n,NoOp(Curl result for incoming call UUID from VerDi: 
> ${curlResult})
> same=>n,Set(CDR(accountcode)=${curlResult})
> same=>n,Set(curIncAccCode=${curlResult},g)
> same=>n,Macro(VCRECORD,stefantestEXT${CALLERID(num)}ACC${CDR(accountco
> de)},$
> {EXTEN})
> same=>n,Queue(queuetest,trhc,,,60,,verdianswer(${curIncAccCode}))
> same=>n,Hangup()
> 
> This results, when executed, in:
> 
> [May  8 15:14:50] WARNING[20921]: app_macro.c:309 _macro_exec: No such 
> context 'macro-verdianswer(2018050815141huzzu4
> ' for macro 'verdianswer(2018050815141huzzu4
> 
> How can one pass a paramter into the macro called by the Asterisk 
> queue application on queue pickup?
> 
> Alternatively, how can a global variable or ASTDB entry be made thread 
> safe to do the same?
> 
> Thank you
> 
> Stefan
> 
> 
> 
> 
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> 
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[asterisk-users] Passing parameter to Queue-called macro

[Stefan Viljoen]

Hi all

I need to pass a parameter in a thread-safe manner to the Queue pickup
macro. This is to know when (and who) picked up an incoming call to a queue
and log that to my back-office system with a CURL to a HTTP endpoint.

However, the Queue application does not appear to allow passing of
parameters to the called queue pickup macro.

E. g. non-working code is:

[queuetest]
timeout = 60
retry = 2
member=>SIP/testnum

[macro-verdianswer]
exten=>s,1,NoOp(Entering Verdi answer macro)
exten=>s,n,NoOp(Value: ${ARG1})
exten=>s,n,MacroExit

[incomingcontext]

exten=>tstqueue,1,NoOp(Incoming call for VerDi)
same=>n,Set(curlResult=${SHELL(/usr/src/verdi/bash/verdiIncGetUUID.sh)})
same=>n,Set(curlResultLength=${LEN(${curlResult})})
same=>n,NoOp(Curl result for incoming call UUID from VerDi: ${curlResult})
same=>n,Set(CDR(accountcode)=${curlResult})
same=>n,Set(curIncAccCode=${curlResult},g)
same=>n,Macro(VCRECORD,stefantestEXT${CALLERID(num)}ACC${CDR(accountcode)},$
{EXTEN})
same=>n,Queue(queuetest,trhc,,,60,,verdianswer(${curIncAccCode}))
same=>n,Hangup()

This results, when executed, in:

 [May  8 15:14:50] WARNING[20921]: app_macro.c:309 _macro_exec: No such
context 'macro-verdianswer(2018050815141huzzu4
' for macro 'verdianswer(2018050815141huzzu4

How can one pass a paramter into the macro called by the Asterisk queue
application on queue pickup?

Alternatively, how can a global variable or ASTDB entry be made thread safe
to do the same?

Thank you

Stefan




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Re: [asterisk-users] Call picked up from queue and transferred gets disconnected - about 0.01% of calls

[Stefan Viljoen]

Hi Khalil

 

Thanks for the reply. Yup, I’ve switched out the phone with another 
new-from-box Yealink, also moved that network endpoint to another RJ45 port on 
the switch...

 

Regards,

From: Khalil Khamlichi [mailto:khamlichi.kha...@gmail.com] 
Sent: Sunday, 11 February 2018 16:48
To: viljo...@verishare.co.za; Asterisk Users Mailing List - Non-Commercial 
Discussion <asterisk-users@lists.digium.com>
Subject: Re: [asterisk-users] Call picked up from queue and transferred gets 
disconnected - about 0.01% of calls

 

maybe extension 3082 has some sort of network issue or maybe hardphone issue. 
test on another phone / network plug.

 

On Feb 9, 2018 2:48 PM, "Stefan Viljoen" <viljo...@verishare.co.za 
<mailto:viljo...@verishare.co.za> > wrote:

Hi Guys

 

I have an issue where a call is picked up from a queue. The caller asks the 
person who answered to attended transfer to extension 3082 (for argument’s 
sake.)

 

3082 picks up the attended transfer and speaks with the outside caller picked 
up initially from the queue.

 

A few seconds after 3082 has started speaking to the outside caller

 

- 3082’s call goes dead in their handset.

 

- The outside caller goes back into the queue, hears queue MOH and gets 
answered by another person in the office as if they are dialing in all over 
again.

 

- 3082’s phone starts ringing again after they hang up in puzzlement and if 
they then pick up they speak to another person who is trying to make an 
OUTGOING call in their call center.

 

This is for a medium sized call center which (along with 17 other centers in 
the same country) run the same dialplan on Asterisk 1.8.32.3 - only happens at 
this location.

 

Literally 100 000+ calls are handled across these 18 centers every day, only 
about 10 or 20 at this one center (with carbon-copy dialplan and SIP phone 
hardware types - Yealink T-21Ps - as at every other branch) keeps disconnecting 
people picked up and transferred from the incoming queue from the person 
transferred to, and then connects them and the transferree to other phones - 
the caller as if he is phoning in AGAIN into the incoming queue, the 
transferred-to person to someone else who is trying to dial out once the 
transferred-to person has hung up after losing the incoming caller.

 

Anybody ever encountered something similar? The same dialplan on the same Ast 
version runs fine in 17 other locations, some with ten or twenty times more 
traffic and none of these issues.

 

No errors or strangeness apparent in the CLI, verbose log, DTMF log...

 

Thanks!

 



 


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[asterisk-users] Call picked up from queue and transferred gets disconnected - about 0.01% of calls

[Stefan Viljoen]

Hi Guys

 

I have an issue where a call is picked up from a queue. The caller asks the
person who answered to attended transfer to extension 3082 (for argument's
sake.)

 

3082 picks up the attended transfer and speaks with the outside caller
picked up initially from the queue.

 

A few seconds after 3082 has started speaking to the outside caller

 

- 3082's call goes dead in their handset.

 

- The outside caller goes back into the queue, hears queue MOH and gets
answered by another person in the office as if they are dialing in all over
again.

 

- 3082's phone starts ringing again after they hang up in puzzlement and if
they then pick up they speak to another person who is trying to make an
OUTGOING call in their call center.

 

This is for a medium sized call center which (along with 17 other centers in
the same country) run the same dialplan on Asterisk 1.8.32.3 - only happens
at this location.

 

Literally 100 000+ calls are handled across these 18 centers every day, only
about 10 or 20 at this one center (with carbon-copy dialplan and SIP phone
hardware types - Yealink T-21Ps - as at every other branch) keeps
disconnecting people picked up and transferred from the incoming queue from
the person transferred to, and then connects them and the transferree to
other phones - the caller as if he is phoning in AGAIN into the incoming
queue, the transferred-to person to someone else who is trying to dial out
once the transferred-to person has hung up after losing the incoming caller.

 

Anybody ever encountered something similar? The same dialplan on the same
Ast version runs fine in 17 other locations, some with ten or twenty times
more traffic and none of these issues.

 

No errors or strangeness apparent in the CLI, verbose log, DTMF log...

 

Thanks!

 



 

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[asterisk-users] Random, uncommanded blind transfers

[Stefan Viljoen]

Hi Guys

 

I've got a situation where an incoming call originated from a trunk provider
will ring in a call center and be answered by an agent.

 

Usually within 15 seconds of the incoming call being answered, it will
randomly blind-transfer to another extension in the same call center.

 

It is as if Asterisk is mis-reading some noise in-band as DTMF and doing the
transfer, or the caller is emitting DTMF to transfer.

 

An examination of CEL records for the relevant call shows the transfer
taking place, but no party (the caller or callee) tried to transfer. The
relevant queue application already has only small "t" in the parameter list
(e. g. only let the called user - e. g. my agent inside - transfer the
incoming call), thereby preventing the caller maybe emitting DTMF down the
line and transferring at their behest.

 

Yet the random, uncommanded blind transfers still take place, always withing
about 15 seconds after call initiation in answering an incoming SIP call.

 

We're using YeaLink SIP phones - T21P - got 17 branches, this only happens
at 3 of them at random times. Same hardware and Asterisk versions (1.8.32.3)
at all of them, same dialplans.

 

Operating over about several years, about 10 000 000 incoming calls handled
so far, and this only started happening now... dialplans are untouched for
months.

 

Any ideas, pointers, anybody encountered this before?

 

Thanks



 

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[asterisk-users] .gsm recordings corrupted when running Asterisk in virtual machine

[Stefan Viljoen]

Hi guys

Has anybody encountered a situation where .gsm recordings are corrupted if
the Asterisk instance is running in a virtual machine?

Specifically running Asterisk in MS Azure cloud VMs on Centos 7 inside the
hosted instance.

Switching to .wav files solves the problem, I can only guess that .gsm
encoding is very time sensitive or reliant on timer behaviour that cannot be
guaranteed in a virtual instance.

Is this assumption correct? Any pointers about how to make GSM encoding work
right on a VM?

Thanks!

Stefan


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[asterisk-users] CDR_TDS driver disappears - "does not provide a license key" on reload attempt

[Stefan Viljoen]

Hi All

I have an Asterisk 1.8.32.3 instance that will at random intervals stop
logging CDR data to MSSQL via FreeTDS.

On investigation I'll find that the FreeTDS module has been unloaded
somehow. It is not listed in cdr show status or show module like.

Trying 

module load cdr_tds

results in

module cdr_tds does not provide a license key

and the module is not loaded.

The only way to get things working again is to kill Asterisk, and do a make
install again.

On restart, FreeTDS is back.

Just replacing cdr_tds.so from a backup fails, Asterisk still won't load it.

You have to replace the asterisk binary -and- the cdr_tds.so file - their
sizes change as well, and a sha256sum when FreeTDS "does not provide a
license key" varies for both the main asterisk binary and the cdr_tds.so
module.

I have this on multiple sites very far apart, different hardware, asterisk
instances compiled at widely varying times.

Anybody encountered this before?

Thanks,








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[asterisk-users] asterisk.conf ignored?

[Stefan Viljoen]

Hi all

 

I'm trying to limit the maximum concurrent calls on my Asterisk to try and
mitigate another problem I posted about earlier.

 

I've edited

 

/etc/asterisk/asterisk.conf

 

And uncommented this line, and put a value of 60 in there:

 

maxcalls = 60

 

in an effort to limit my Asterisk to 60 simultaneous calls.

 

I did a 

 

core reload

 

in the CLI after doing that.

 

Any idea why my running instance totally ignores this setting? I still goes
right ahead and services unlimited numbers of simultaneous calls - we have
90 extensions or so and it will happily service 90 simultaneous calls in
spite of asterisk.conf clearly stating

 

maxcalls = 60

 

The "maxload" specification is also ignored, load can go anywhere the
Asterisk instance keeps taking more calls despite load exceeding, for
example

 

maxload = 10

 

in /etc/asterisk/asterisk.conf

 

What am I doing wrong that the asterisk binary is apparently ignoring
settings in /etc/asterisk/asterisk.conf?

 

Thanks

 

Stefan

 

 

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[asterisk-users] Asterisk sip_autodestruct messages - extensions locked

[Stefan Viljoen]

Hi guys

 

Does anybody have any opinion on what causes tens of thousands of these
messages per hour to pop up in the CLI:

 

[Jun 30 14:24:59] WARNING[2209]: chan_sip.c:4057 __sip_autodestruct:
Autodestruct on dialog '7e9597ae6ce95fef23374f4b380a9b70@192.168.0.1:5060'
with owner SIP/1148-0005bb2d in place (Method: BYE). Rescheduling
destruction for 1 ms

[Jun 30 14:25:01] WARNING[2209]: chan_sip.c:4057 __sip_autodestruct:
Autodestruct on dialog '6faefcc24547f1e774864ca87e3ff335@192.168.0.1:5060'
with owner SIP/1028-0005bb3f in place (Method: BYE). Rescheduling
destruction for 1 ms

[Jun 30 14:25:02] WARNING[2209]: chan_sip.c:4057 __sip_autodestruct:
Autodestruct on dialog '0d64480f4052a9e9054153552f1af7ba@192.168.0.1:5060'
with owner SIP/1412-0005bb5d in place (Method: BYE). Rescheduling
destruction for 1 ms

 

Symptoms are that an extension will dial, converse and hang up, but then be
unable to dial for up to four minutes after the initial call that extension
made - all the while thousands of the above messages scroll by in the CLI.
Along with then hundreds of "too many calls" warnings for each extension
which are limited to 1 call per extension at a time in sip.conf

 

I've googled intensively, no AGI is being run, hangup literally calls
Hangup(). I've already disabled and unloaded CDR-TDS, and ODBC CEL and CDR
logging to MySQL (in case db issues are causing a lock or something when
writing CDRs)

 

I see on big G that many many people have had this issue, but nowhere is
there any kind of concrete end result or answer to any of the questions.
This has started to happen at our site this Monday... it comes and goes
driven by no factor I can determine, except maybe workload. Server load
average when this is going on is 8 in top for a quad-Core i7 machine with
Centos 7 and 8GB of RAM, but the machine has hit 16 as load avg without this
happening. It also sometimes happens with the load average at 4 or less...

 

The particular server has been running undisturbed for about three years and
has handled tens of millions of calls, and only this Monday started
exhibiting this behaviour. The dialplan was last changed about three months
(and about 3 million calls) ago.

 

Any body got ANY advice or ideas where I can begin to diagnose this?

 

Thanks,

 

Stefan

 

 

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Re: [asterisk-users] ODBC locks warning in CLI - Asterisk 1.8.32.3

[Stefan Viljoen]

Hi all

Just to report back if it is of interest to anybody, I managed to solve
this.

Had nothing to do with Asterisk, rather with Percona 5.6 / MySQL which is
the database used via ODBC from Asterisk to store CELs and CDRs.

I added the following in the /etc/my.cnf file for the Percona instance I am
running:

innodb_lock_wait_timeout=120
transaction-isolation=READ-COMMITTED

and restarted the Percona instance.

This appears to have definitively removed the below warning from my Asterisk
instance's CLI.

Using READ-COMMITTED (instead of the default REPEATABLE-READ) transaction
isolation level apparently has optimised InnoDB table row locking (at least
in my use case, Percona / MySQL version and load) to such a degree that I no
longer see the warning below come up.

Everything is still reliably logging as far as I can determine, and all my
MySQL stored procedures which process my CDRs and CELs into useful form for
my dialing back-end are still working fine.

Just in case somebody can be helped by this.

>Hi all

>I get this warning in the Asterisk CLI about once every ten minutes or so:

>[Nov 23 14:47:36] WARNING[2544]: res_odbc.c:647
ast_odbc_prepare_and_execute: SQL Execute returned an error -1: HY000:
[MySQL][ODBC 5.1 Driver][mysqld-5.1.73]Deadlock found when trying to get
lock; try restarting transaction (105) [Nov >23 14:47:36] WARNING[2544]:
res_odbc.c:659 ast_odbc_prepare_and_execute: SQL Execute error -1! Verifying
connection to cdr [asterisk-cdr]...
>[Nov 23 14:47:36] WARNING[2544]: res_odbc.c:763 ast_odbc_sanity_check:
Connection is down attempting to reconnect...
>[Nov 23 14:47:36] NOTICE[2544]: res_odbc.c:1541 odbc_obj_connect:
Connecting cdr [Nov 23 14:47:36] NOTICE[2544]: res_odbc.c:1573
odbc_obj_connect: res_odbc: Connected to cdr [asterisk-cdr]

>Does this imply that I'm missing the ODBC CELs and / or CDRs that were
trying to write to MySQL over ODBC when the above occurred?

>Or will the ODBC module in Asterisk (or ODBC itself?) recover gracefully
and re-emit the CEL or CDR insert that hit the lock and were therefore NOT
written to MySQL?

>Thanks,

>Stefan






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[asterisk-users] Subject: Re: ODBC locks warning in CLI - Asterisk

[Stefan Viljoen]

Hi Jonathan

Thx for the reply.

Yup, have tried them, may just be our incompetence and inexperience with
Asterisk, but cannot get either of them to work right with our particular
setup.

Due to legacy issues we run very different dialplans at 17 different sites,
and some in-house custom software for Asterisk, and from testing 13 and 14
it appears each and every one of the sites will need custom rebuilding and
redesigning to work right with the newer versions. We also use different
hardware (DAHDI wise) at each site, different, -very- old PRI cards
manufactured by different companies, etc.

Plus, been monitoring the group closely for about two years now, the
problems and bugs apparent with 13 and 14 (some of which were solved,
granted) are spine chilling - if we run into some of the issues I've seen
around, our business will collapse.

PJSIP especially appears to be an absolutely horrendous nightmare -
extremely complex and difficult to configure for the type of situations we
have where 1.8.32.3 has been doing fine for years, over several tens of
millions of calls.

But just my two cents, I could be completely wrong - if I can put the below
issue to bed definitively,  the people I report to will probably stay on
standard 1.8.32.3 till it can no longer be compiled in a
whenever-contemporary Linux / libc / gcc environment...

>It might be worth pointing out that 1.8x was released 6 years ago, went
into security fix only over 2 years ago, and reached "end of life/no further
fixes" over a year ago.

>11.x went into "security fix only" last month - 13 and 14 are the current
versions - can you try with them?


On 23 November 2016 at 12:52, Stefan Viljoen <viljo...@verishare.co.za>
wrote:
> Hi all
>
> I get this warning in the Asterisk CLI about once every ten minutes or so:
>
> [Nov 23 14:47:36] WARNING[2544]: res_odbc.c:647
> ast_odbc_prepare_and_execute: SQL Execute returned an error -1: HY000:
> [MySQL][ODBC 5.1 Driver][mysqld-5.1.73]Deadlock found when trying to 
> get lock; try restarting transaction (105) [Nov 23 14:47:36] 
> WARNING[2544]: res_odbc.c:659
> ast_odbc_prepare_and_execute: SQL Execute error -1! Verifying 
> connection to cdr [asterisk-cdr]...
> [Nov 23 14:47:36] WARNING[2544]: res_odbc.c:763 ast_odbc_sanity_check:
> Connection is down attempting to reconnect...
> [Nov 23 14:47:36] NOTICE[2544]: res_odbc.c:1541 odbc_obj_connect: 
> Connecting cdr [Nov 23 14:47:36] NOTICE[2544]: res_odbc.c:1573 
> odbc_obj_connect: res_odbc:
> Connected to cdr [asterisk-cdr]
>
> Does this imply that I'm missing the ODBC CELs and / or CDRs that were 
> trying to write to MySQL over ODBC when the above occurred?
>
> Or will the ODBC module in Asterisk (or ODBC itself?) recover 
> gracefully and re-emit the CEL or CDR insert that hit the lock and 
> were therefore NOT written to MySQL?
>
> Thanks,
>
> Stefan
>
>
>
>
>
>
>
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Message: 3
Date: Wed, 23 Nov 2016 20:58:52 +0200
From: christopher kamutumwa <chriskamutu...@gmail.com>
To: asterisk-users <asterisk-users@lists.digium.com>
Subject: [asterisk-users] Asterisk Installation
Message-ID:

[asterisk-users] ODBC locks warning in CLI - Asterisk 1.8.32.3

[Stefan Viljoen]

Hi all

I get this warning in the Asterisk CLI about once every ten minutes or so:

[Nov 23 14:47:36] WARNING[2544]: res_odbc.c:647
ast_odbc_prepare_and_execute: SQL Execute returned an error -1: HY000:
[MySQL][ODBC 5.1 Driver][mysqld-5.1.73]Deadlock found when trying to get
lock; try restarting transaction (105)
[Nov 23 14:47:36] WARNING[2544]: res_odbc.c:659
ast_odbc_prepare_and_execute: SQL Execute error -1! Verifying connection to
cdr [asterisk-cdr]...
[Nov 23 14:47:36] WARNING[2544]: res_odbc.c:763 ast_odbc_sanity_check:
Connection is down attempting to reconnect...
[Nov 23 14:47:36] NOTICE[2544]: res_odbc.c:1541 odbc_obj_connect: Connecting
cdr
[Nov 23 14:47:36] NOTICE[2544]: res_odbc.c:1573 odbc_obj_connect: res_odbc:
Connected to cdr [asterisk-cdr]

Does this imply that I'm missing the ODBC CELs and / or CDRs that were
trying to write to MySQL over ODBC when the above occurred?

Or will the ODBC module in Asterisk (or ODBC itself?) recover gracefully and
re-emit the CEL or CDR insert that hit the lock and were therefore NOT
written to MySQL?

Thanks,

Stefan







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Re: [asterisk-users] Asterisk 11.24.1 garbled audio

[Stefan Viljoen]

Date: Tue, 15 Nov 2016 17:52:07 +0100
From: Olivier 
To: Asterisk Users Mailing List - Non-Commercial Discussion

Subject: Re: [asterisk-users] Asterisk 11.24.1 garbled audio
Message-ID:

Content-Type: text/plain; charset="utf-8"

>Hi,

>How can I double check which timer is currently is use in a running system
?
>core show settings doesn't tell anything, if I'm not mistaken.

How about

module show like timi

?

I use DAHDI timing, and this command on my system it shows:

module show like timi 
Module Description  Use
Count
res_timing_dahdi.soDAHDI Timing Interface   147
1 modules loaded

To get this in my modules.conf I specify (among other things):

load => res_timing_dahdi.so
noload => res_timing_pthread.so
noload => res_timing_timerfd.so

I'm using 1.8.32.3, but should apply to you as well.

Using DAHDI timing as I have had extensive problems under a heavily loaded
system with both pthread and timerfd timing.

Not sure if this still applies on your more recent Asterisk version though,
but anyway - this is how you should be able to see what timing is being used
on a running system.

Regards

Stefan


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Re: [asterisk-users] 1.8.32.3 - billsec field does not not increment after call answer - what triggers it? (Joshua Colp)

[Stefan Viljoen]

Hi Joshua

Thanks for the response.

Interesting that you mention that toll-free numbers can do this, this
problem trunk happens to receive calls from the national telecoms provider
here (Telkom SA) sourced from a toll-free number. The SIP trunk provider has
ported that toll free state telecoms company number for us to a local trunk
number with which we authenticate as a SIP peer. (Not sure if what Telkom
here in South Africa define as "toll free" on a technical level match what
you mention as "toll free" on an American / United States model.)

The thing is the number stays in the unanswered state (while in fact
answered and working perfectly) for the entire duration of the call, so then
it appears 200 OK is not received for the invite.

I therefore assume that the actual RTP negotiation process does NOT need the
200 OK for the invite, for the call itself is fine - it is just that
Asterisk never "realises", as regards the CDR, that the call was in fact
answered.

At least it is working and audio flows back and forth.

I'll see if I can come up with a SIP trace.

Thank you!

---
Stefan Viljoen wrote:



>
> Only this one trunk consistenly has this problem for all calls received
over
> it. The trunk provider is using sippy on their side.
>
> What setting / config option for the particular SIP "problem trunk" have
my
> trunk provider changed on their side to stop Asterisk from recognising
that
> a call has been answered when it comes in over that trunk?
>
> It appears some SIP traffic is not being sent by them (or not received by
my
> Asterisk) that indicates to it a call has been ANSWERED and that it must
> start the billsec timer?

I can't really speak for the provider but some numbers will stay in 
inband progress (unanswered) for a bit. Some toll-frees for example.

The specific SIP message that would show it as answered would be a 200 
OK to the INVITE we sent though. If you provided the SIP log then we 
could see.

Cheers,

-- 
Joshua Colp


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[asterisk-users] 1.8.32.3 - billsec field does not increment after call answer - what triggers it?

[Stefan Viljoen]

Hi Guys

I've got a strange problem - on my asterisk instance, when a call starts to
ring, I do 

core show channel  

and I get the usual output with the duration and billsec fields included.

For most of my calls, things are normal, e. g. duration field starts
incrementing as the SIP phone rings, and the moment it is answered / the
call goes offhook the duration timer continues running, and the billsec
timer starts up. Disposition goes from NO ANSWER to ANSWERED the moment
billsec starts incrementing.

However, for certain calls from a certain SIP trunk provided by a local
trunk provider, this never happens.

E. g. the call comes in on this "problem trunk" and duration timer starts
running - RTP starts and the call is totally normal, both parties have
crystal clear bi-directional audio and the call records correctly - but the
billsec timer never starts incrementing and forever remains at 0.
Disposition forever remains at NO ANSWER -  even though the call is in
progress and has been answered, and is working perfectly.

Other calls from other trunks provided by the same provider on the same
logical and physical Asterisk instance work correctly - if the call is
answered, it becomes ANSWERED in "core show channel" display, and the
billsec timer starts incrementing.

Only this one trunk consistenly has this problem for all calls received over
it. The trunk provider is using sippy on their side.

What setting / config option for the particular SIP "problem trunk" have my
trunk provider changed on their side to stop Asterisk from recognising that
a call has been answered when it comes in over that trunk?

It appears some SIP traffic is not being sent by them (or not received by my
Asterisk) that indicates to it a call has been ANSWERED and that it must
start the billsec timer?

Thanks!

Stefan



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[asterisk-users] Proper way to start Asterisk on CentOS 7? (Carlos Chavez)

[Stefan Viljoen]

Hi Carlos

I have experienced something similar starting Asterisk 1.8.32.3 on Centos 7
on commodity / whitebox hardware.

The problem was that Asterisk was starting "too quickly" in the systemd
startup sequence, before the required services it needs to run were up and
ready.

I eventually came up with this systemd script to start it, and this now
starts Asterisk on all our our new Centos 7-using deployments.

Maybe you can try it out...? This starts our Asterisk 1.8 instances without
problems on Centos 7 on boot:

--

[Unit]
Description=Asterisk
After=network.target
After=network-online.target
After=startup.service
After=mysql.service
Wants=mysql.service

[Service]
Type=idle
User=root
Group=root
ExecStart=/usr/sbin/asterisk -f
ExecStop=/usr/sbin/asterisk -rx 'core stop now'
ExecReload=/usr/sbin/asterisk -rx 'core reload'
TimeoutSec=300

LimitCORE=infinity
LimitNOFILE=8096
Restart=always
RestartSec=4

[Install]
WantedBy=multi-user.target

--

I place this in /etc/systemd/system as "asterisk.service" and then install
it via

systemctl enable asterisk.service

Hope this helps.

Stefan
-
Message: 1
Date: Thu, 5 May 2016 12:07:37 -0500
From: Carlos Chavez 
To: "asterisk-users@lists.digium.com"

Subject: [asterisk-users] Proper way to start Asterisk on CentOS 7?
Message-ID: <572b7dd9.4090...@telecomabmex.com>
Content-Type: text/plain; charset=utf-8; format=flowed

 I am having a strange problem with Asterisk 13 on a CentOS 7 plataform.
I have several servers running on this configuration but a particular
installation on a Dell PowerEdge 220 server is the one giving me the most
problems.  All installations are automated via a script so there is no
difference but the hardware.  All dependencies are installed and there are
no errors during installation.  We use realtime mysql for our
configurations.

 The problem comes when Asterisk is started from init.d/asterisk.  
Asterisk loads but phones will not register and the Asterisk CLI will not
respond to commands.  Something like "pjsip show endpoints" shows nothing
and sometimes the CLI hangs. Asterisk itself is not crashing and if I open
another terminal I can get into the CLI even though the other terminal is
not responding. If I start Asterisk manually from the command line
everything works as expected, phones register and calls flow.

 I tried a systemd startup script someone published but I get the same
result as the initd script.  I wonder why I am only getting this problem on
the Dell 220 server but not on other servers (even a Dell T20 server does
not have the same problem).  My office server is an HP ML110 and has been
working with Asterisk 13 and CentOS 7 for over 6 months now with no issues.
It uses the init.d script that comes with Asterisk and it has never hanged.

 Anyone know if the PowerEdge 220 server has special issues with Linux
or Asterisk in particular?  How do you recommend starting Asterisk?

--
Telecomunicaciones Abiertas de M?xico S.A. de C.V.
Carlos Ch?vez
+52 (55)9116-91161


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Re: [asterisk-users] Best timing source? (Carlos Chavez)

[Stefan Viljoen]

Hi Carlos

We had similar issues with Asterisk 1.8.11.0 and Asterisk 1.8.32.2 with
timerfd and pthread timing.

The server instance would run for 20 or 30 minutes and then producde legions
of error lines about timerfd or pthread, whichever we were using.

The symptoms were voice quality issues - jitter and packet drops spiked
alarmingly during periods where these errors about timterFD and pthread were
being emitted.

This was also on Centos 7.

The solution in our case was two fold.

- We switched from Asterisk 1.8.11.0 to Asterisk 1.8.32.2

This was step one - since when we used timerfd or pthreads with 1.8.32.3
(even though it was much newer than 1.8.11.0) we STILL had the problem.

- Once on Asterisk 1.8.32.2, we also switched to DAHDI timing

This was step two - when we did this as well, the problems with voice
quality and the tens of thousands of error messages disappeared from the
console and it has been running fine for months since, done about 800 000+
calls so far.

The interesting thing is we have no DAHDI hardware in that system, or any of
the other systems (we run 17 Asterisk instances on Centos 7 real-metal
machines countrywide) where we use DAHDI timing. AFAIK the DAHDI timer is
then emulated in software in the res_timing_dahdi.so module.

So it is not that it is "better" but the upgarding to 1.8.32.2 and using
DAHDI timing definitely solved our voice quality problems along with making
the constantly repeated timing error messages we were seeing in the console
go away.

Kind regards,

On 4/5/16 3:17 PM, Joshua Colp wrote:
> Carlos Chavez wrote:
>> I am currently having a voice quality problem with one of our 
>> Asterisk servers. We have checked the network and we have found no 
>> problems that could cause the voice to sound cracked and with small 
>> interruptions. I am looking at the timing source for Asterisk and it 
>> is currently using timerfd even though we have an E1 card installed. 
>> Is timerfd better than dahdi? Any recommendations to test if timing 
>> may be a problem for voice quality and DTMF?
>
> What is the scenario and the channels involved? Timing is only used 
> for things such as playback, music on hold, and ConfBridge. If it's 
> strictly a two party call then Asterisk forwards media as received.
>

I am starting to think that the problem may be in the server
hardware itself.  We are using a Dell R220 with 8gb of ram and 2 hard disks
in a Raid 1 configuration (Linux Raid).  We are using CentOS 7.  We had to
remove the raid card from the server to install an E1 card (the raid card
was Windows only so no loss there really).  Internally everything sounds
good (from E1 to a conference or music) but once you hit a network interface
we start getting pops and drops.

Anyone with this server and Asterisk ever had issues like these?



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Re: [asterisk-users] CEL entries over ODBC several hours late (Vinicius Fontes)

[Stefan Viljoen]

Hi Vinicius

Thanks for replying.

>Sorry for the probably obvious question, but it's better to cover all
bases.

>The DBMS is running on the same box as Asterisk is? If that's the case then
maybe the DBMS is using too much CPU and starving Asterisk?

I don't think so - I think I have a locking issue in the DB specifically
with my stored procedure -and- Asterisk trying to access the CEL table
simultaneously.

Load averages appear normal even if the DB is running in the background. The
autodestruct problem occurs whenever the CEL table (InnoDB, so should NOT be
locked for inserts while being queried in MySQL "nolock" equivalent, but it
IS) is busy being read by the SP and Asterisk tries to write into it.

This then back-flows to the Asterisk and delays the H extension execution to
such a degree that handsets get locked tight until the CEL table lock held
by the SP is released and Asterisk can write to it again over ODBC.

I understand that CEL logging and call hangup should be an atomic operation
(billing related) from there though the question about why it apparently
(from observed effects) is a synchronous operation - but considering it must
be atomic, it does make sense I suppose to write CELs synchronously with
call teardown.

It just requires that you keep the CEL table in MariaDB (or whatever) in an
"un-locked" state at all times - which makes it pretty hard since to have
any practical use out of the CEL table you should be able to query it
without locking it - which InnoDB in MariaDB is supposd to provide, but
clearly does not in my type of setup here.

Anyway, thanks for the reply.

Regards

Stefan


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Re: [asterisk-users] CEL entries over ODBC several hours late (Matthew Jordan)

[Stefan Viljoen]

Hi Matthew

Thank you very much for the reply.

I must have something seriously wrong somewhere else then - I retested now
and the "apparent" effect is as  I describe but your info definitely
contradicts that. But you're obviously correct.

One more question - I've noted that if I run a combination of queries in the
CEL backing DB (MariaDB) and the CEL table is locked, this severely affects
the Asterisk instance - thousands of occurrences of

chan_sip.c:4057 __sip_autodestruct: Autodestruct on dialog
'6a9f5d3543b619655e07c81437373a32@172.17.12.3:5060' with owner
SIP/3034-000207c8 in place (Method: BYE). Rescheduling destruction for 1
ms

appear in the CLI and users complain that if they hang up then they cannot
make another call on the same SIP handset for several minutes.

This is obviously because the dialplan gets delayed in the H extension, and
cannot write to the CEL table, waiting for the MariaDB instance to clear the
locks so it can write again. The above apparently comes from a watchdog
process that watches how "fast" the H extension is and if it takes "too
long" it forces the channels closed.

Is this assumption correct?

Addtionally, it seems that the writing of CEL in Asterisk is NOT async? E.g.
it appears the thread that was running the conversation ALSO does the CEL
writing / pushing to the CEL core as you describe in a synchronous manner.

For 1.8, is this correct that CELs are synchronous, and do newer Asterisk
versions do it async?

E. g. my point being if there are major DB issues, it is quite a bit
kryptonite to have that "spill back" into Asterisk and start blocking off
users from calling out - wouldn't it be much better to simply have failed
CEL writes just die in a distal thread instead of the main call thread for
the channels running on that handset.

Or am I completely misunderstanding things?

Anyway, thanks for the reply. :)

Kind regards,

> Hi guys
>
> I'm running 1.8.32.3 with CEL logging over ODBC to MariaDB 5.5.41 on the
> same Centos 7 machine.
>
> I've noticed that the CDR entries made are all in-time, e. g. the call
will
> take place and the CDR entry is immediately written into the CDR table in
> the MariaDB database.
>
> However, CEL events for that CDR (which I need to process for a realtime
> display feature in my dialer software) are always several hours after the
> fact. E. g. I will make a call at 09:30, see the call immediate pop up in
> the MariaDB CDR table, but only at about 15:15 that afternoon will I see
> that call's CEL events come into the CEL table, from Asterisk I have
> examined the `show processlist` in MariaDB exhaustively to establish this
> fact.
>
> The system doesn't appear loaded, load average is about 1.1 - it's a
> quad-coare HT Intel Xeon E3-1225 with 8GB of DRAM running on an SSD for
> main
> storage.
>
> The system processes about 30 000 calls every 8 hour day, and services 90
> SIP phones.
>
> I can stop and restart the MariaDB instance for several minutes, when I
> restart it it immediately picks up on the "slow" CELs from where it was
> interrupted - more evidence that Asterisk is very slowly streaming the
CELs
> through. I thought MariaDB was the bottleneck, but apparently not?
>
> If I make test inserts from a script into the CEL table, all of them
> complete so quickly a time indication does not even register for the query
> in MariaDB. Simple test queries on the CEL table are also instant, not
even
> counting in the internal MariaDB query duration timer.
>
> Can anybody explain why this is that the CELs asterisk emits over ODBC are
> so delayed? Are CELs intended NOT to be realtime?
>
> So, logically, Asterisk appears to be caching CELs to the tune of hundreds
> of thousands of them at any given time - meaning if it is stopped (either
> killed, or core stop gracefully'ed, or just "core stop now")  potentially
> hundreds of thousands of CELs will just evaporate irretrivably.
>
> What can I do to mitigate this extremely slow populating of CELs over
ODBC?
>
>
Asterisk does not buffer CEL entries. If anything, it pushes the entries
out to ODBC much more aggressively than what you would get with CDRs.

An event is generated in Asterisk that corresponds to the CEL entry. That
entry is pushed over a message bus (the 'event' message bus in 1.8 - 11;
'stasis' in 12+) and is picked up by the CEL core. The events are
immediately sent to the registered backends, who also immediately write it
out to the backend they support. In the case of ODBC, this immediately does
an INSERT into the appropriate table.

In Asterisk 1.8, you can look for a verbose level 11 message that will show
when this occurs:

ast_verb(11, "[%s]\n", ast_str_buffer(sql));

In later versions, this was turned into a debug level 3 message (as
anything over a verbose 5/debug 5 was cleaned up).

If you see that message, then that will tell you when Asterisk *believes*
it has written the CEL entry. If that doesn't show up in the database, then
it is either in the ODBC 

[asterisk-users] CEL entries over ODBC several hours late

[Stefan Viljoen]

Hi guys

I'm running 1.8.32.3 with CEL logging over ODBC to MariaDB 5.5.41 on the
same Centos 7 machine.

I've noticed that the CDR entries made are all in-time, e. g. the call will
take place and the CDR entry is immediately written into the CDR table in
the MariaDB database.

However, CEL events for that CDR (which I need to process for a realtime
display feature in my dialer software) are always several hours after the
fact. E. g. I will make a call at 09:30, see the call immediate pop up in
the MariaDB CDR table, but only at about 15:15 that afternoon will I see
that call's CEL events come into the CEL table, from Asterisk I have
examined the `show processlist` in MariaDB exhaustively to establish this
fact.

The system doesn't appear loaded, load average is about 1.1 - it's a
quad-coare HT Intel Xeon E3-1225 with 8GB of DRAM running on an SSD for main
storage.

The system processes about 30 000 calls every 8 hour day, and services 90
SIP phones.

I can stop and restart the MariaDB instance for several minutes, when I
restart it it immediately picks up on the "slow" CELs from where it was
interrupted - more evidence that Asterisk is very slowly streaming the CELs
through. I thought MariaDB was the bottleneck, but apparently not?

If I make test inserts from a script into the CEL table, all of them
complete so quickly a time indication does not even register for the query
in MariaDB. Simple test queries on the CEL table are also instant, not even
counting in the internal MariaDB query duration timer.

Can anybody explain why this is that the CELs asterisk emits over ODBC are
so delayed? Are CELs intended NOT to be realtime?

So, logically, Asterisk appears to be caching CELs to the tune of hundreds
of thousands of them at any given time - meaning if it is stopped (either
killed, or core stop gracefully'ed, or just "core stop now")  potentially
hundreds of thousands of CELs will just evaporate irretrivably.

What can I do to mitigate this extremely slow populating of CELs over ODBC?

Thanks


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[asterisk-users] 1.8.32.3 - no timing indicated, tens of thousands of __sip_autodestruct error messages

[Stefan Viljoen]

Hi list

Just to let everybody know I think I've got to the bottom of the above
problem / error.

Turns out that the issues described in my previous post were caused by
problems in an MSSQL database that the Asterisk 1.8.32.3 instance was
writing to...!

Once I dropped the FreeTDS driver (while trying to diagnose what was going
on) the Asterisk instance immediately started performing normally and the
__sip_autodestruct error messages disappeared.

Channels linked to calls that were hung up closed normally once again and
everything was golden.

The problem on MSSQL turned out to be that some of the indices on the table
to which the Asterisk instance was writing required rebuilding as the table
had grown immensely since the Asterisk instance was created. The issue was
that an insert that would have normally taken about 500 milliseconds was now
taking up to 30 seconds to a minute. Therefore as a call is finished
Asterisk attempted to write the call into the CDR database but was extremely
delayed due to slow database performance. The __sip_autodestruct error
messages apparently is a watchdog thread / process that then detects that
channels that should be closed and gone have not been closed / disposed - it
then posts a warning and then later tries to close the channels involved,
which are still hanging around after the call they are linked to has
finished.

This had the effect that users could only make one call, then had to wait
until the channels disappeared (sometimes up to a minute later as the DB
managed to finish inserting that call's details in to the table with the
index problems) before they could make a second call, while the above
watchdog process complained loudly about channels that were hanging around
that should not be as Asterisk waited for MSSQL.

So the fix was simply to rebuild the indexes on the relevant tabe in MSSQL -
this definitively solved the problem and tens of thousands of calls have now
been made after the reindex in MSSQL and everything is fine.

This is interesting as it seems that Asterisk does NOT use FreeTDS
asynchronously, but synchronously as calls progress...

Anyway, hope this helps someone who runs into something similar - the
databases you touch from the CDR subsystem in Asterisk must be -FAST- and
able to be inserted into in milliseconds. Apparently anything taking a
second or more in a FreeTDS accessed DB linked to Asterisk can start causing
problems in Asterisk itself.

Regards

Stefan





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[asterisk-users] 1.8.32.3 - no timing indicated, tens of thousands of __sip_autodestruct error messages

[Stefan Viljoen]

Hi all

I'm running a 1.8.32.3 instance of Asterisk on Centos 7.

About 80 SIP phones connected.

I'm getting tens of thousands of error messages in the CLI, and callers are
having extreme difficulty dialing, they can dial once and then if they try
to dial again their extension is shown as busy.

The server has been running for a year and a half, until this Wedenesday.
When I encountered this problem I then replaced the server with a new
physical machine with the same version of Asterisk that has been working for
the past year.

E. g. the hardware on the server is completely new, still having the exact
same problem as with the older server.

This is what is appearing in the CLI:

[Oct 30 11:02:16] WARNING[16831]: chan_sip.c:4057 __sip_autodestruct:
Autodestruct on dialog '702825cb5d9de4fc6a29675f158f000f@192.168.1.2:5060'
with owner SIP/centra-out-0140 in place (Method: BYE). Rescheduling
destruction for 1 ms
[Oct 30 11:02:24] WARNING[16831]: chan_sip.c:4057 __sip_autodestruct:
Autodestruct on dialog '18eea85e1c27f2fd533586452b41f9a7@172.7.1.3:5060'
with owner SIP/3088-01b0 in place (Method: BYE). Rescheduling
destruction for 1 ms
[Oct 30 11:02:37] WARNING[16831]: chan_sip.c:4057 __sip_autodestruct:
Autodestruct on dialog '4004649c0c903ee1676f5b5c2d47e0eb@172.7.1.3:5060'
with owner SIP/3055-0137 in place (Method: BYE). Rescheduling
destruction for 1 ms
[Oct 30 11:02:39] WARNING[16831]: chan_sip.c:4057 __sip_autodestruct:
Autodestruct on dialog '3dbbe75d30e0c7bd72ddf69716be8806@192.168.1.2:5060'
with owner SIP/centra-out-01b1 in place (Method: BYE). Rescheduling
destruction for 1 ms
[Oct 30 11:02:40] WARNING[16831]: chan_sip.c:4057 __sip_autodestruct:
Autodestruct on dialog '2ff26c957b75c41f11b6523c7c53a134@192.168.1.2:5060'
with owner SIP/centra-out-00cb in place (Method: BYE). Rescheduling
destruction for 1 ms
[Oct 30 11:02:40] WARNING[16831]: chan_sip.c:4057 __sip_autodestruct:
Autodestruct on dialog '1627933358aec1323968a60564e76d1a@172.7.1.3:5060'
with owner SIP/3065-00ca in place (Method: BYE). Rescheduling
destruction for 1 ms
[Oct 30 11:02:43] WARNING[16831]: chan_sip.c:4057 __sip_autodestruct:
Autodestruct on dialog '41a753fb01f16d5f0d366a314495685b@192.168.1.2:5060'
with owner SIP/centra-out-0138 in place (Method: BYE). Rescheduling
destruction for 1 ms
[Oct 30 11:02:46] WARNING[16831]: chan_sip.c:4057 __sip_autodestruct:
Autodestruct on dialog '1d90ffab7a7241d9236bd3816a9889fd@172.7.1.3:5060'
with owner SIP/3054-013f in place (Method: BYE). Rescheduling
destruction for 1 ms
[Oct 30 11:02:48] WARNING[16831]: chan_sip.c:4057 __sip_autodestruct:
Autodestruct on dialog '702825cb5d9de4fc6a29675f158f000f@192.168.1.2:5060'
with owner SIP/centra-out-0140 in place (Method: BYE). Rescheduling
destruction for 1 ms
[Oct 30 11:02:56] WARNING[16831]: chan_sip.c:4057 __sip_autodestruct:
Autodestruct on dialog '18eea85e1c27f2fd533586452b41f9a7@172.7.1.3:5060'
with owner SIP/3088-01b0 in place (Method: BYE). Rescheduling
destruction for 1 ms
[Oct 30 11:02:59] WARNING[16831]: chan_sip.c:4057 __sip_autodestruct:
Autodestruct on dialog '47c4a97d2d1a85e95c85a5b42ba9c27f@172.7.1.3:5060'
with owner SIP/3008-01d2 in place (Method: BYE). Rescheduling
destruction for 1 ms
[Oct 30 11:03:00] WARNING[16831]: chan_sip.c:4057 __sip_autodestruct:
Autodestruct on dialog '32fe9add31a3305674bca7e108ff9746@172.7.1.3:5060'
with owner SIP/3014-01d7 in place (Method: BYE). Rescheduling
destruction for 1 ms
[Oct 30 11:03:03] WARNING[16831]: chan_sip.c:4057 __sip_autodestruct:
Autodestruct on dialog '72e37d2a700418d12c9e5a2e645f5993@172.7.1.3:5060'
with owner SIP/3061-01d5 in place (Method: BYE). Rescheduling
destruction for 1 ms
[Oct 30 11:03:09] WARNING[16831]: chan_sip.c:4057 __sip_autodestruct:
Autodestruct on dialog '4004649c0c903ee1676f5b5c2d47e0eb@172.7.1.3:5060'
with owner SIP/3055-0137 in place (Method: BYE). Rescheduling
destruction for 1 ms
[Oct 30 11:03:11] WARNING[16831]: chan_sip.c:4057 __sip_autodestruct:
Autodestruct on dialog '3dbbe75d30e0c7bd72ddf69716be8806@192.168.1.2:5060'
with owner SIP/centra-out-01b1 in place (Method: BYE). Rescheduling
destruction for 1 ms
[Oct 30 11:03:12] WARNING[16831]: chan_sip.c:4057 __sip_autodestruct:
Autodestruct on dialog '2ff26c957b75c41f11b6523c7c53a134@192.168.1.2:5060'
with owner SIP/centra-out-00cb in place (Method: BYE). Rescheduling
destruction for 1 ms

If I do

asterisk*CLI>module show like timing 

I get

module show like timing
Module Description  Use
Count
res_timing_dahdi.soDAHDI Timing Interface   0
asterisk*CLI>

My modules.conf specifies that only dahdi timing.

I have never seen any of my other 17 Asterisk servers at other sites showing
NO use of ANY timing interface...

What does these __sip_autodestruct messages mean? How can I fix this
problem?

Calls in 

Re: [asterisk-users] 786 000 files limit Centos 7 - Asterisk (Stefan Viljoen)

[Stefan Viljoen]

Hi List

Regarding this Asterisk instance as discussed previously (Asterisk 1.8.11.0)
that was consuming enormous amounts of file descriptors (100 000+ for about
50 simultaneous calls) it appears I have managed to solve my problem by
upgrading the 1.8.11.0 Asterisk instance to an 1.8.32.3 Asterisk instance.

Also, the file descriptors apparently leaking were paired with timer
problems in 1.8.11.0 whenever I went above about 50 concurrent calls on the
box while running on 1.8.11.0.

The thing is in our setup we have about 15 instances of 1.8.11.0 at the
various branches of the company, all running 1.8.11.0, BUT at none of these
sites do we ever exceed 40 simultaneous calls.

The defining factor was (in our case, with our dialplan) to run 1.8.11.0 and
try to run 50+ concurrent calls.

What would happen was that thousands of these messages would come up in the
CLI:

[Aug 13 09:41:38] ERROR[25193]: res_timing_dahdi.c:89 dahdi_timer_set_rate:
Failed to configure DAHDI timing fd for 0 sample timer ticks

when we reached or exceeded 50 calls.

The same happened whether pthread timing or kernel timerfd timing was used.

Several other weird errors would manifest in the CLI, to whit:

---
format_gsm.c:102 gsm_write: Bad write (32/33): Destination address required

[Aug 12 12:23:33] WARNING[29436]: channel.c:1474 __ast_queue_frame:
Exceptionally long voice queue length queuing to Local/number@local-3E1C;1

WARNING[8210]: res_rtp_asterisk.c:1773 ast_rtcp_read: RTCP Read error: Bad
file descriptor.  Hanging up.

[Aug 12 09:56:55] WARNING[29931]: file.c:198 ast_writestream: Translated
frame write failed
---

when we were spamming the timer errors. Practical effects were dropped
calls, calls with bad quality / what sounds like severe jitter, and
mixmonitor recording files that were not written or corrupt.

The solution (so far, still checking) was simply to upgrade to 1.8.32.0 and
most of our problems disappeared, for us in our setup, with our dialplans.
The upgrade was painless, since we stayed in the 1.8 range, we did not have
to modify any of our config files or dialplans.

Maybe this can assist someone else struggling with older 1.8 series timer
issues.

Regards

Stefan


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Re: [asterisk-users] One way audio - doesn't seem to be NAT issue

[Stefan Viljoen]

Hi D'Arcy

 that the server IP for RTP as specified in the initial SIP is correct?

Both the server and client are outside of NAT so I don't know what this
might mean.  They both have public IPs.

This was a problem we had when the RTP server negotiated in SIP with our
VOIP ITSP on one side of the connection, differed from the IP we were
expecting on that side of the connection and was blocked in our firewall.

Once we perused the SIP traffic we noted this and added the extra IP to the
firewall for RTP traffic.

 We had slightly different parameters, e. g. that we would have no RTP 
 at all, but a call that did connect to total silence, dialed from 
 either side.

Was NAT involved?

Yes, NAT was being done at both ends, but it turned out that NATing was not
the problem.

 Also check what RTP port ranges are being used - I have had this 
 one-directional problem where the port range in /etc/asterisk/rtp.conf 
 was too broad, and the firewall on my server was only allowing a 
 smaller subset of RTP ports.

rtpstart=1
rtpend=2

which is exactly what my packet filter allows through.

I assume you have tried turning your packet filter or firewall off
completely (just for a moment) to see if it helped?


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[asterisk-users] One way audio - doesn't seem to be NAT issue

[Stefan Viljoen]

Hi D'arcy

Have you checked your RTP port ranges (I'm sure you have), and also that the
server IP for RTP as specified in the initial SIP is correct?

Not sure how this will relate to your setup, but we had something similar
here using Asterisk 1.8.11.0 on both sides of the connection, via a VOIP
service provider in the middle.

We had slightly different parameters, e. g. that we would have no RTP at
all, but a call that did connect to total silence, dialed from either side.

We subscribe to two trunk numbers provided by the VOIP service provider at
each site in Asterisk.

It turned out after carefully looking at the SIP flowing back and forth that
the service provider was providing an RTP server IP that specified not the
same IP as the SIP server (which is their standard practice) but a
-different- RTP server IP.

Due to the routing we have, neither system on either side of the SIP
negotiated call could send packets to this new RTP server IP.

We therefore added a route that specifically allowed that new RTP server
IP to be reached by both machines on both sides of the VOIP service provider
link.

So can you carefully check that the SIP-negotiated RTP streams are going to
IPs that are reachable in BOTH directions?

Also check what RTP port ranges are being used - I have had this
one-directional problem where the port range in /etc/asterisk/rtp.conf was
too broad, and the firewall on my server was only allowing a smaller subset
of RTP ports.

E. g. /etc/asterisk/rtp.conf specified 1 - 5 as allowable RTP ports,
but my firewalld firewall under Centos was only allowing 1 - 2 - so
I'd regularly get that my SECOND call to test the server would have audio in
one direction - because
Asterisk was allocating an RTP port on one side of the SIP call that was
outside the range my firewalld was allowing.

It might require some careful tracing of SIP messages, maybe you can try
this? Specifically try to determine what RTP port number is being negotiated
when you have your zero-audio back from the remote party - what RTP port and
RTP server IP is he using at that moment on his side?

Is that port allowed through all the PPP / network segments between you? Is
the IP / IPs between you used to transfer RTP reachable from his side?

Message: 1
Date: Tue, 11 Aug 2015 15:10:44 -0400
From: D'Arcy J.M. Cain da...@vex.net
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: [asterisk-users] One way audio - doesn't seem to be NAT issue
Message-ID: 20150811151044.79872ce9@imp
Content-Type: text/plain; charset=US-ASCII

Given that both of us can make and accept calls and the server is simply
connecting two separate channels I can't see where the problem might lie.
Can anyone suggest a possible setup issue?

I have tried so many things but I am willing to try them again.  Feel free
to make any suggestion no matter how silly.  I really need to fix this.

Cheers.


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Re: [asterisk-users] 786 000 files limit Centos 7 - Asterisk

[Stefan Viljoen]

Hi Tony

Thanks again for replying.

Ok, I've set the SYSMAXFILES and MAXFILES in the script, hopefully it will
have an effect.

Kind regards,

Message: 12
Date: Tue, 11 Aug 2015 16:50:29 + (UTC)
From: t...@softins.co.uk (Tony Mountifield)
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] 786 000 files limit Centos 7 - Asterisk
keepcomplaining
Message-ID: assp.06651285b6.mqd94l$qb1$1...@softins.softins.co.uk

In article 002b01d0d414$36af31b0$a40d9510$@verishare.co.za,
Stefan Viljoen viljo...@verishare.co.za wrote:
  Anybody else ran into this?
 
 No, but I would ask myself why so many file descriptors are being used.
 It sounds like you have a file descriptor leak (not being closed when
 finished with).
 
 Hi Tony
 
 Thanks for replying.
 
 I suspected something like that, though repeatedly running
 
 lsof | wc -l
 
 Always stays quite low - 100 000 open files, which is still 8 times less
 than the system maximum as confirmed by running ulimit -n

From what you said below, the above is probably not relevant...

 
 SOMETHING NEW
 
 I have now ran
 
 /usr/bin/prlimit --pid `pidof asterisk` 
 
 and I have noticed that even though I have 800 000 files specified, the
 ACTUAL limit in place on Asterisk for numbers of files is only 1024?!

Yes, this is likely. Have a look in /usr/sbin/safe_asterisk, at the
commented-out settings for SYSMAXFILES and MAXFILES, and try setting those.

Cheers
Tony
-- 
Tony Mountifield
Work: t...@softins.co.uk - http://www.softins.co.uk
Play: t...@mountifield.org - http://tony.mountifield.org



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End of asterisk-users Digest, Vol 133, Issue 13
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Re: [asterisk-users] 786 000 files limit Centos 7 - Asterisk

[Stefan Viljoen]

Hi Markus

Thanks for the reply, I have set those files previously as well...

It seems the problem for me is on my Centos 7 box that the Asterisk binary
does not know that these are the limits, and imposes a 1024 open file
limit count for some reason.

It seems using prlimit to force the running binary to use a higher file
limit is working - but it has to be done manually each time the asterisk
process is started, the prlimit cronjob I set up to force the file limit
higher does not seem to work - you have to physically type in 

/usr/bin/prlimit --pid `pidof asterisk` --nofile=786000:786000

and running that from a script (As root) doesn't seem to work.

Anyway, thanks for the advice and taking the time to reply.

Kind regards,

Message: 11
Date: Tue, 11 Aug 2015 18:29:01 +0200
From: Markus Weiler markus_wei...@mailworks.org
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] 786 000 files limit Centos 7 - Asterisk
keep complaining
Message-ID: 55ca22cd.4060...@mailworks.org
Content-Type: text/plain; charset=windows-1252; format=flowed

Hi Stefan,

we ran into a similar problem using Debian.

There we are able to check the current limits using:

pidof asterisk  - 23351
cat /proc/23351/limits

Output:
Limit Soft Limit   Hard Limit Units
Max open files1024 1024 files


I think that in the end

/etc/security/limits.conf

* hardnofile  50
* softnofile  50
root  hardnofile  50
root  softnofile  50


did the trick. We also tried

vi /etc/sysctl.conf
fs.file-max = 50

not sure what the solution in the end was. But I remember rebooting was 
important.


Markus



Am 11.08.2015 um 11:00 schrieb Stefan Viljoen:
 Anybody else ran into this?
 No, but I would ask myself why so many file descriptors are being used.
 It sounds like you have a file descriptor leak (not being closed when
 finished with).
 Hi Tony

 Thanks for replying.

 I suspected something like that, though repeatedly running

 lsof | wc -l

 Always stays quite low - 100 000 open files, which is still 8 times less
 than the system maximum as confirmed by running ulimit -n

 I also note that this number will increase to about 125 000 but never go
 higher than that, then, as calls hang up, decreate again - during times
when
 the CLI is spammed with 100s of broken pipe errors due to insuffiecient
 file descriptors, this number never reaches beyond 125  000 out of the
 available 800 000 open files.

 You might also want to look at the output of lsof (or at least some of
it)
 to see what all these file descriptors are pointing to, and whether it is
 indeed Asterisk that is consuming them.
 If I grep by asterisk on the output of lsof the few thousand lines I have
 looked at all seem to indicate legitimate uses - there are at least two
 files for each conversation in progress (I assume for inward and outward
 RTP) plus one for each file being mixmonitored (which also seems logical)
 and also number-of-active-calls connections to res_timing_dahdi - which
all
 looks correct...

 If it is Asterisk, it's quite possible, even probable, that such a leak
 has been found and fixed, even in the 1.8 series. 1.8.11.0 is rather old
-
 the latest is 1.8.32.3, so it would be best to update to that version and
 see if the problem persists.
 Ok, I will have to consider that. The thing is the problem is not
consistent
 - I can (for example) run 60 calls, with no problems and no reported
 failures in opening files, then calls will -decrease- to about 40 and then
 later spike to 70, but around 50 calls I get the errors coming up
thousands
 of times in the CLI, then suddenly stop as the calls -increase- which
 doesn't make sense. But this kind of behaviour does seem consistent with a
 possible leak.

 SOMETHING NEW

 I have now ran

 /usr/bin/prlimit --pid `pidof asterisk`

 and I have noticed that even though I have 800 000 files specified, the
 ACTUAL limit in place on Asterisk for numbers of files is only 1024?!

 # prlimit --pid `pidof asterisk`
 RESOURCE   DESCRIPTION SOFT  HARD UNITS
 AS address space limitunlimited unlimited bytes
 CORE   max core file size unlimited unlimited blocks
 CPUCPU time   unlimited unlimited seconds
 DATA   max data size  unlimited unlimited bytes
 FSIZE  max file size  unlimited unlimited blocks
 LOCKS  max number of file locks held  unlimited unlimited
 MEMLOCKmax locked-in-memory address space 65536 65536 bytes
 MSGQUEUE   max bytes in POSIX mqueues819200819200 bytes
 NICE   max nice prio allowed to raise 0 0
 NOFILE max number of open files1024  4096
 NPROC  max number of processes30861 30861
 RSSmax resident set size

Re: [asterisk-users] 786 000 files limit Centos 7 - Asterisk keep complaining

[Stefan Viljoen]

Hi Steve

Just running about 50 calls??

If I do

lsof | grep asterisk | wc -l

to narrow the realm of what is reported I still get just under 100 000
files:

lsof | grep asterisk | wc -l
95903

with 50 calls running.

So apparently this is excessive? I can only guess that there must be a MAJOR
bug in 1.8.11.0 given that you are running 347 calls on 2161 open files with
Ast 11...?

I'm on Centos 7.

As I mentioned previously, I have found that my problem with Asterisk
running out of files is that the Asterisk binary, on startup, does NOT
detect that the open file limit it 768 000, it always selects 1024 as the
soft limit for open files and 4096 as the hard limit for open files. I have
to manually (cron does not work - wonder why?) do

/usr/bin/prlimit --pid `pidof asterisk` --nofile=786000:786000

in order to get the running Asterisk instance to ACTUALLY have a soft and
hard number of files limit of 786 000.

The machine under discussion never runs more than about 60 calls, at which
point it consumes about 125 000 file descriptors / handles in Asterisk
(total open files in the kernel goes to about 195 000) - I thought this was
normal!

Note that besides some timer problems that pop up occassionally (unsure if
it is related) the box is totally stable and has spent days at 100 000+ open
file descriptors operating Asterisk 1.8.11.0 quite happily with no other
problems save that when the binary started up for that run it did not have a
sufficiently high file limit to actually work.

Thanks for the reply!

Kind regards,

 I suspected something like that, though repeatedly running

 lsof | wc -l

 Always stays quite low - 100 000 open files, which is still 8 times 
 less than the system maximum as confirmed by running ulimit -n

What the 'h' are you doing that takes x00,000 open files?

I'm running Asterisk 11.17.0 on CentOS 6.7 and my 'numbers' seem
insignificant by comparison.

sudo /usr/sbin/asterisk -r -x 'core show channels' | grep active
347 active channels
344 active calls

sudo lsof | wc -l
3945

sudo lsof | grep asterisk | wc -l
2161

--
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST



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Re: [asterisk-users] 786 000 files limit Centos 7 - Asterisk keep complaining

[Stefan Viljoen]

 Anybody else ran into this?

No, but I would ask myself why so many file descriptors are being used.
It sounds like you have a file descriptor leak (not being closed when
finished with).

Hi Tony

Thanks for replying.

I suspected something like that, though repeatedly running

lsof | wc -l

Always stays quite low - 100 000 open files, which is still 8 times less
than the system maximum as confirmed by running ulimit -n

I also note that this number will increase to about 125 000 but never go
higher than that, then, as calls hang up, decreate again - during times when
the CLI is spammed with 100s of broken pipe errors due to insuffiecient
file descriptors, this number never reaches beyond 125  000 out of the
available 800 000 open files.

You might also want to look at the output of lsof (or at least some of it)
to see what all these file descriptors are pointing to, and whether it is
indeed Asterisk that is consuming them.

If I grep by asterisk on the output of lsof the few thousand lines I have
looked at all seem to indicate legitimate uses - there are at least two
files for each conversation in progress (I assume for inward and outward
RTP) plus one for each file being mixmonitored (which also seems logical)
and also number-of-active-calls connections to res_timing_dahdi - which all
looks correct...

If it is Asterisk, it's quite possible, even probable, that such a leak
has been found and fixed, even in the 1.8 series. 1.8.11.0 is rather old -
the latest is 1.8.32.3, so it would be best to update to that version and
see if the problem persists.

Ok, I will have to consider that. The thing is the problem is not consistent
- I can (for example) run 60 calls, with no problems and no reported
failures in opening files, then calls will -decrease- to about 40 and then
later spike to 70, but around 50 calls I get the errors coming up thousands
of times in the CLI, then suddenly stop as the calls -increase- which
doesn't make sense. But this kind of behaviour does seem consistent with a
possible leak.

SOMETHING NEW

I have now ran

/usr/bin/prlimit --pid `pidof asterisk` 

and I have noticed that even though I have 800 000 files specified, the
ACTUAL limit in place on Asterisk for numbers of files is only 1024?!

# prlimit --pid `pidof asterisk`
RESOURCE   DESCRIPTION SOFT  HARD UNITS
AS address space limitunlimited unlimited bytes
CORE   max core file size unlimited unlimited blocks
CPUCPU time   unlimited unlimited seconds
DATA   max data size  unlimited unlimited bytes
FSIZE  max file size  unlimited unlimited blocks
LOCKS  max number of file locks held  unlimited unlimited
MEMLOCKmax locked-in-memory address space 65536 65536 bytes
MSGQUEUE   max bytes in POSIX mqueues819200819200 bytes
NICE   max nice prio allowed to raise 0 0
NOFILE max number of open files1024  4096
NPROC  max number of processes30861 30861
RSSmax resident set size  unlimited unlimited pages
RTPRIO max real-time priority 0 0
RTTIME timeout for real-time tasksunlimited unlimited microsecs
SIGPENDING max number of pending signals  30861 30861
STACK  max stack size   8388608 unlimited bytes

Accordingly I have put this into a cronjob ran each minute:

prlimit --pid `pidof asterisk` --nofile=786000:786000

to try and force the running binary to keep a high file limit (sources say
to keep it less than the ACTUAL system file limit, in my case 800 000 files)
on the live Asterisk process.

I'll see if this maybe helps - the above runs via cron each minute.

So it appears for some reason somehow the live running asterisk process
loses track of how many open files it may have, or when it starts it
somehow does not start with the correct number of maximum open files, as set
in the system / kernel config?

Anyway, thank you for replying, I'll monitor this new Cronjob fixup I'm
trying and see if it helps.

No wonder it is complaining about running out of file handles if it ACTUALLY
was only using 1024!

Kind regards

Stefan


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[asterisk-users] 786 000 files limit Centos 7 - Asterisk keeps complaining

[Stefan Viljoen]

Hi Guys

I keep getting messages in the Asterisk 1.8.11.0 CLI that there are not
enough file descriptors available on my Centos 7 box.

I also get regular error messages that RTP connections are failing due to
bad file descriptors.

I have already edited /etc/sysctl.conf by setting fs.file-max to 786 000
files, and running sysctl -p afterward.

I have also edited /etc/security/limits.conf and entered

* soft nofile 786000
* hard nofile 786000

and rebooted my system.

When I ge the Asterisk error messages in the CLI that there are not enough
file descriptors, I immediately ran

lsof | wc -l

and I only get around 100 000 files open - while my limit is 786 000 - yet
still Asterisk keeps complaining.

I have also edited /etc/asterisk/asterisk.conf to have 

maxfiles = 786000

and restarted Asterisk / rebooted, but this has no effect.

Why does Asterisk 1.8.11.0 complain that there are insufficient file
descriptors when my limit is 786 000 and the kernel itself states that only
100 000+ handles / decriptors are in use at the moment Asterisk is
complaining about insuffiecient descriptors?

Anybody else ran into this?

Thanks

Stefan

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
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Sent: Wednesday,August 5, 2015 07:00 PM
To: asterisk-users@lists.digium.com
Subject: asterisk-users Digest, Vol 133, Issue 5

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When replying, please edit your Subject line so it is more specific than
Re: Contents of asterisk-users digest...


Today's Topics:

   1. Re: Looking for PRI Card with automatic fail over (Eric Klein)
   2. Re: Looking for PRI Card with automatic fail over (Sam Basan)
   3. Re: Looking for PRI Card with automatic fail over
  (Technical Support)
   4. Update: Planned NASA trip around Astricon (Eric Klein)


--

Message: 1
Date: Wed, 5 Aug 2015 12:01:15 +0300
From: Eric Klein eric.kl...@greenfieldtech.net
To: asterisk-users asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Looking for PRI Card with automatic fail
over
Message-ID:
CAD4dZ0mvbu+yc=wsOGJLcjKE_K717tMVoG+1XQ1J6=8x6r_...@mail.gmail.com
Content-Type: text/plain; charset=utf-8

Responding to Matt Riddell, M Shokuie, A J Stiles, and Telium Support (all
below)

Thanks for the suggestions, but a separate gateway is a problem in the
design as the configuration is basically:

Carrier --- PRI --- Asterisk (with failover)  Legacy PBX
  |
  - SIP / IAX2 -- our service or remote office

As this is already acting as a gateway adding another box to the mix seems a
bit over kill (unless I can embed it and the asterisk PBX into a 2 U pizza
box). But since most gateways are designed to be used for fail over servers
/ PBXs not lines I am not sure that this would benefit us.

So far I have found 1 company that seems to have a gateway card and that is
the beroNet PCI / PCIe Gateway Cards which they say has the card level
failover. I will update the list should this meet the use case.
Eric

--

 Message: 1
 Date: Mon, 3 Aug 2015 13:54:05 -0500
 From: Matt Riddell li...@venturevoip.com
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 Subject: Re: [asterisk-users] Looking for PRI Card with automatic fail
 over
 Message-ID: 2539d3b3-4981-468f-95af-34d9d6a17...@venturevoip.com
 Content-Type: text/plain; charset=us-ascii

 Yep it was red fone

 http://red-fone.com

 7 years ago :-)

 http://www.venturevoip.com/detail.php?news_id=1927

 --
 Cheers,

 Matt Riddell
 ___

 http://www.venturevoip.com/news.php (Daily Asterisk News) 
 http://www.venturevoip.com/pabx_on_disk.php (PABX on a Disk) 
 http://www.venturevoip.com/exchange.php (Full ITSP Solution) 
 http://www.venturevoip.com/cc.php (Call Centre Solutions)

 -- next part -- An HTML attachment was 
 scrubbed...
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 http://lists.digium.com/pipermail/asterisk-users/attachments/20150803/
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 --

 Message: 2
 Date: Mon, 3 Aug 2015 23:31:29 +0430
 From: M Shokuie sena...@gmail.com
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 Subject: Re: [asterisk-users] Looking for PRI Card with automatic fail
 over
 Message-ID:

[asterisk-users] CEL eventtime incorrect, but CDR times are correct - 1.8.11.0

[Stefan Viljoen]

Hi list

I have a huge problem with a 1.8.11.0 Asterisk instance not logging CEL
events with the correct eventtimes.

I'm logging via ODBC to MariaDB 15.1 Distrib 10.0.20-MariaDB

I'm logging into a MyISAM table.

If I start the Asterisk instance, logged times are correct, but the longer
the box runs the more the eventtime in the CEL rows created by Asterisk via
ODBC drift backwards.

E. g. the clock on the server says 08:15 for example (I enter the date
command in the terminal) and if I run a query and check the most recent
CELs immediately and about ten minutes after startup, they are correct.

However, from about 15 minutes onward the CEL eventtime starts regressing.
It will be 08:30 according to the server clock, but the CELs logged will
have a time of 08:20. The longer the asterisk instance runs the more time on
the CEL eventtime field falls behind. After four hours, when real time on
the server clock is 12:00 (for example) the CELs entered into the db by
Asterisk are only up to 10:45... while CDRs are 100% correct and show times
of 11:59, 12:00, 12:01, etc

I thought this was load, but in normal use the CEL table entries will go
idle was my users stop phoning - e. g. it is not as if the entries are
lagging due to the table being too slow to write into from Asterisk's side.

If a new user phones after, say, ten minutes of inactivity, his new CELs
will still be behind, just as far as they were behind on eventime when calls
stopped and CELs stopped being entered.

The only way to fix it is to stop and restart the Asterisk instance.
Strangely, CDR times (logged to the same MariaDB db via ODBC) are 100%
correct and totally accurate no matter how long the Asterisk instance runs,
or how high the load goes.

What causes this? How can I correct it?

Any replies appreciated!

Thanks

Stefan




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[asterisk-users] Filters

[Stefan Viljoen]

Hi list

I'm using Asterisk 1.8.11.0 - is there any way to apply (for example) a
bandpass filter to Asterisk RTP audio in the realtime audio stream?

I'm looking for a way to (for example) filter out a 50Hz AC hum present in
some calls I push through my asterisk.

Thanks

Stefan


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Re: [asterisk-users] Centos 6.5 Asterisk 1.8.11.0 - starts in rc.local, but not contactible?

[Stefan Viljoen]

Hi JG

Thanks for replying.

Depending on the hardware you are using, simply calling asterisk might not
be enough, as there 
could be dependencies on third party drivers. Depending on how asterisk was
installed, one 
probably also has to look at various permissions. For example, asterisk -r
might fail simply 
because you are calling it from an account with insufficient rights. It's
difficult to tell 
given your information.

Maybe the following will help. If you look into the contrib/init.d
directory (inside the src 
tree) you'll find the rc.redhat.asterisk script. Rename it and put it into
the /etc/init.d 
directory and issue chkconfig --add asterisk as well as chkconfig
asterisk on and your 
problem should be solved. You can check the current settings with
chkconfig --list asterisk. 
The Redhat script works nicely under CentOS.

Hmm ok - I made sure to run

make config

in the Asterisk source folder which installed the correct scripts into rc.d
and so forth.

I then did

chkconfig asterisk on

and rebooted the box.

The parameters remain the same, asterisk is there if you do a ps -aux | grep
asterisk but it still is in a non-working state and not contacible via
asterisk -r.

Since it is an old box and the reason for trying to get it going is mostly
academic, I think I'm just going to dump the box and reformat it with Centos
7.

Strange though, I have installed about 17 other boxes exactly this way on
broadly the same hardware and all are currently running fine with Centos 6.5
and Asterisk 1.8.11.0

Thanks anyway, the problem is clearly deeper than I though since even the
official way you detail above fails to start Asterisk as an account that
can start it on system boot.

Even when root is the only account on the machine - which leads me to
believe I have some basic error in my Centos 6.5 installation so I'll just
try it again or try Centos 7.

Thanks a lot.

Kind regards

Stefan


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[asterisk-users] Centos 6.5 Asterisk 1.8.11.0 - starts in rc.local, but not contactible?

[Stefan Viljoen]

Hi list

I'm trying to get Asterisk 1.8.11.0 to start automatically when my Centos
6.5 box boots.

I've done this many times before, but for some reason, on this box and
hardware (older Core i3 system, 4GB RAM) I cannot get Asterisk to be
contactible after boot.

E. g. in rc.local I have, as the last line

---
asterisk
---

as in all my other Asterisk boxes with Centos 6.5 and Asterisk 1. 8.11.0

This -does- start asterisk on boot, but you cannot connect to it using

asterisk -r

the error being

---
[root@asterisk ~]# asterisk -r
Asterisk 1.8.11.0, Copyright (C) 1999 - 2012 Digium, Inc. and others.
Created by Mark Spencer marks...@digium.com
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for
details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it
under
certain conditions. Type 'core show license' for details.
=
Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?)
[root@asterisk ~]#
---

And I note that /var/run/asterisk.ctl does NOT exist, which is why asterisk
-r does not work...

I do note that there are error messages in /var/log/asterisk/messages, to
whit:

---
[Jul 23 10:59:31] WARNING[1863] loader.c: Error loading module
'codec_g729.so': /usr/lib/asterisk/modules/codec_g729.so: cannot enable
executable stack as shared object requires: Permission denied
[Jul 23 11:02:22] ERROR[1900] cdr_tds.c: Unable to connect: Adaptive Server
is unavailable or does not exist (20009)
[Jul 23 11:02:22] ERROR[1900] cdr_tds.c: Permission denied (13)
[Jul 23 11:02:22] ERROR[1900] cdr_tds.c: Unable to connect to sqlserver
---

However, if I kill -9 the uncontactible asterisk process and start it
manually as root, everything works and runs fine, and I can connect to the
running asterisk instance via asteriskr -r in a terminal.
The above error messages are not emitted if asterisk is started by hand like
this.

Why does Asterisk have persmission problems when started out of
/etc/rc.local (as root) vs. when I start it by hand while logged in as root?

No config changes made to Centos 6.5 except for installing it from DVD and
running a yum update.

Thanks

Stefan


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[asterisk-users] CEL eventtime drift

[Stefan Viljoen]

Hi list

I'm running MariaDB with Asterisk 1.8.11.0 over ODBC for CEL and CDR
logging.

CDRs work fine, times logged in the DB are correct consistently.

However, I've noticed that for CELs eventtimes start lagging severely.

E. g. I'd start Asterisk at 12:15 and entries made into CEL and CDR
immediately as users start phoning away have the correct times in the
relevant fields.

However, as time passes CELs start lagging behind. E. g. at 12:30 real-time,
CDRs still log as 12:30, but by now CELs have lagged behind and are showing
12:24 (for example) for -current- entries which should be in the 12:30
range.

The longer the Asterisk stays up, the larger the lag, e. g. afer an hour
CDRs still log correct at current time, but CELs have drifted almost 30
minutes back... e. g. real-world time on the Asterisk's clock is 14:00, CDRs
log as 14:00 but CEL log as if  it is 12:40.

Anybody encountered this before - how can I stop CEL time drift?

Thank you,

Stefan


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[asterisk-users] Asterisk 13.4.0 - mixmonitor only records one side's perspective

[Stefan Viljoen]

Hi All

I have a problem with mixmonitor in 13.4.0 doing the following:

1. Caller phones in

2. Reception picks up

3. Talks to caller

4. Does attended transfer, talks to manager to screen the caller wanting to
speak to him

5.  Complete the transfer by putting down her handset so the caller can
speak to the manager

6. Caller talks to the manager

The problem is that mixmonitor only records steps (3) and (6) - for step (4)
in the recording file, you hear the MOH the outside caller hears while the
receptionist is screeing the call.

We want to record the -entire- conversation, including the receptionist
screening with the manager.

E. g. you literally only hear the perspective of the originator of the call
- you never hear the receptionist - manager leg of the call if a transfer
occurs during a call.

Here's how an incoming call is handled - on 1.8 the below code recorded the
entire conversation flawlessly, including the screening conversation of
reception - manager

[inc]

exten=_[123]xxx,1,Macro(VCRECORD,${MACRO_CONTEXT}EXT${CALLERID(num)}ACC${CD
R(accountcode)},${ARG2})
exten=_[123]xxx,n,Set(__TRANSFER_CONTEXT=call-redirect)
exten=_[123]xxx,n(checkacc),NoOp(MY Account code is ${CDR(accountcode)})
;exten=_[123]xxx,n(checkacc),NoOp(OTHER Account code is
${CDR(accountcode)})
exten=_[123]xxx,n(dodial),Dial(Sip/${EXTEN},120,tTg)
exten=_[123]xxx,n,NoOp(Dialstatus: ${DIALSTATUS})
exten=_[123]xxx,n,GotoIf($[${DIALSTATUS}=NOANSWER]?takevoicemail:checkd
ont)
exten=_[123]xxx,n(checkdont),GotoIf($[${DIALSTATUS}=DONTCALL]?takevoice
mail:donecall)
exten=_[123]xxx,n,NoOp(Taking a voicemail...)
exten=_[123]xxx,n(takevoicemail),VoiceMail(${EXTEN}@default)
exten=_[123]xxx,n(donecall),Hangup()

[call-redirect]

include = parkedcalls

exten=_[123]xxx,1,NoOp(Transferring Call. This Channel ${CHANNEL}, Other
channel ${BLINDTRANSFER})
exten=_[123]xxx,n(dodial),Dial(Sip/${EXTEN},120,tTg)

[macro-VCRECORD] ; MACRO To setup Recording
;${ARG1} Description To Save
;${ARG2} Dialed Number
exten=s,1,NoOp(Start of MixMonitor recording)
exten=s,n,Set(IAXVAR(accountcode)=${CDR(accountcode)})
exten=s,n,GoToIf($[${MIXMONITOR_FILENAME} = ]?startrec:finrec)
exten=s,n(startrec),Set(recDir=${STRFTIME(${EPOCH},,%y%m/%d)})
exten=s,n,Set(recFile=${recDir}/${STRFTIME(${EPOCH},,%y%m%d%H%M%S)}D${ARG1}
N${ARG2}ID${UNIQUEID}.gsm)
exten=s,n(setacc),set(recFile=${recDir}/${CDR(linkedid)}.gsm)
exten=s,n(makedir),System(/bin/mkdir -p
/var/spool/asterisk/monitor/${recDir})
exten=s,n,MixMonitor(${recFile},a)
exten=s,n(finrec),Set(AUDIOHOOK_INHERIT(MixMonitor)=yes)
exten=s,n,Set(CDR(userfield)=${MIXMONITOR_FILENAME})
exten=s,n,Set(__chanrecording=/var/spool/asterisk/monitor/${recFile})
exten=s,n,NoOp(Recording to ${MIXMONITOR_FILENAME})
exten=s,n,UserEvent(RecordingToFile,Uniqueid: ${UNIQUEID},Channel:
${CHANNEL},FileName: ${MIXMONITOR_FILENAME})
exten=s,n,MacroExit

Can somebody help or offer a suggestion how to get MixMonitor to record an
ENTIRE conversation in 13.4.0, not just record from the perspective of the
initiator of a call?

Thanks


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[asterisk-users] For a failed retransmission - what were the IP addresses?

[Stefan Viljoen]

Hi Guys

Given these occassional errors on my Asterisk CLI:

[Jul  2 10:23:36] WARNING[2060]: chan_sip.c:3641 retrans_pkt: Retransmission
timeout reached on transmission
17bb3a993ad10f8818970ae952b81e73@192.168.11.31:5060 for seqno 102 (Critical
Request) -- See
https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32000ms with no response
 [Jul  2 10:23:49] WARNING[2060]: chan_sip.c:3641 retrans_pkt:
Retransmission timeout reached on transmission
2f6fa425581373c11c2ae58a276751bb@192.168.11.31:5060 for seqno 102 (Critical
Request) -- See
https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32000ms with no response

E. g. I have the transmission numbers

17bb3a993ad10f8818970ae952b81e73

and

2f6fa425581373c11c2ae58a276751bb

on which packets timed out.

Is there any way I can determine the IP addresses where those packets
referred to by these errors timed out on?

What is Sequence 102?

192.168.11.31 is my Asterisk 1.8.11.0 server.

Thanks!

Regards

Stefan


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Re: [asterisk-users] Strange and complete failure of Asterisk 1.8

[Stefan Viljoen]

Re: Strange and complete failure of Asterisk 1.8 (Duncan Turnbull)
Re: Strange and complete failure of Asterisk 1.8 (Markus Weiler)

Thanks Marcus  Duncan

Pulled the machine and replaced it with a brand new one. Same network and
same DNS server active there.

New system is running the same Asterisk (1.8.11.0) and the same Centos, same
Asterisk config files - still running beautifully since yesterday.

Got the sick system at our lab now and pulling it apart - even isolated from
the LAN (e. g. with no access at all to a DNS server) it still is exhibiting
the same problem.

Connecting it to the LAN with our HeadOffice DNS server present (which also
serves our live HO asterisk) it still exhibits the problem.

But thanks anyway, if it happens again well just pull the live system and
replace from the ready-reserve we maintain of already configured Ast boxes.

I'll keep digging awhile with the sick system and see if I can come up with
something, will focus on DNS - thanks for the pointers!

Kind regards

Stefan


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[asterisk-users] FW: Strange and complete failure of Asterisk 1.8 - part 2

[Stefan Viljoen]

Hi guys

I just did a ps -Af | grep asterisk on the machine and got several screens
full of this:

root  6970  6946  0 13:10 ?00:00:00 rasterisk rxcore show
channels verbose
root  6987  6948  0 13:10 ?00:00:00 rasterisk rxcore show
channels verbose
root  7005  6985  0 13:10 ?00:00:00 rasterisk rxcore show
channels verbose
root  7021  7003  0 13:10 ?00:00:00 rasterisk rxcore show
channels verbose
root  7062  7053  0 13:11 ?00:00:00 rasterisk rxcore show
channels verbose
root  7077  7058  0 13:11 ?00:00:00 rasterisk rxcore show
channels verbose
root  7097  7060  0 13:11 ?00:00:00 rasterisk rxcore show
channels verbose
root  7118  7093  0 13:12 ?00:00:00 rasterisk rxcore show
channels verbose
root  7137  7095  0 13:12 ?00:00:00 rasterisk rxcore show
channels verbose
root  7155  7135  0 13:12 ?00:00:00 rasterisk rxcore show
channels verbose
root  7170  7152  0 13:12 ?00:00:00 rasterisk rxcore show
channels verbose
root  7196  7186  0 13:13 ?00:00:00 rasterisk rxcore show
channels verbose
 
I see that rasterisk is a symlink in /usr/sbin that points to whatever is
at /usr/sbin/asterisk - and that /usr/sbin/asterisk is the normal asterisk
binary.

Trying to killall rasterisk or kill -9 a specific PID of rasterisk just
results in it being immediately respawned...

Same with killall asterisk or kill -9 the single running asterisk instance.
It immediate respawns. I have no scripts or facilities running which will
restart it.

Any ideas?


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[asterisk-users] Strange and complete failure of Asterisk 1.8

[Stefan Viljoen]

Hi all

We've had a very strange failure on an Asterisk 1.8 install that has been
running for about a year at a customer site.

The physical hardware is fine, all other services off the Centos 6.5 server
are running. Only Asterisk is not working...

The first symptom was that no calls can be made over the SIP phones used
with it, and no calls could be received over the SIP trunk connected to it.

I checked and noted that 

sip show peers

in the CLI would either do nothing (e. g. just show asterisk*cli again,
with no response) or it would return only this:

asterisk*CLI sip show peers
Name/username  HostDyn
Forcerport ACL Port Status
asterisk*CLI

A module show like sip also does literally nothing, just

asterisk*cli module show like sip
asterisk*CLI

Soon after this, I lost the capacity to get any response if I do an asterisk
-r on the commandline - it would just hang indefintely.

Did a reboot, and then, I couldnt start asterisk at all - entering

# asterisk

would also just hang.

So, I recompiled asterisk from source and reinstalled the executable and all
the module files. Still the same.

I happened to have an older asterisk executable from a few months before
laying around and sha256summed it - and there was a difference in the
checksum vs. the non-working asterisk binary - BUT it turned out that the
newly recompiled asterisk binary has the SAME SHA256 checksum as the
non-working asterisk binary.

System seems fine otherwise, nothing relevant in /var/log/messages or dmesg
indicating a hardware failure. /var/log/asterisk/messages also contains no
strange warnings or errors.

Anybody got any idea why I cannot resuscitate my Asterisk install, even
after recompiling it from scratch from source? Why would asterisk die like
this in the first place?

Thanks


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[asterisk-users] Too many open files - 786 000 already specified as max num open files?

[Stefan Viljoen]

Hi guys

I have a site on Asterisk 1.8.11.0 running in Centos 6.5 that has about 150
concurrent callers.

I keep getting these types of messages in the CLI:

[May 21 11:39:21] WARNING[18469]: channel.c:1189 __ast_channel_alloc_ap:
Channel allocation failed: Can't create alert pipe! Try increasing max file
descriptors with ulimit -n
[May 21 11:39:21] WARNING[18469]: chan_sip.c:7041 sip_new: Unable to
allocate AST channel structure for SIP channel
[May 21 11:39:21] WARNING[18469]: res_rtp_asterisk.c:459 create_new_socket:
Unable to allocate RTCP socket: Too many open files
[May 21 11:39:21] ERROR[18469]: acl.c:706 ast_ouraddrfor: Cannot create
socket

I have specified this on the commandline:

ulimit -n 768000

I have also tried 

ulimit -n 512000

ulimit -n 65535

before going up to 768000

and then restarted Asterisk.

In /etc/sysctl.conf I have at the bottom

fs.file-max=768000

after also trying 65535, then 512000 .

I have run sysctl -p as root asfter adjusting the fs.file-max limit in
/etc/sysctl.conf

Furthermore in /etc/security/limits.conf I have at the bottom

asterisk soft nofile 768000
asterisk hard nofile 768000

/etc/asterisk/asterisk.conf contains in the maxfiles line 768 000 and the
line has been uncommented

After changing all of the above I have restarted asterisk.

SELinux is completely disabled.

I'm still getting the CLI error messages that there are too many open files
- can I really be exceeding 768 000 open files with 150 callers using normal
SIP phones?

The system is a fairly standard Core i7 with 8GB of RAM, load averages on it
rarely go about 0.5 with all callers running, and memory usage never going
above 4GB of 8GB physical ram accoridng to top.

Anybody got any advice or comments?

Thanks!

Stefan


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Re: [asterisk-users] Monitoring SIP Service (Jai Rangi)

[Stefan Viljoen]

Interesting approach.

What we've done is to write an app that runs on a separate machine that
simply does some asterisk -rx calls to the running Asterisk instance via an
SSH library and then evaluate the string returned.

For example, to monitor our registered SIP service providers, we compare the
output for the sip show registry CLI command when the system is in the
correct state, to the output we get on timed checks every five minutes.

If a SIP trunk goes down, the just-obtrained string does not match the
stored, correct string anymore, and our app starts sending SMSes / raising
alarms.

Been running for about three years now and works suffieciently for our
widely separated branches all over South Africa.


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Re: [asterisk-users] asterisk-users Digest, Vol 130, Issue 14

[Stefan Viljoen]

- Original Message -
 From: Steve Davies davies...@gmail.com
 To: Asterisk Users Mailing List - Non-Commercial Discussion 
 asterisk-users@lists.digium.com
 Sent: Wednesday, May 13, 2015 11:39:29 AM
 Subject: Re: [asterisk-users] Retransmission Timeout results in 
 dropped calls after 32 seconds
 
 Hi,
 
 In my experience, all Yealink phones work just fine with Asterisk, we 
 have hundreds (perhaps even low-thousands) out there with customers on 
 Asterisk 1.2, 1.6.2, 1.8 and 11.
 
 If you are accurately representing the SIP trace on the phone and the 
 SIP trace on Asterisk, then I would strongly suggest a SIP ALG exists 
 in the network between the two devices and that SIP ALG does not 
 understand SIP properly. The two halves simply do not match, so 
 something must surely be interfering.
 
 In my experience it is often an innocent looking Cisco router. Cisco's 
 SIP implementation is SIP By Cisco rather than RFC compliant SIP. 
 If that is the case Cisco call it a SIP fixup and you just need to
disable it.
 
 Hope that helps,
 Steve
 
Steve,

That is an interesting point - the server and the phone are both connected
to Netgear switches where I have enabled their Auto-VoIP feature, which
remarks packets based on protocol (SIP, SCCP, etc) for better QoS:
http://kb.netgear.com/app/answers/detail/a_id/21758

I wonder if this remarking process is modifying another part of the packet
too?
Both devices are on the same subnet, so although these switches do route
traffic as well, that shouldn't be coming into play here.

We had a very specific issue at a client recently with Yealink phones and
Cisco routers. No Yealink we plugged in would pick up an IP from the DHCP
server so it could be configured.

To correct it we had to explicitly turn off LLDP (link layer discovery
protocol) on the Yealinks in order to get them to work on the Cisco'ed
network.

Did not fool around with the Cisco to try and solve the issue, and still
don't know exactly what was going wrong, but it seems defined that using
recent model Yealink phones on a network with a recent Cisco router, either
turn off LLDP on the phones or the router to prevent problems.

The above still did not solve the issues completely, still getting one or
two complaints from this customer every few days which also seem to be
related to the fact of using Yealink phones with a network where a Cisco
router is present... so your milage may vary.

Stefan


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[asterisk-users] Linking Asterisk 1.8 to late model Samsung PABX over PRI - transfer issues

[Stefan Viljoen]

Hi all

I've got a setup where I use a Sangoma PRI card driven via Sangoma WanPipe
to connect to a legacy Samsung PABX (I'm not sure which model) form Asterisk
1.8.11.0.

The reason is the customer has a large installed base of Samsung phones
physically connected to it and on each users desk. They did not want to
spring for a complete replace of all their Samsung phones with generic, and
dump their existing Samsung PABX (worth several hundred thousand in local
currency) for a 100% SIP phone + Asterisk setup.

They do want the calls recorded and accessed in a custom system, ergo
Asterisk - raison d'etre for the setup - it records and the 3rd party system
interfaces to it via the AMI to playback and record.

The Asterisk stands between the Samsung and the telco's SIP trunk, and
presents PRI NET to the Samsung, so it thinks it is directly connected to
the telco PRI trunk.

The Asterisk in turn connects to the telco --SIP-- trunk. (E. g. we protocol
translate on Asterisk level - it speaks PRI to the Samsung and SIP to the
telco)

Everything works fine, EXCEPT when users do transfers. If they Asterisk
transfer (#number) on the Samsung phones, it works fine, but ONLY if the
target Samsung phone is not busy with a conversation.

If the target Samsung phone IS busy with a conversation, that conversation
on the Samsung is cut off, and the person outside wanting to talk to the
consultant is also cut off - MOH ends and Asterisk emits standard CLI info
and runs the H extension on  the call, as it should for a normal hangup.

If they Samsung transfer (e. g. use the physical transfer button on the
Samsung handset) it works fine - if the user is busy, it rings back, if not,
the call goes through.

The reason they have to Asterisk transfer is that Asterisk can keep track
of which extension is actually linked to a call - this is vital for the 3rd
party system we have interrogating asterisk for extensions status for
business logic purposes over AMI.

If they Samsung transfer asterisk is oblivious of the new extension on
which the call is running, and therefore business logic breaks - it only has
that 1000 (reception) is busy, and has no inlking of the fact that, in the
Samsung, 1000 has transferred to 1023 (for example) and business logic
should run on 1023, NOT 1000...

Can anybody offer any comments?

Thanks

Stefan


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Re: [asterisk-users] System() command refuses to execute bash script

[Stefan Viljoen]

Hi all

I got this solved.

Turns out the script WAS executing, but I forgot that apparently you need to
follow cron rules in any BASH scripts executed via System() from an
Asterisk dialplan.

E. g. all paths must be fully and absolutely specified, there are no
relative path references available.

So I changed the the file wireless.sh which was:

#!/bin/bash
touch wireless-executed

to

#!/bin/bash
touch /root/wireless-executed

and now if I phone 800 on my Asterisk context I can see that the file 

wireless-executed

is created by the touch binary, which indicates that Asterisk has run the
script successfully.

So the problem was not Asterisk or BASH or permissions, but rather that it
appears that all paths in any System() script must be absolutely, not
relatively, specified.

Hope this helps somebody - simply follow cron rules when writing
Asterisk-callable BASH scripts which you plan to trigger via the System()
Asterisk dialplan application.

Regards

Stefan


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[asterisk-users] System() command refuses to execute bash script

[Stefan Viljoen]

Hi All

I'm using this extension to  try and get Asterisk 1.8.11.0 to run a bash
script:

exten=802,n,System(/bin/sh -f /root/wireless.sh)

This file is

-rwxr-xr-x  1 root root   171 Mar  2 16:23 wireless.sh

e.g. root owns the file, and it has execute permissions for all users.

Asterisk runs as root as well.

Asterisk executes the command without any errors at max verbosity.

The file wireless.sh contains:

#!/bin/bash
touch wireless-executed

In my /root folder however, the file

wireless-executed

is never written - indicating the script does not execute, even though it is
owner by the same user asterisk runs as, and is world-executable.

How can I use System to run a bash script?

Thanks!

Regards

Stefan


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Re: [asterisk-users] TimerFD errors if MTU size is set incorrectly - SIP trunk

[Stefan Viljoen]

Hi Guys

Regarding this I found the following links which appear relevant:

https://issues.asterisk.org/jira/browse/ASTERISK-19347

https://issues.asterisk.org/jira/browse/ASTERISK-18223

It seems that this issue is related to that, NOT to a too-large MTU size.

Don't know if anybody can comment? Is the MTU size a red-herring as relates
the timer-fd errors?

Thank you very much

Stefan


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[asterisk-users] Timer_fd, pthreads, or DAHDI timer for timing under 1.8.11.0?

[Stefan Viljoen]

Hi guys

I have some questions regarding the above

1. Why are there different options for timing?

2. What are the differences between these types of timing sources?

3. When should you use what?

4. Is one timer type more reliable for an Asterisk system under heavy
loads than another while NOT using any DAHDI hardware - SIP only?

I'm of course referring to the familiar spamming timer_fd error messages in
the CLI with increasing load until asterisk crashes phenomenon.

Using 1.8.11.0 on Centos 6.5

Any thoughts appreciated!

Thank you

Stefan


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[asterisk-users] TimerFD errors if MTU size is set incorrectly - SIP trunk

[Stefan Viljoen]

Hi all

Is there a relation between the above?

I'm having a problem where I suspect my internet access provider (through
whom I go to a SIP trunk provider) have got MTU size problems.

My asterisk (1.8.11.0) is constantly going into the situation where a
TimerFD error is spammed in the CLI, load goes up and up until the system is
completely unusable.

I have an admission by the ISP that their MTU size on their fiber NTU may
be incorrect.

Can an incorrect MTU size on an ethernet connection cause Asterisk to
experience errors with the kernel timing source?

Thank you

Stefan


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[asterisk-users] 1.8.11.0 - CLI error res_timing_timerfd

[Stefan Viljoen]

Hi all

Sometimes (about every three months) some of my Asterisk 1.8 boxes will
start running this message thousands of times in the CLI:

[Feb 12 14:18:23] ERROR[28129]: res_timing_timerfd.c:180 timerfd_timer_ack:
Call to timerfd_gettime() error: Invalid argument
[Feb 12 14:18:23] ERROR[28129]: res_timing_timerfd.c:180 timerfd_timer_ack:
Call to timerfd_gettime() error: Invalid argument
[Feb 12 14:18:23] ERROR[28129]: res_timing_timerfd.c:180 timerfd_timer_ack:
Call to timerfd_gettime() error: Invalid argument
[Feb 12 14:18:23] ERROR[28129]: res_timing_timerfd.c:180 timerfd_timer_ack:
Call to timerfd_gettime() error: Invalid argument
[Feb 12 14:18:23] ERROR[28129]: res_timing_timerfd.c:180 timerfd_timer_ack:
Call to timerfd_gettime() error: Invalid argument
[Feb 12 14:18:23] ERROR[28129]: res_timing_timerfd.c:180 timerfd_timer_ack:
Call to timerfd_gettime() error: Invalid argument
[Feb 12 14:18:23] ERROR[28129]: res_timing_timerfd.c:180 timerfd_timer_ack:
Call to timerfd_gettime() error: Invalid argument

ad infinitum

Sometimes, this will disappear after a few minutes, and not re-appear. On
other occassions, after this started appearing in the CLI, the Asterisk will
crash without any slowdown or problems. On yet other occassions, the box
will start getting slower, load average rising and rising until it become
virtually frozen and it has to be rebooted in order to come back up in a
functional state again.

I've also noticed that stopping the Asterisk process and restarting it does
not help  if this error is in effect and a slowdown (rising load average)
has started - the whole physical machine has to be rebooted in toto.

I'm running on Centos 6.5.

Anybody else seen this message before? What does it mean?

Thanks

Kind regards

Stefan


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Re: [asterisk-users] constantly increasing load in Asterisk 11.14 (Sebastian Damm)

[Stefan Viljoen]

Have you considered doing a daily reboot?

In our shop (about 14 sites, busiest doing about 90 000 calls per day) we
found it best to reboot each Asterisk instance at 23:45 - we're still
running an ancient version (1.8.0.11).

Once this regime was instituted our major Asterisk issues stopped,
especially load-related issues linked to uptime.

Note that we do not just restart the Asterisk process, we reboot the entire
machine. We're running on bare-metal on Centos 6.5.

Maybe you can try doing that, if at all practical? Our business needs
fortunately this side is telecoms are only requires 08:00 - 17:00...

Regards

Stefan

Message: 2
Date: Thu, 5 Feb 2015 11:53:38 +0100
From: Sebastian Damm d...@sipgate.de
To: Asterisk asterisk-users@lists.digium.com
Subject: [asterisk-users] constantly increasing load in Asterisk 11.14
Message-ID:
cabkwsfwpe_cnaj8hnezkminys6odrocf6cv+vzyezh+tfty...@mail.gmail.com
Content-Type: text/plain; charset=utf-8

Hi,

we have quite a few Asterisk machines running and try to keep them on a
current version of the Asterisk 11 branch. But since we upgraded to 11.14.0
a couple weeks ago, we have to restart the Asterisk process every week
because the load gets too high and our monitoring complains.

Those machines are doing only SIP-to-SIP call relay, the dialplan is quite
complex, transcoding is done only on a few percent of the calls processed.
During the daytime, there are at max around 200 SIP channels (100 calls)
running at the same time. After one week, one machine has processed about
170k calls.

I have uploaded a comparison of cacti load graphs for one week of a machine
running with 11.14.0 and one running with 11.6.0: http://pbrd.co/1v0SO3R

As you can see, after a restart, both machines have about the same load.


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[asterisk-users] Meaning of core show hint output

[Stefan Viljoen]

Hi all

If I have the following in my dialplan:

exten=25001,hint,SIP/25001

Doing a

core show hint 25001

results in

   25001@local   : SIP/25001
State:IdleWatchers  0
1 hint matching extension 25001

in the Asterisk CLI.

What does the 

Watchers 0

mean?

I use the hints table output via core show hints for logic in my dialler
application - but what is a watcher in the context of dialplan hints?

Is this something I can make use of? My dialler app uses SSH to get the
asterisk -rx output of core show hints and then string processes the result
to determine which extensions are busy, et al.

Can I for example somehow subscribe as a watcher to a hint or somesuch?

Thank you

Stefan



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Re: [asterisk-users] Asterisk executable suddenly about 40KB

[Stefan Viljoen]

What you may want to consider is if you have a network management system
such as Nagios is create a service that checks the size of the binary every
5 minutes. You're notified if the size goes over a certain threshold. You
can also take the perf data and graph it using one of the many Nagios
graphing tools available. You can even use something like Munin for a task
like this. I couldn't get along without this. On some PBX's I have, I
monitor over 600 different metrics spread out every 1,5,10,15,30, and 60
minutes. Because they're spread out, the load average from these checks is
zero. Just a suggestion.
Regards;
John

Hi John

Thanks for the suggestions I'll keep them in mind.

Just to confirm back to the list, the chattr +i and turning off prelinking
seems to have solved the issue with the executable file being corrupted /
touched. Been running fine for a week now.

We have a very simple monitoring app running which we wrote ourselves which
monitors some metrics on the Asteisk, but nothing near as comprehensive as
you apparently use...

Anyway, thanks for the reply!

Kind regards

Stefan


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Re: [asterisk-users] Asterisk executable suddenly about 40KB larger - modules (Andres)

[Stefan Viljoen]

I would also start by putting an audit rule on the binary. Something like
this:
auditctl -w /usr/sbin/asterisk  -p war -k asterisk-bin

then you can get a report on who modified it and when by using:
ausearch -f /usr/sbin/asterisk

Its a start, but eventually you might need to monitor even keystrokes with
pam_tty_audit.so to understand who is doing this:
http://poorlydocumented.com/2014/05/enabling-pam_tty_audit-on-rhel-centos-o
r-scientific-linux/

Thanks I'll keep that in mind.

Just to report back, stopping pre-linking as detailed yesterday and setting
immutable with chattr on the Asterisk executable on the Head Office box here
appears to have solved the problem. The box did not crash this morning as it
did the previous two days and is working fine... strange, but good.

Previous to the problem starting on Tuesday, the box had been running fine
for about three years 24/7 - so I might still have some kind of compromise
going on.

Anyway thanks for the assistance everyone

Regards

Stefan



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