from:"Joseph Smith"

[asterisk-users] Connect Two Existing Channels and Stop Listening

2017-10-02 Thread Joseph Smith

Hello all,

In my scenario I have two channels connected to Asterisk and in a stasis app.

I can put them both in a bridge and audio between them works as expected.  
However, I would like to free up the resource and no longer have Asterisk 
involved in the call if possible.


I'm currently playing with "redirect" with and without creating the bridge but 
not have any success.


Do you have advice on how to accomplish this?


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

2017-09-05 Thread Joseph Smith

Thank you for the response Mike,

I did run into a CDR bottleneck as well and have already disabled it,

> module show like cdr
Module Description  Use 
Count  Status  Support Level
0 modules loaded

# grep enable= /etc/asterisk/cdr.conf
enable=no

At this point I'm really just not sure what the current bottleneck is and how 
to prevent the tasks for pooling.  I expected that the CPU would cap out before 
this occurred.  I do feel like there must be something I'm missing but just 
can't to it.

Any further suggestions are very welcome.

Thanks
Joseph

From: asterisk-users-boun...@lists.digium.com 
<asterisk-users-boun...@lists.digium.com> on behalf of Mike 
<mich...@virtutel.ca>
Sent: Friday, September 1, 2017 3:46 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] ERROR during high volume MoH dialplan

I had that problem before – I believe “task processor queue reached 500 
scheduled tasks” crashing means your CDR records (queue) are being written as 
the call ends, and if you had many thousands of entries being written to disk 
it crashes asterisk (each ring to one phone is an entry, so it goes up fast – 
for example 10 busy phones, with a between-ring delay of 1 second means every 
second there are 10 entries being put in memory)

I was using a MySQL CDR, but I had left the “CSV” type of CDR on. I 
removed/disabled the CSV CDR module, kept on the SQL CDR only and things have 
been working fine ever since.

Mike

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joseph Smith
Sent: September 1, 2017 16:41
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] ERROR during high volume MoH dialplan

Thanks for the suggestion Tony,

I installed each codec for MoH, core sounds, and extra sound packages.  
Unfortunately the tests produce the same results.

[Sep  1 20:36:45] ERROR[10081][C-7fe5]: frame.c:343 ast_frdup: FRACK!, 
Failed assertion Excessive refcount 10 reached on ao2 object 0x20380b0 (

continuously for a while followed by a

[Sep  1 20:36:46] WARNING[7761][C-770d]: taskprocessor.c:888 
taskprocessor_push: The 'subp:PJSIP/sipp-0020' task processor queue reached 
500 scheduled tasks.

Then this time Asterisk actually crashed. :(

From: asterisk-users-boun...@lists.digium.com 
<asterisk-users-boun...@lists.digium.com> on behalf of Tony Mountifield 
<t...@softins.co.uk>
Sent: Friday, September 1, 2017 11:01 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] ERROR during high volume MoH dialplan

In article 
<cy4pr2201mb14643c2177c953fa27ac9e2ba8...@cy4pr2201mb1464.namprd22.prod.outlook.com>,
Joseph Smith <warlock1...@hotmail.com> wrote:
>
> Thanks for the feedback.
>
> I do agree with having multiple smaller servers.  When I was first approached 
> with this task I mentioned as much.
> However, the current desire is to work with already existing hardware.  That 
> is out of my hands at the moment unless it
> just can't be done.  I will explore Freeswitch a bit soon to compare it as 
> well.
>
>
> I am struggling to find what the bottle neck is in this scenario.  Does 
> anyone have any advice on what that could be or
> on steps to discover it?   Do you think that tasks are pooling up because of 
> transcoding?  If so would it help to change
> the codec that is being used?  I am not sure about the MoH but the audio 
> files I am using are gsm.

You will find it less taxing on the server if you have MoH files and sounds 
files
available in all the possible native formats. Then Asterisk can use the 
appropriate
one for the channel without transcoding.

On my systems I have MoH and sounds installed in wav, ulaw, alaw, gsm and g729.

They will also sound better than transcoding from the gsm versions.

Cheers
Tony
--
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Work: t...@softins.co.uk - http://www.softins.co.uk
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www.softins.co.uk
Welcome. Software Insight Ltd is a small but expert company specialising in 
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Re: [asterisk-users] ERROR during high volume MoH dialplan

2017-09-01 Thread Joseph Smith

Thanks for the suggestion Tony,

I installed each codec for MoH, core sounds, and extra sound packages.  
Unfortunately the tests produce the same results.

[Sep  1 20:36:45] ERROR[10081][C-7fe5]: frame.c:343 ast_frdup: FRACK!, 
Failed assertion Excessive refcount 10 reached on ao2 object 0x20380b0 (

continuously for a while followed by a

[Sep  1 20:36:46] WARNING[7761][C-770d]: taskprocessor.c:888 
taskprocessor_push: The 'subp:PJSIP/sipp-0020' task processor queue reached 
500 scheduled tasks.

Then this time Asterisk actually crashed. :(

From: asterisk-users-boun...@lists.digium.com 
<asterisk-users-boun...@lists.digium.com> on behalf of Tony Mountifield 
<t...@softins.co.uk>
Sent: Friday, September 1, 2017 11:01 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] ERROR during high volume MoH dialplan

In article 
<cy4pr2201mb14643c2177c953fa27ac9e2ba8...@cy4pr2201mb1464.namprd22.prod.outlook.com>,
Joseph Smith <warlock1...@hotmail.com> wrote:
>
> Thanks for the feedback.
>
> I do agree with having multiple smaller servers.  When I was first approached 
> with this task I mentioned as much.
> However, the current desire is to work with already existing hardware.  That 
> is out of my hands at the moment unless it
> just can't be done.  I will explore Freeswitch a bit soon to compare it as 
> well.
>
>
> I am struggling to find what the bottle neck is in this scenario.  Does 
> anyone have any advice on what that could be or
> on steps to discover it?   Do you think that tasks are pooling up because of 
> transcoding?  If so would it help to change
> the codec that is being used?  I am not sure about the MoH but the audio 
> files I am using are gsm.

You will find it less taxing on the server if you have MoH files and sounds 
files
available in all the possible native formats. Then Asterisk can use the 
appropriate
one for the channel without transcoding.

On my systems I have MoH and sounds installed in wav, ulaw, alaw, gsm and g729.

They will also sound better than transcoding from the gsm versions.

Cheers
Tony
--
Tony Mountifield
Work: t...@softins.co.uk - http://www.softins.co.uk
Software Insight - Welcome<http://www.softins.co.uk/>
www.softins.co.uk
Welcome. Software Insight Ltd is a small but expert company specialising in 
software and systems development and systems administration. We pride ourselves 
in ...

Play: t...@mountifield.org - http://tony.mountifield.org
[http://tony.mountifield.org/images/tony2.jpg]<http://tony.mountifield.org/>

Tony Mountifield's Home Page<http://tony.mountifield.org/>
tony.mountifield.org
Tony Mountifield's Home Page. This page is still under construction (despite 
having been started a long time ago!) It will grow as I think of more things to 
put in ...

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

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

2017-09-01 Thread Joseph Smith

Thanks for the feedback.

I do agree with having multiple smaller servers.  When I was first approached 
with this task I mentioned as much.  However, the current desire is to work 
with already existing hardware.  That is out of my hands at the moment unless 
it just can't be done.  I will explore Freeswitch a bit soon to compare it as 
well.

I am struggling to find what the bottle neck is in this scenario.  Does anyone 
have any advice on what that could be or on steps to discover it?   Do you 
think that tasks are pooling up because of transcoding?  If so would it help to 
change the codec that is being used?  I am not sure about the MoH but the audio 
files I am using are gsm.

Thanks

Joseph

From: asterisk-users-boun...@lists.digium.com 
<asterisk-users-boun...@lists.digium.com> on behalf of Pete Mundy 
<p...@fiberphone.co.nz>
Sent: Thursday, August 31, 2017 2:42 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] ERROR during high volume MoH dialplan

>> On Thu, 31 Aug 2017, Joseph Smith wrote:
>>
>> So I am looking for a better way to allow several thousand callers to listen 
>> to this IVR menu at the same time.

> On 1/09/2017, at 7:10 AM, Steve Edwards <asterisk@sedwards.com> wrote:
>
> I'm thinking multiple hosts.
>
> I'm not a fan of 4,000 eggs in one basket.

+1 for horizontal scaling as the best solution in this situation.

Pete

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

2017-08-31 Thread Joseph Smith

It is meant to simulate simultaneous calls on an IVR.  I have also tested with 
a separate set of audio files closer to what the actual IVR menu.  This 
produced the same result.


I apologize for not clearly stating the use case up front.  I will try to give 
a bit more detail on that now.


I have an IVR menu and submenu that users may dial into. I initially tested 
with the IVR audio files.  When I began experiencing this issue I used MoH as 
an attempt to narrow down the problem to the simplest dialplan possible.


If I continue my test at this volume or a higher volume, I begin to get errors 
about reaching the maximum queue size for that particular taskprocessor.  
Since, these error proceeded that I thought that they may be the key to 
preventing the queue from maxing out.


It sounds like Richard is saying that these refcount logs may not actually be 
errors and can be ignored in this scenario.  If that is the case then is there 
anything that can be done about the task processor queue size?  Is that simply 
a side effect of having so many callers listening to the IVR at the same time?

pjsip.conf is currently setup with a trunk allowing incoming calls from a 
specific IP.  This is the task processor that is maxing out.

So I am looking for a better way to allow several thousand callers to listen to 
this IVR menu at the same time.

Thank you for the feedback thus far.

Any info and advice is helpful.

Thanks
Joseph




From: asterisk-users-boun...@lists.digium.com 
<asterisk-users-boun...@lists.digium.com> on behalf of Antony Stone 
<antony.st...@asterisk.open.source.it>
Sent: Thursday, August 31, 2017 11:58 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] ERROR during high volume MoH dialplan

On Thursday 31 August 2017 at 18:15:54, Joseph Smith wrote:

> I was hoping Asterisk would handle more than 4k simultaneous calls.

I know from experience that Asterisk can handle more than 4k simultaneous
calls, however it's an extreme case to have all of them playing music on hold.

I think that if you tested 4k simultaneous calls with standard media streams
on the majority of them, you would not experience the problem.

Is this a real problem for you - that Asterisk can't manage 4k MoH sessions
simultaneously, even though it can manage 4k standard phone calls?


Antony.

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...




New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started
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Re: [asterisk-users] ERROR during high volume MoH dialplan

2017-08-31 Thread Joseph Smith

Is there any more information I can provide to give insight to these errors?

Any further advice on avoiding these during high call volume?


I was hoping Asterisk would handle more than 4k simultaneous calls.

Thanks

Joseph



From: asterisk-users-boun...@lists.digium.com 
<asterisk-users-boun...@lists.digium.com> on behalf of Joseph Smith 
<warlock1...@hotmail.com>
Sent: Monday, August 28, 2017 5:00 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] ERROR during high volume MoH dialplan


Hi Richard,

Thank you for the reply


Correct, I did mean 13.15.


I set no optimize and better backtrace through "make menuselect" and the output 
is now


[Aug 28 21:41:16] ERROR[17171][C-392d]: frame.c:343 ast_frdup: FRACK!, 
Failed assertion Excessive refcount 10 reached on ao2 object 0x21962b0 (0)

Got 26 backtrace records

#0: [0x61923f] main/utils.c:2475 __ast_assert_failed() (0x6191bb+84)

#1: [0x45ffc9] main/astobj2.c:543 __ao2_ref() (0x45fc3d+38C)

#2: [0x5320ce] main/frame.c:345 ast_frdup() (0x531e4c+282)

#3: [0x531a99] main/frame.c:196 ast_frisolate() (0x531a76+23)

#4: [0x60be51] main/translate.c:459 ast_trans_frameout() (0x60bd6e+E3)

#5: [0x60be75] main/translate.c:464 default_frameout()

#6: [0x60c46a] main/translate.c:579 ast_translate() (0x60c192+2D8)

#7: [0x4c0bf1] main/channel.c:5290 ast_write() (0x4bfb3e+10B3)

#8: [0x7fdef8345486] res/res_musiconhold.c:455 moh_files_generator()

#9: [0x4ba212] main/channel.c:3014 generator_force()

#10: [0x4bc23d] main/channel.c:3872 __ast_read()

#11: [0x4be29b] main/channel.c:4399 ast_read() (0x4be27e+1D)

#12: [0x4b6312] main/channel.c:1568 ast_safe_sleep_conditional() (0x4b6229+E9)

#13: [0x4b64c9] main/channel.c:1613 ast_safe_sleep() (0x4b64a1+28)

#14: [0x7fdef8346caa] res/res_musiconhold.c:834 play_moh_exec()

#15: [0x5970a3] main/pbx_app.c:491 pbx_exec() (0x596f87+11C)

#16: [0x582edf] main/pbx.c:2923 pbx_extension_helper()

#17: [0x586c30] main/pbx.c:4155 ast_spawn_extension() (0x586bcc+64)

#18: [0x5878bb] main/pbx.c:4328 __ast_pbx_run()

#19: [0x589061] main/pbx.c:4651 pbx_thread()

#20: [0x61624e] main/utils.c:1233 dummy_start()



* What codecs are you using in this setup?
In pjsip.conf I have disallow=all and allow=ulaw.  If I can provide more 
information or a better response to this question please guide me on how to do 
that.


Thanks
Joseph



From: asterisk-users-boun...@lists.digium.com 
<asterisk-users-boun...@lists.digium.com> on behalf of Richard Mudgett 
<rmudg...@digium.com>
Sent: Monday, August 28, 2017 2:17 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] ERROR during high volume MoH dialplan



On Mon, Aug 28, 2017 at 1:04 PM, Joseph Smith 
<warlock1...@hotmail.com<mailto:warlock1...@hotmail.com>> wrote:

Hello,

I've recently setup a small load test against an instance of Asterisks.  I've 
tested on asterisk 13.5 and 14.6 with the same results.

I think you mean 13.15.0 as the excessive ref count trap is not in 13.5.0.

I am using PJSIP.  My dial plan is,

[test]

exten => 1001,1,Answer

exten => 1001,n,MusicOnHold(15)

exten => 1001,n,Hangup

I am using SIPP to test.  I can share XML if desired but it simply waits on the 
line while music plays for 8 seconds.  I used sippycup to generate it with the 
following steps in the yaml file.


steps:

  - invite

  - wait_for_answer

  - ack_answer

  - sleep 8

  - send_bye

At around 500 calls per second I begin to see the following ERRORs,


[Aug 28 17:46:14] ERROR[26150][C-5594]: frame.c:343 ast_frdup: Excessive 
refcount 10 reached on ao2 object 0x26bffc0

[Aug 28 17:46:14] ERROR[26150][C-5594]: frame.c:343 ast_frdup: FRACK!, 
Failed assertion Excessive refcount 10 reached on ao2 object 0x26bffc0 (0)

Got 19 backtrace records

#0: [0x45d229] /usr/sbin/asterisk(__ao2_ref+0x1a9) [0x45d229]

#1: [0x526ce6] /usr/sbin/asterisk(ast_frdup+0x116) [0x526ce6]

#2: [0x5fa616] /usr/sbin/asterisk(ast_translate+0x306) [0x5fa616]

#3: [0x4bf16b] /usr/sbin/asterisk(ast_write+0x104b) [0x4bf16b]

#4: [0x7efeb578230b] /usr/lib/asterisk/modules/res_musiconhold.so(+0x430b) 
[0x7efeb578230b]

#5: [0x4b5b52] /usr/sbin/asterisk() [0x4b5b52]

#6: [0x4c259c] /usr/sbin/asterisk() [0x4c259c]

#7: [0x4c4a45] /usr/sbin/asterisk() [0x4c4a45]

#8: [0x7efeb578478d] /usr/lib/asterisk/modules/res_musiconhold.so(+0x678d) 
[0x7efeb578478d]

#9: [0x58ec79] /usr/sbin/asterisk(pbx_exec+0xb9) [0x58ec79]

#10: [0x582e84] /usr/sbin/asterisk() [0x582e84]

#11: [0x584e7c] /usr/sbin/asterisk() [0x584e7c]

#12: [0x5863fb] /usr/sbin/asterisk() [0x5863fb]

#13: [0x60002a] /usr/sbin/asterisk() [0x60002a]

This inline backtrace would be more useful if you had BETTER_BACKTRACES enabled.



I've also seen similar behavior when using playback instead of MusicOnHold.  
CPU usage gets around 50%.  Can any

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

2017-08-28 Thread Joseph Smith

Hi Richard,

Thank you for the reply


Correct, I did mean 13.15.


I set no optimize and better backtrace through "make menuselect" and the output 
is now


[Aug 28 21:41:16] ERROR[17171][C-392d]: frame.c:343 ast_frdup: FRACK!, 
Failed assertion Excessive refcount 10 reached on ao2 object 0x21962b0 (0)

Got 26 backtrace records

#0: [0x61923f] main/utils.c:2475 __ast_assert_failed() (0x6191bb+84)

#1: [0x45ffc9] main/astobj2.c:543 __ao2_ref() (0x45fc3d+38C)

#2: [0x5320ce] main/frame.c:345 ast_frdup() (0x531e4c+282)

#3: [0x531a99] main/frame.c:196 ast_frisolate() (0x531a76+23)

#4: [0x60be51] main/translate.c:459 ast_trans_frameout() (0x60bd6e+E3)

#5: [0x60be75] main/translate.c:464 default_frameout()

#6: [0x60c46a] main/translate.c:579 ast_translate() (0x60c192+2D8)

#7: [0x4c0bf1] main/channel.c:5290 ast_write() (0x4bfb3e+10B3)

#8: [0x7fdef8345486] res/res_musiconhold.c:455 moh_files_generator()

#9: [0x4ba212] main/channel.c:3014 generator_force()

#10: [0x4bc23d] main/channel.c:3872 __ast_read()

#11: [0x4be29b] main/channel.c:4399 ast_read() (0x4be27e+1D)

#12: [0x4b6312] main/channel.c:1568 ast_safe_sleep_conditional() (0x4b6229+E9)

#13: [0x4b64c9] main/channel.c:1613 ast_safe_sleep() (0x4b64a1+28)

#14: [0x7fdef8346caa] res/res_musiconhold.c:834 play_moh_exec()

#15: [0x5970a3] main/pbx_app.c:491 pbx_exec() (0x596f87+11C)

#16: [0x582edf] main/pbx.c:2923 pbx_extension_helper()

#17: [0x586c30] main/pbx.c:4155 ast_spawn_extension() (0x586bcc+64)

#18: [0x5878bb] main/pbx.c:4328 __ast_pbx_run()

#19: [0x589061] main/pbx.c:4651 pbx_thread()

#20: [0x61624e] main/utils.c:1233 dummy_start()



* What codecs are you using in this setup?
In pjsip.conf I have disallow=all and allow=ulaw.  If I can provide more 
information or a better response to this question please guide me on how to do 
that.


Thanks
Joseph



From: asterisk-users-boun...@lists.digium.com 
<asterisk-users-boun...@lists.digium.com> on behalf of Richard Mudgett 
<rmudg...@digium.com>
Sent: Monday, August 28, 2017 2:17 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] ERROR during high volume MoH dialplan



On Mon, Aug 28, 2017 at 1:04 PM, Joseph Smith 
<warlock1...@hotmail.com<mailto:warlock1...@hotmail.com>> wrote:

Hello,

I've recently setup a small load test against an instance of Asterisks.  I've 
tested on asterisk 13.5 and 14.6 with the same results.

I think you mean 13.15.0 as the excessive ref count trap is not in 13.5.0.

I am using PJSIP.  My dial plan is,

[test]

exten => 1001,1,Answer

exten => 1001,n,MusicOnHold(15)

exten => 1001,n,Hangup

I am using SIPP to test.  I can share XML if desired but it simply waits on the 
line while music plays for 8 seconds.  I used sippycup to generate it with the 
following steps in the yaml file.


steps:

  - invite

  - wait_for_answer

  - ack_answer

  - sleep 8

  - send_bye

At around 500 calls per second I begin to see the following ERRORs,


[Aug 28 17:46:14] ERROR[26150][C-5594]: frame.c:343 ast_frdup: Excessive 
refcount 10 reached on ao2 object 0x26bffc0

[Aug 28 17:46:14] ERROR[26150][C-5594]: frame.c:343 ast_frdup: FRACK!, 
Failed assertion Excessive refcount 10 reached on ao2 object 0x26bffc0 (0)

Got 19 backtrace records

#0: [0x45d229] /usr/sbin/asterisk(__ao2_ref+0x1a9) [0x45d229]

#1: [0x526ce6] /usr/sbin/asterisk(ast_frdup+0x116) [0x526ce6]

#2: [0x5fa616] /usr/sbin/asterisk(ast_translate+0x306) [0x5fa616]

#3: [0x4bf16b] /usr/sbin/asterisk(ast_write+0x104b) [0x4bf16b]

#4: [0x7efeb578230b] /usr/lib/asterisk/modules/res_musiconhold.so(+0x430b) 
[0x7efeb578230b]

#5: [0x4b5b52] /usr/sbin/asterisk() [0x4b5b52]

#6: [0x4c259c] /usr/sbin/asterisk() [0x4c259c]

#7: [0x4c4a45] /usr/sbin/asterisk() [0x4c4a45]

#8: [0x7efeb578478d] /usr/lib/asterisk/modules/res_musiconhold.so(+0x678d) 
[0x7efeb578478d]

#9: [0x58ec79] /usr/sbin/asterisk(pbx_exec+0xb9) [0x58ec79]

#10: [0x582e84] /usr/sbin/asterisk() [0x582e84]

#11: [0x584e7c] /usr/sbin/asterisk() [0x584e7c]

#12: [0x5863fb] /usr/sbin/asterisk() [0x5863fb]

#13: [0x60002a] /usr/sbin/asterisk() [0x60002a]

This inline backtrace would be more useful if you had BETTER_BACKTRACES enabled.



I've also seen similar behavior when using playback instead of MusicOnHold.  
CPU usage gets around 50%.  Can anyone enlighten me on the meaning and cause of 
the error?  Is there some steps (config etc) that can be taken to alleviate the 
issue?

This particular FRACK is meant to help find ao2 object reference leaks.  It 
acts as an early warning for excessive references to any particular ao2 object 
used in the code.  The FRACK itself is benign.  Based upon the inline backtrace 
the ao2 object is likely to be a codec format.

* What codecs are you using in this setup?

* With 500 calls/sec and the calls lasting 8 seconds that comes to 4000 active 
channels.  Hitting the FRACK would result in

Re: [asterisk-users] What version of Linux?

2017-08-28 Thread Joseph Smith

Hello Ira,

 I recently installed on AMI to test out a bit before moving to physical 
hardware.  I had to install a number of packages to get it working their. You 
might be having a similar issue.  I installed the following packages before 
getting a completed configure and make.


** Caused error during configure

gcc-c++
ncurses-devel
libuuid-devel
libxml2-devel
sqlite-devel
patch
jansson-devel

**caused error duing make
openssl-devel
m2crypto


If you still have problems, paste some of the error generated by configure for 
reference


Good Luck
Joseph


From: asterisk-users-boun...@lists.digium.com 
 on behalf of Ira 

Sent: Monday, August 28, 2017 2:28 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] What version of Linux?

Hello Asterisk,

I've been running CentOS since 2006 or so and support for the 32
bit version recently ended. CentOS no longer offers a 32 bit
version so I thought I'd try Fedora 26 as they have 32 bit and
support. Got it installed, then downloaded Asterisk 14.6.0 but
can't seem to get it built. The configure script fails with some
error about CPP not working correctly? I did discover that
kernel-devel was not installed so I fixed that but I'm still
stuck.

Is the latest Fedora a good choice for an Asterisk box or should
I try something else. The machine is an Intel Atom board with a
Digium PCI analog board for my one last analog line.

I believe the board is limited to a 32 bit OS.

So two questions, is Fedora a good choice and if not, what
should I use for a machine running only Asterisk and Samba?

Is there a list of dependencies I need to install before
Asterisk will compile?

Thanks, Ira


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[asterisk-users] ERROR during high volume MoH dialplan

2017-08-28 Thread Joseph Smith

Hello,

I've recently setup a small load test against an instance of Asterisks.  I've 
tested on asterisk 13.5 and 14.6 with the same results.

I am using PJSIP.  My dial plan is,

[test]

exten => 1001,1,Answer

exten => 1001,n,MusicOnHold(15)

exten => 1001,n,Hangup

I am using SIPP to test.  I can share XML if desired but it simply waits on the 
line while music plays for 8 seconds.  I used sippycup to generate it with the 
following steps in the yaml file.


steps:

  - invite

  - wait_for_answer

  - ack_answer

  - sleep 8

  - send_bye

At around 500 calls per second I begin to see the following ERRORs,


[Aug 28 17:46:14] ERROR[26150][C-5594]: frame.c:343 ast_frdup: Excessive 
refcount 10 reached on ao2 object 0x26bffc0

[Aug 28 17:46:14] ERROR[26150][C-5594]: frame.c:343 ast_frdup: FRACK!, 
Failed assertion Excessive refcount 10 reached on ao2 object 0x26bffc0 (0)

Got 19 backtrace records

#0: [0x45d229] /usr/sbin/asterisk(__ao2_ref+0x1a9) [0x45d229]

#1: [0x526ce6] /usr/sbin/asterisk(ast_frdup+0x116) [0x526ce6]

#2: [0x5fa616] /usr/sbin/asterisk(ast_translate+0x306) [0x5fa616]

#3: [0x4bf16b] /usr/sbin/asterisk(ast_write+0x104b) [0x4bf16b]

#4: [0x7efeb578230b] /usr/lib/asterisk/modules/res_musiconhold.so(+0x430b) 
[0x7efeb578230b]

#5: [0x4b5b52] /usr/sbin/asterisk() [0x4b5b52]

#6: [0x4c259c] /usr/sbin/asterisk() [0x4c259c]

#7: [0x4c4a45] /usr/sbin/asterisk() [0x4c4a45]

#8: [0x7efeb578478d] /usr/lib/asterisk/modules/res_musiconhold.so(+0x678d) 
[0x7efeb578478d]

#9: [0x58ec79] /usr/sbin/asterisk(pbx_exec+0xb9) [0x58ec79]

#10: [0x582e84] /usr/sbin/asterisk() [0x582e84]

#11: [0x584e7c] /usr/sbin/asterisk() [0x584e7c]

#12: [0x5863fb] /usr/sbin/asterisk() [0x5863fb]

#13: [0x60002a] /usr/sbin/asterisk() [0x60002a]


I've also seen similar behavior when using playback instead of MusicOnHold.  
CPU usage gets around 50%.  Can anyone enlighten me on the meaning and cause of 
the error?  Is there some steps (config etc) that can be taken to alleviate the 
issue?

Thanks
Joseph




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

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[asterisk-users] Connect Two Existing Channels and Stop Listening

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

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

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

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

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

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

Re: [asterisk-users] What version of Linux?

[asterisk-users] ERROR during high volume MoH dialplan

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