Re: [asterisk-dev] Asterisk 13.17.1 Crash on ConfBridge - NetGen ATA

[Bryant Zimmerman]

George
  
 Thank you we will get 13.18.4 on the test box ASAP, and thank you for the 
wiki link so we can pull together additional information to resolve this 
issue. I have upgraded to 13.18.4 and the system is not crashing now. Is 
there any specific item that may point to why the issue was occurring?

Thanks

Bryant Zimmerman (ZK Tech Inc.)
616-855-1030 Ext. 2003
  


 From: "George Joseph" <gjos...@digium.com>
Sent: Tuesday, December 19, 2017 2:59 PM
To: brya...@zktech.com, "Asterisk Developers Mailing List" 
<asterisk-dev@lists.digium.com>
Subject: Re: [asterisk-dev] Asterisk 13.17.1 Crash on ConfBridge - NetGen 
ATA   
 On Tue, Dec 19, 2017 at 12:45 PM, Bryant Zimmerman 
<brya...@zktech.com> wrote:   We are having an issue with asterisk 13.17.1 
dumping when a call from a NetGen Smart ATA drops into a confbridge
 The call props up and then things just go wrong. I have talked with the 
support guys at NetGen and they have requested I work start with the 
asterisk dev group so we can figure out what is causing this issue and why 
asterisk is dumping. They are willing to fix anything from their end but we 
have not been able to figure out what in their rtp stream is triggering 
this.  Their ATA's seem to work out side of the confbridge without issues 
so far. Any ideas are appreciated. The asterisk dump is by far my biggest 
concern. 
  
 Below is the first part of the dump Backtrack. I have attached a copy of 
the complete Backtrack. I need to know what more would be needed to get to 
the bottom of this issue. As it stands now the NetGen Smart ATA will cause 
asterisk 13 to crash if placed into a confbridge.  
http://www.netgencommunications.com/
 The support guy said we could contact them at 
supp...@netgencommunications.com
   
 There's not a whole lot of info in this backtrace for us to really know 
what's going on but can you try with 13.18.4?  There have been recent crash 
fixes that may help.
  
 If 13.18.4 doesn't help, recompiling with debugging turned on, re-creating 
the issue, then following the wiki instructions to get a backtrace will 
help us figure out what's up.
 https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace
  
  
  
  
  
  
  
   
-- Executing [s@Core_ConfBridge_Basic:11] 
ConfBridge("PJSIP/6162480909.351-", 
"6162480909.~Promo~GA1,,,sample_user_menu") in new stack
   > 0x7f3ff800b4c0 -- Probation passed - setting RTP source address to 
192.168.209.194:10020
-- Channel CBAnn/6162480909.~Promo~GA1-;2 joined 'softmix' 
base-bridge <4a00cdad-91cb-4924-8abe-8dc9cad08f10>
--  Playing 'conf-onlyperson.ulaw' 
(language 'en')
UBNTU-ROSSI-GUEST*CLI> *** Error in `/usr/sbin/asterisk': malloc(): memory 
corruption: 0x7f3fac00c220 ***
=== Backtrace: =
/lib/x86_64-linux-gnu/libc.so.6(+0x777e5)[0x7f402147b7e5]
/lib/x86_64-linux-gnu/libc.so.6(+0x8213e)[0x7f402148613e]
/lib/x86_64-linux-gnu/libc.so.6(__libc_malloc+0x54)[0x7f4021488184]
/usr/sbin/asterisk(ast_json_malloc+0xa)[0x52a23a]
/usr/lib/x86_64-linux-gnu/libjansson.so.4(json_object+0xb)[0x7f40227ab7bb]
/usr/lib/x86_64-linux-gnu/libjansson.so.4(+0x6505)[0x7f40227aa505]
/usr/lib/x86_64-linux-gnu/libjansson.so.4(json_vpack_ex+0x99)[0x7f40227aaa09
]
/usr/sbin/asterisk(ast_json_vpack+0x34)[0x52b6a4]
/usr/sbin/asterisk(ast_json_pack+0xa1)[0x52b7c1]
/usr/lib/asterisk/modules/res_rtp_asterisk.so(+0x10df3)[0x7f3f90994df3]
/usr/lib/asterisk/modules/res_rtp_asterisk.so(+0x11a99)[0x7f3f90995a99]
/usr/lib/asterisk/modules/res_rtp_asterisk.so(+0x13bcb)[0x7f3f90997bcb]
/usr/sbin/asterisk(ast_rtp_instance_read+0x36)[0x588076]
/usr/lib/asterisk/modules/chan_pjsip.so(+0x8cd7)[0x7f3f7c57ccd7]
/usr/sbin/asterisk[0x4bc042]
/usr/sbin/asterisk[0x50f3f1]
/usr/sbin/asterisk(ast_stream_and_wait+0x56)[0x511bbe]
/usr/lib/asterisk/modules/app_confbridge.so(+0xb716)[0x7f3f915d7716]
/usr/lib/asterisk/modules/app_confbridge.so(+0xd5f6)[0x7f3f915d95f6]
/usr/sbin/asterisk(pbx_exec+0xbd)[0x579155]
/usr/sbin/asterisk[0x56e0c3]
/usr/sbin/asterisk(ast_spawn_extension+0x18)[0x56feb8]
/usr/lib/asterisk/modules/app_macro.so(+0x2c02)[0x7f3f6f8a6c02]
/usr/sbin/asterisk(pbx_exec+0xbd)[0x579155]
/usr/sbin/asterisk[0x56e0c3]
/usr/sbin/asterisk[0x5703d1]
/usr/sbin/asterisk[0x57190b]
/usr/sbin/asterisk[0x5e45fd]
/lib/x86_64-linux-gnu/libpthread.so.0(+0x76ba)[0x7f4021f436ba]
/lib/x86_64-linux-gnu/libc.so.6(clone+0x6d)[0x7f402150b3dd]
  

Thanks

Bryant Zimmerman (ZK Tech Inc.)
616-855-1030 Ext. 2003

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--   George Joseph Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsv

Re: [asterisk-dev] One sip stack to rule them all....

[Bryant Zimmerman]

I would agree with this. We have tried to deploy pjsip several times over the 
last year with limited success.
 We have had nothing but issues with database real-time deployments. Tables not 
working from one 13.x release to another.
 Table builders sorcery failing out.

 Issues when there are multiple transports on varying networks were udp is not 
routed correctly through the asterisk servers. No matter the settings.

 Connectivity issues with varying success by carrier.

 Unexplained audio quality issues that don't occur on the same spec running 
chan_sip

 We want to move to pjsip but the functionality and stability have only proven 
out for limited applications.


 Bryant



 From: "Daniel Journo" 
Sent: Sunday, October 8, 2017 3:12 PM
To: "Asterisk Developers Mailing List" 
Subject: Re: [asterisk-dev] One sip stack to rule them all

> What is _also_ needed, however, is more use of PJSIP and reports of  specific 
> problems, and specific deficits of PJSIP so that the fear can be eased 
> before, at some point many years from now, chan_sip just doesn't work any 
> more.

There are a number of specific issues on issue tracker which still need 
addressing before more people will take it on properly. Some issues probably 
require a semi-major rethink and probably won't be dealt with for months.
Making chan_sip depreciated would leave Asterisk with no production grade sip 
stack that is officially being maintained.


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Re: [asterisk-dev] Park patch to silence slot number

[Bryant Zimmerman]

You can create dynamic lots with different contexts in the fly. You are 
actually parking at a context you just use an extension to identify access 
to the slot. If you have any identifying unique account info. Each lot is 
based on a parking lot context as a template. The lot name is dyanmically 
unique per lot. When you first create it. We use unified dialplan contexts 
for parking. We just augment our lots based on the account number for each 
client (prkAcctNumLotNum.SlotNym).  It is very flexible you can pass in all 
the parameters on the fly using a combination of special channel dynamic 
variables and options in the park and parked command. We even specify 
dynamic ring back contexts with parameters in them so ring backs can be 
directed correctly. And accounts can only ever pickup their lots.  We have 
customers with multiple parking lots all on the same account and the 
parking extensions are set in a database. The only limitation is you can't 
change the parking template context or shrink it once a lot is created 
without a restart of Asterisk. You can grow the number of parking slots 
dyanmically. 

 

The way we do this when you park directly on a slot the ring back time can 
be specified on a per slot basis as well. Customers love this.

 

Sent from my Windows 10 phone

 

From: David Cunningham
Sent: Tuesday, October 3, 2017 6:46 PM
To: Asterisk Developers Mailing List
Subject: Re: [asterisk-dev] Park patch to silence slot number

 Hi Byrant,

Thank you for the reply (and voicemail). Are you referring to the patch by
Igor Goncharovsky? We had issues because it requires a different context
for each overlapping parking slot, and our multi-tenant environment 
doesn't
use a different context per tenant. What he said is below. However, we've
just noticed that Asterisk 13 has an 's' option on the Park command to
silence the slot number, so sorry for the trouble!

"It is the how parking works in core of asterisk. When call parked,
asterisk automatically put extensions for call park and for taking a call
from park. I.e. when you want to park two calls at same extension and same
context asterisk unable to do it, because unable to insert second time 
same
extension in dialplan. I see no problem in using different extensions 
range
for all parking lots in same context of using different contexts for all
users"


On 3 October 2017 at 16:23, Bryant Zimmerman <brya...@zktech.com> wrote:

> You don't need a patch this is possible with the current tools if you 
keep
> track of parks pickups and ring backs. We use the current system to do
> dynamic parks all of the time in multi tenant environment . We create
> dynamic lots per tenant and address them per sub account. This allows 
for
> each tenent to have parking slots with any number even when others use 
the
> same.  I paid part of the bounty to get the original dynamic parking 
system
> working.
>
>
>
> Sent from my Windows 10 phone
>
>
>
> *From: *David Cunningham <dcunning...@voisonics.com>
> *Sent: *Monday, October 2, 2017 11:09 PM
> *To: *asterisk-dev@lists.digium.com
> *Subject: *[asterisk-dev] Park patch to silence slot number
>
>
> Hello,
>
> We'd like to get a patch written for Asterisk's Park command, so that 
with
> a given option it won't play the parking slot number. The idea is that 
we
> can allow multiple calls to be parked on the same apparent slot number 
by
> playing the apparent slot number ourselves, and parking calls on a
> different actual slot number. For example we'd play 701 but actually 
park
> the call on 123701.
>
> Does anyone know of someone who'd be willing to write this patch, and
> submit it for inclusion in Asterisk? We will of course pay a bounty. If 
I'm
> asking in the wrong place then apologies and please let me know the 
right
> one.
>
> Thanks in advance,
>
> --
> David Cunningham, Voisonics Limited
> http://voisonics.com/
> USA: +1 213 221 1092 <+1%20213-221-1092>
> Australia: +61 (0) 2 8063 9019 <+61%202%208063%209019>
>
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-- 
David Cunningham, Voisonics Limited
http://voisonics.com/
USA: +1 213 221 1092
Australia: +61 (0) 2 8063 9019
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Re: [asterisk-dev] Park patch to silence slot number

[Bryant Zimmerman]

You don't need a patch this is possible with the current tools if you keep 
track of parks pickups and ring backs. We use the current system to do 
dynamic parks all of the time in multi tenant environment . We create 
dynamic lots per tenant and address them per sub account. This allows for 
each tenent to have parking slots with any number even when others use the 
same.  I paid part of the bounty to get the original dynamic parking system 
working.

 

Sent from my Windows 10 phone

 

From: David Cunningham
Sent: Monday, October 2, 2017 11:09 PM
To: asterisk-dev@lists.digium.com
Subject: [asterisk-dev] Park patch to silence slot number

 Hello,

We'd like to get a patch written for Asterisk's Park command, so that with
a given option it won't play the parking slot number. The idea is that we
can allow multiple calls to be parked on the same apparent slot number by
playing the apparent slot number ourselves, and parking calls on a
different actual slot number. For example we'd play 701 but actually park
the call on 123701.

Does anyone know of someone who'd be willing to write this patch, and
submit it for inclusion in Asterisk? We will of course pay a bounty. If 
I'm
asking in the wrong place then apologies and please let me know the right
one.

Thanks in advance,

-- 
David Cunningham, Voisonics Limited
http://voisonics.com/
USA: +1 213 221 1092
Australia: +61 (0) 2 8063 9019
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Re: [asterisk-dev] sip Messages - No acces to variables set on the peer.- Solved

[Bryant Zimmerman]

I will just use sippeer to read the values I need.
  
 Thanks

Bryant 


   
  I am working on an application that uses sip messages to send sms.



   I am bumping into what I think may be  a bug.

   The message is being sent to the asterisk server from the extension, but 
I have no access to the setvar variables or any variables that would 
normally be on a standard channel such as accountcode set for the peer for 
processing in the dial plan. Is there any way to access these outside of a 
call during message handling?

   
Thanks

Bryant


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Re: [asterisk-dev] sip Messages - No acces to variables set on the peer.

[Bryant Zimmerman]

I am working on an application that uses sip messages to send sms.
  
 I am bumping into what I think may be  a bug.
 The message is being sent to the asterisk server from the extension, but I 
have no access to the setvar variables or any variables that would normally 
be on a standard channel such as accountcode set for the peer for 
processing in the dial plan. Is there any way to access these outside of a 
call during message handling?

Thanks

Bryant

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Re: [asterisk-dev] ARI versioning in 13 and 14

[Bryant Zimmerman]

+1 to option 2.

Thanks

Bryant Zimmerman (ZK Tech Inc.)
616-855-1030 Ext. 2003

 

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Re: [asterisk-dev] Issue AGI Get Full Variable

[Bryant Zimmerman]

I am calling the GET FULL VARIABLE agi command.
  
 I am passing in the variable name and the channel name.
 It is responding with variable name back to the script and not the value.
  
 Failed
  SIP/6167761066.2012-0060AGI Rx  GET FULL VARIABLE aatAct0 
SIP/6167761066.2012-0060
SIP/6167761066.2012-0060AGI Tx  200 result=1 (aatAct0)
 
 Expected
  SIP/6167761066.2012-0060AGI Rx  GET VARIABLE aatAct0
SIP/6167761066.2012-0060AGI Tx  200 result=1 
(Macro~SBussniessMSIP-Operator~1)

  
 I am on asterisk 11.10.2
  
 I am I using the Get Full Variable wrong any ideas?
  
 Thanks

Bryant Zimmerman (ZK Tech Inc.)
616-855-1030 Ext. 2003

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Re: [asterisk-dev] Issue AGI Get Full Variable - Solution

[Bryant Zimmerman]

From: Bryant Zimmerman brya...@zktech.com
Sent: Tuesday, September 2, 2014 3:10 AM
To: Asterisk Developers Mailing List asterisk-dev@lists.digium.com
Subject: Re: [asterisk-dev] Issue AGI Get Full Variable   
 I am calling the GET FULL VARIABLE agi command.

   I am passing in the variable name and the channel name.
   It is responding with variable name back to the script and not the 
value.

   Failed
SIP/6167761066.2012-0060AGI Rx  GET FULL VARIABLE aatAct0 
SIP/6167761066.2012-0060
SIP/6167761066.2012-0060AGI Tx  200 result=1 (aatAct0)
 
 Expected
  SIP/6167761066.2012-0060AGI Rx  GET VARIABLE aatAct0
SIP/6167761066.2012-0060AGI Tx  200 result=1 
(Macro~SBussniessMSIP-Operator~1)


   I am on asterisk 11.10.2

   I am I using the Get Full Variable wrong any ideas?

 ---
  
 Ok I have figured this one out. The issue is the variable name must be 
wrapped in the ${variablename} format.
  
 The agi show commands topic get full variable docs do not show this format 
requirement.  How do we get them updated so others do not get bit by this?
  
  
-= Info about agi 'get full variable' =-
 [Syntax]
get full variable variablename [channel name]
 Should read something Like: get full variable ${variablename} [channel 
name]
  Or Like: get full variable ${FUNC(variablename)} [channel name]

  
 [Description]
Returns '0' if variablename is not set or channel does not exist. 
Returns
'1' if variablename is set and returns the variable in parenthesis.
 This does an eval on all var expressions.  variablename should be 
included in ${variablename}, ${FNC(variablename)}, or $[expr]
Understands complex variable names and builtin variables, unlike GET 
VARIABLE.

Example return code: 200 result=1 (testvariable value)
  
 [Synopsis]
Evaluates a channel expression
 [Runs Dead]
Yes
 [See Also]
Not available
  

  
  
  
  
   Thanks

Bryant Zimmerman (ZK Tech Inc.)
616-855-1030 Ext. 2003


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Re: [asterisk-dev] Asterisk and video conferencing

[Bryant Zimmerman]

Sangoma has trans coding solutions that allow for use with virtual 
machines. They had video codecs in their road map. I am not sure if they 
have them yet?
  
 Thanks

Bryant Zimmerman (ZK Tech Inc.)
616-855-1030 Ext. 2003
  


 From: Dan Cropp d...@amtelco.com
Sent: Monday, March 31, 2014 12:02 PM
To: Asterisk Developers Mailing List asterisk-dev@lists.digium.com
Subject: Re: [asterisk-dev] Asterisk and video conferencing   
I've been lobbying hardware manufacturers to provide video cards for 
Asterisk where we can have licenses to do transcoding and reformatting, so 
far with no success.

I passed this onto someone in our hardware department to look into.

I do worry about the thought of hardware for a video solution. We go into a 
lot of hospitals. They only want virtual servers. Not sure that a hardware 
based video solution will go over very well in many markets.

For those worried about bandwidth of video, would it be possible to offload 
that work to another Asterisk box B? Put audio on box A. If you need video 
conferencing, have box A send that to box B?

-Original Message-
From: asterisk-dev-boun...@lists.digium.com 
[mailto:asterisk-dev-boun...@lists.digium.com] On Behalf Of Olle E. 
Johansson
Sent: Monday, March 31, 2014 6:42 AM
To: Asterisk Developers Mailing List
Cc: Olle E Johansson
Subject: Re: [asterisk-dev] Asterisk and video conferencing

On 31 Mar 2014, at 12:47, Johan Wilfer li...@jttech.se wrote:

 Hi!

 I've spent some time scratching my head thinking about video conferencing 
and how to go about it. Right now we use Meetme as a audio bridge for pstn 
connectivity and so on. But our users ask for video and screen sharing. I 
can see four distinct ways to go about this:

 1. Asterisk right now - supports 1-1 video, and with confbridge 1 video 
stream can be sent to the other participants. (The codec must match 
thought, and a key-frame are not sent immediate if the video source is 
changed so there will be a garbled video stream until next key-frame.)

 2. MCU - multiple video streams encoded in single stream. For video do 
the same as with audio. That means decode each stream, compose a new stream 
with all the participants layered out nicely. While this works for audio it 
consumes huge amounts of cpu to do this for video.

 3. p2p - multiple video streams sent peer to peer. Each participant sends 
the audio/video to every other participant. This eats a lot of bandwidth 
for the users and can work for smaller conferences, but in a conference 
with 10 participants each will have to have a very good upstream 
connection.

 4. Jitsi Videobridge - multiple video streams from server, but send only 
your stream to the server. The jitsi videobridge the distributes the stream 
to all other clients. This will eat a lot of bandwidth for the server, but 
not for the clients. This is also how Google Hangouts works. So if you are 
10 participants you will send one stream to the server with your 
audio/video and receive 9 streams from the server for the other 
participants.


 To be able to scale reasonably I think option 2 is out of the question. 
And option 3, p2p, eats to much bandwidth for the clients (and doesn't 
require an asterisk anyway).

 What you lose with option 4 is everything asterisk excels at: pstn 
connectivity, fine-grained control of each participant in the bridge.

 What are your thoughts on adding Jitsis approach in regards to video to 
Asterisk for confbridge or even ARI? No composing of video, just relaying 
the other participants streams to each other in the bridge. Then it's up to 
the client in the other end to display these streams in a reasonable way 
(like Google Hangout, and https://meet.jit.si/).

Why? The jitsi video bridge exists and work fine :-)

What you are forgetting here is the thing that has stopped us from doing 
really cool stuff with video - patents and licensing. The jitsi video 
bridge is a nice workaround, but not optimal if you have a lot of different 
devices. You put the load on the device and in bandwidth-constrained 
environments that's not good.

Video is heavily dependend on peer2peer negotiation and doesn't really fit 
well in a PBX b2bua architecture... The jitsi model could work - but the 
SDP o/a handling would be really hard to get right in Asterisk.

I've been lobbying hardware manufacturers to provide video cards for 
Asterisk where we can have licenses to do transcoding and reformatting, so 
far with no success. Cisco's H264 codecs recently became available for us 
in the Open Source world thanks to a generous solution by Cisco. I guess 
funding is needed to add anything cool to Asterisk using them. We can do 
MCU-style stuff, reformatting - but to do transcoding we need another codec 
:-)

Google VP8 is around, I don't know what Digium's legal team have to say 
about us using it.

Random thoughts...

/O