Re: [asterisk-users] Bug in main/bridge.c:ast_bridge_update_talker_src_video_mode

2017-08-28 Thread Richard Mudgett 
On Mon, Aug 28, 2017 at 6:35 PM, Richard Kenner  wrote:

> I've had two Asterisk crashes today that seem to be caused by errors
> where chan->tech_pvt is pointing to something that can't be deallocated
> and I think I see a reference count bug in the above function.
>
> It contains:
>
> if (data->chan_old_vsrc) {
> ast_channel_unref(data->chan_old_vsrc);
> }
>
> Shouldn't this also have:
>
> data->chan_old_vsrc = NULL;
>
> It seems to me that if it doesn't and the next condition also isn't
> true, then the next time this same code is executed, it'll decrement
> the reference count of the old channel again, which is wrong since it
> hasn't been decremented.
>

Yes, doing that would be a good thing.  What you point out does leave a
dangling
channel pointer in data->chan_old_vsrc if the pointer is not set to NULL.
Please
create an issue for the dangling pointer.  The patch needs to be done for
v13+.

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

2017-08-28 Thread Steve Edwards 

On Mon, 28 Aug 2017, Ira wrote:


The machine is an Intel Atom board...

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


My Intel(R) Atom(TM) CPU D525 seems to be quite happy running CentOS 
release 6.9 (Final) in 64 bit mode:


sedwards:~$ uname --hardware-platform --machine --processor
x86_64 x86_64 x86_64

sedwards:~$ getconf LONG_BIT
64

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[asterisk-users] Bug in main/bridge.c:ast_bridge_update_talker_src_video_mode

2017-08-28 Thread Richard Kenner 
I've had two Asterisk crashes today that seem to be caused by errors
where chan->tech_pvt is pointing to something that can't be deallocated
and I think I see a reference count bug in the above function.

It contains:

if (data->chan_old_vsrc) {
ast_channel_unref(data->chan_old_vsrc);
}

Shouldn't this also have:

data->chan_old_vsrc = NULL;

It seems to me that if it doesn't and the next condition also isn't
true, then the next time this same code is executed, it'll decrement
the reference count of the old channel again, which is wrong since it
hasn't been decremented.

What am I missing?

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

2017-08-28 Thread Doug Lytle 

On 08/28/2017 06:00 PM, Joseph Smith wrote:



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




Please ignore the noise, I need to slow down when I read.

Doug

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

2017-08-28 Thread Doug Lytle 

On 08/28/2017 06:00 PM, Joseph Smith wrote:



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




menuselect => Compiler Flags => Better Backtraces

Doug

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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 
 on behalf of Richard Mudgett 

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 
> 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 an average of 25 references to the 
format

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

2017-08-28 Thread Greg Woods 
On Mon, Aug 28, 2017 at 1:28 PM, Ira  wrote:

> Hello Asterisk,
>
> 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.
>

I run an asterisk server on Fedora 26. It's easy to install asterisk from
packages, so you don't have to go through the hassle of building it. Just
"dnf install asterisk".

The only problem is that Fedora does not package some of the hardware
drivers (e.g. dahdi) because they are not free software. I did have to
build the dahdi kernel module, but of course that depends on what hardware
you've got. If you are only using network (e.g. SIP or IAX), you should be
able to run completely from packages.


>
> Is the latest Fedora a good choice for an Asterisk box or should
> I try something else.


How critical is your need for that particular version of asterisk?  The
current packaged version for Fedora is 13.9.1

The usual warning about Fedora is that each version is only supported for
about 13 months. After that you have to upgrade or run without support
(dangerous, because that also means no security fixes).

The machine is an Intel Atom board with a
> Digium PCI analog board for my one last analog line.
>

That may or may not require compliation of the driver. I have a "Digium PCI
analog board" that uses the dahdi driver, and I did have to compile that
(see previous thread on this list regarding that).



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

Mine isn't. It is identified as a "Wildcard TDM410P", and it works fine on
my 64-bit OS.


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

Of course, if you install from packages with dnf, it will install all the
necessary dependencies.

My asterisk box is also a Samba server, again running from packages.

--Greg
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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] What version of Linux?

2017-08-28 Thread Ira 
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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Re: [asterisk-users] ERROR during high volume MoH dialplan

2017-08-28 Thread Richard Mudgett 
On Mon, Aug 28, 2017 at 1:04 PM, Joseph Smith 
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 an average of 25
references to the format per channel.  This is a simplistic calculation as
there are going to be some references that have nothing to do with a call.
The number of base references would depend upon which codec is involved.

* There is no user configurable option to change the excessive ref count
trigger value.  However, you could change the EXCESSIVE_REF_COUNT define
value in the main/astobj2.c file and recompile.

Richard
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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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Re: [asterisk-users] Outbound Calls via Proxy to use Call ID from registration

2017-08-28 Thread Joshua Colp 
On Mon, Aug 28, 2017, at 05:45 AM, Benoit Panizzon wrote:
> Hello List
> 
> > I work at an SIP Provider and we have added and SBC in front of our
> > Voice Switch to protect it.
> 
> Well using two peers for incomming and outgoing calls solve the
> previous issue.
> 
> Now I have a new one.
> 
> The SBC in use needs to match incomming calls from the asterisk with
> the call id used in the registration.
> 
> We have tested this with a couple of PBX, which do use the call ID used
> during registration automatically for outbound invites.
> 
> Not so my asterisk server.
> 
> So I assumed that when I refer to a 'peer' definition in the register
> statement, I could make asterisk understand, that the registration and
> outgoing peers belong together and then use the same call ID.

Can you define what exactly you mean by call id? If you are referring to
the Call-ID SIP header that's not how it works. It's unique within a
dialog and not reused like that[1][2].

[1] https://tools.ietf.org/html/rfc3261#page-37
[2] https://tools.ietf.org/html/rfc3261#section-20.8

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Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org

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[asterisk-users] Outbound Calls via Proxy to use Call ID from registration

2017-08-28 Thread Benoit Panizzon 
Hello List

> I work at an SIP Provider and we have added and SBC in front of our
> Voice Switch to protect it.

Well using two peers for incomming and outgoing calls solve the
previous issue.

Now I have a new one.

The SBC in use needs to match incomming calls from the asterisk with
the call id used in the registration.

We have tested this with a couple of PBX, which do use the call ID used
during registration automatically for outbound invites.

Not so my asterisk server.

So I assumed that when I refer to a 'peer' definition in the register
statement, I could make asterisk understand, that the registration and
outgoing peers belong together and then use the same call ID.

But how do I refer to a peer in the registration statement?

I did try different variants of:

register => sip-user@sip-outbound

[sip-outbound]
username=sip-user
secret=go-fishing
type=peer
host=asterisk-pbs.example.com
outboundproxy=ip.address.of.proxy
insecure=invite,port
qualify=yes
dtfmmode=auto
canreinvite=yes
context=from-sip
nat=no
t38pt_udptl=yes

But that is not matching.

If course if I do:

register=>sip-u...@asterisk-pbs.example.com:go-fish...@ip.address.of.proxy

registration is successfull, but invites sent via dial
sip://callerid@sip-outbound do not match the call ID used during
registration.

Anyone a hint how to make asterisk properly use the call ID of the
registration?

-Benoît Panizzon-
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