Re: [asterisk-users] Asterisk 12 - zombie processes

2014-10-29 Thread Yaron Nachum 
No,
I went over all my scripts.

Thanks for the help.

Yaron

On Wed, Oct 29, 2014 at 6:11 PM, Paul Belanger  wrote:

> On Tue, Oct 28, 2014 at 11:10 AM, Yaron Nachum 
> wrote:
> > Mathew,
> > When I run 'ps -ef|grep asterisk' the following processes are displayed:
> > root  6861 1  0 Aug27 ?00:00:00 /bin/sh
> > /ivr/app/asterisk/sbin/safe_asterisk -U asterisk -G asterisk -C
> > /ivr/app/asterisk/etc/asterisk/asterisk.conf
> > asterisk  8062  6861  3 Oct27 ?00:44:56
> > /ivr/app/asterisk/sbin/asterisk -f -U asterisk -G asterisk -C
> > /ivr/app/asterisk/etc/asterisk/asterisk.conf -vvvg -c
> > root 20776  2200  0 11:20 pts/200:00:33 tail -f asterisk.log
> > asterisk 23076  8062  0 17:01 ?00:00:00 [asterisk] 
> > asterisk 23897  8062  0 17:03 ?00:00:00 [asterisk] 
> >
> > also when I run top the same amount of zombie processes are displayed:
> > Tasks: 185 total,   1 running, 182 sleeping,   0 stopped,   2 zombie
> >
> > Regarding the AGI - we are using AGI in order to run php scripts for
> > external logic. I have printed the PIDs of the php scripts and none of
> them
> > are related to the PID's of those zombie processes.
> > Do you have any idea how to find out what are these processes?
> > Yaron.
> >
> Are you doing anything like:
>
> # asterisk -rx 'core show channels'
>
> via an external process?
>
> --
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> https://twitter.com/pabelanger
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Re: [asterisk-users] Asterisk 13 : SILK codec ?

2014-10-29 Thread sean darcy 

On 10/29/2014 08:06 PM, Matthew Jordan wrote:

On Wed, Oct 29, 2014 at 5:16 PM, sean darcy  wrote:

Can we expect a SILK codec for 13 ? Or does the one for 12 work for 13?



codec_silk for Asterisk 12 will most likely not work in Asterisk 13. A
number of performance improvements in the media handling in Asterisk
required some codec compatibility changes.

I would expect said modules to be available in the next few weeks.


Great. Thanks.
sean


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Re: [asterisk-users] Asterisk 13 : SILK codec ?

2014-10-29 Thread Matthew Jordan 
On Wed, Oct 29, 2014 at 5:16 PM, sean darcy  wrote:
> Can we expect a SILK codec for 13 ? Or does the one for 12 work for 13?
>

codec_silk for Asterisk 12 will most likely not work in Asterisk 13. A
number of performance improvements in the media handling in Asterisk
required some codec compatibility changes.

I would expect said modules to be available in the next few weeks.

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[asterisk-users] Asterisk 13 : SILK codec ?

2014-10-29 Thread sean darcy 

Can we expect a SILK codec for 13 ? Or does the one for 12 work for 13?

sean


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Re: [asterisk-users] AstriDevCon 2014: Agenda item Deprecate AMI/AGI (Ben Klang)

2014-10-29 Thread Paul Albrecht 

On Oct 28, 2014, at 5:03 PM, Ben Langfeld  wrote:

> On 28 October 2014 19:47, Derek Andrew  wrote:
> What is the alternative to the dial plan? Is everyone talking about getting 
> rid of the statements like:
> exten => s,1,
> 
> what is the alternative? 
> 
> Remote applications based on APIs like ARI. This is the start of the 
> discussion, and please remember that nothing has been decided or even 
> presented as a robust plan yet. This is brain-storming.
> 

We’re not at the start of the “discussion” to deprecate the dial plan. The 
start of the “discussion” began when some developers decided to try standing 
Asterisk on its head by adding  “asynchronous AGI.” Evidently, that was good so 
then they continued the “discussion” by adding ARI/Stasis. Now the “discussion” 
is in full career as ARI/Stasis has metastasized beyond its original scope to 
encompass all of Asterisk. None of said “discussion” ever happened on the lists 
nor was the broader Asterisk community involved as far as I can determine. A 
parallel “discussion” was started by a shill at AstiCon this year to begin to 
get the “vast unwashed” onboard with ARI/Stasis, that is, so that Matt could 
come back from AstiCon claiming that the broader Asterisk community is in 
agreement that ARI/Stasis is the future of Asterisk and that the dial plan can 
be deprecated. The inevitable result of these parallel paths is a completely 
predictable train wreck when the developers designing features that users don’t 
want crash into users who have been using Asterisk as originally designed.

> Additionally, note that the original proposal was to deprecate AMI/AGI in 
> favour of ARI once it is feature complete with those protocols; an entirely 
> lesser change than the removal of the dialplan in its entirety.

So you're saying that deprecating the dial plan is not on the table? How then 
do you explain statements like this: "Leif: we're in a transition, moving from 
dialplan model to external control model.  Probably need external application 
to be built for us to move completely away from AMI/AGI.” or  this "Paul: take 
away apps, and whatever is in the core is what we should care about.”

>  
> 
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Re: [asterisk-users] Asterisk 12 Dialplan

2014-10-29 Thread Matthew Jordan 
On Wed, Oct 29, 2014 at 1:21 PM, Murthy Gandikota  wrote:
> I am happy to report that
> https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Applications+REST+API
> has the answer to my dilemma. It seems an app has to subscribe to channel
> events before it can receive the events like ChannelVarset...
>

That's correct. You are only implicitly subscribed to channels that
are in the Stasis application your websocket is for (in your case,
'hello-world'). Otherwise, you have to subscribe to various event
sources through the applications resource.

The "Introduction to ARI and Channels" page on the wiki has more on this here:

https://wiki.asterisk.org/wiki/display/AST/Introduction+to+ARI+and+Channels#IntroductiontoARIandChannels-ChannelsinaStasisApplication

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[asterisk-users] OT: script to remove leading and trailing silence

2014-10-29 Thread Steve Edwards 
Anybody care to share a script or snippet of what they use to remove 
leading and trailing silence from customer recorded files?


I've fiddled with sox to remove the leading; reverse the file; remove the 
now leading; and reverse the file again, but I'm not really happy with my

results.

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Newline  Fax: +1-760-731-3000

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Re: [asterisk-users] Astricom 2014 presentations

2014-10-29 Thread Bryant Zimmerman 
 
   On 10/29/2014 05:50 AM, Bogdan Cristea wrote:
> Hi
>
> Will the presentations made at Astricom 2014 be made public as recorded 
videos ?
>
> thanks
> Bogdan

I'll second the request for that, and also ask if the sessions on
Kamailio will be similarly available.

Cheers,

j

That would be awesome if they chose to do this.
  
 Bryant
 


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Re: [asterisk-users] Asterisk 12 Dialplan

2014-10-29 Thread Murthy Gandikota 
I am happy to report that
https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Applications+REST
+API has the answer to my dilemma. It seems an app has to subscribe to
channel events before it can receive the events like ChannelVarset... 

 



From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Murthy
Gandikota
Sent: Tuesday, October 28, 2014 2:49 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 12 Dialplan

 

Tried this:

 

wscat -c
"ws://myhost.mydomain.net:8090/ari/events?api_key=secret:secret&app=hell
o-world"

 

It is only showing the stasis related events. I am interested in AMI
events, specifically Varset.

 

Thanks



From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Murthy
Gandikota
Sent: Monday, October 27, 2014 7:54 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 12 Dialplan

 

 

 



From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matthew
Jordan
Sent: Monday, October 27, 2014 3:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 12 Dialplan

 

 

 

On Mon, Oct 27, 2014 at 2:40 PM, Murthy Gandikota 
wrote:

 

 

I am unable to detect the Manager_Setvar event using ARI.

Can you please let me know, in ARI lingo, the curl or javascript code to
detect the AMI Manager_Setvar event for myvar in the following dialplan:

 

[default]

 exten => 1000,1,NoOp()

 same =>  n,Answer()

 same =>  n,set(myvar=test)

 same =>  n,Stasis(hello-world)

 same =>  n,Hangup()

 

Thanks




 

Perhaps it would be easier if you provided some information about the
ARI application you've written. Have you connected a WebSocket? Are you
receiving other ARI events?


-- 

Matthew Jordan

Digium, Inc. | Engineering Manager

445 Jan Davis Drive NW - Huntsville, AL 35806 - USA

Check us out at: http://digium.com & http://asterisk.org

 

I am using ari4java to capture stasis events like StasisStart,
StatisEnd, etc. However,  I am unable to capture the Varset event as
explained before. In particular the myvar variable is not associated
with any app It is perhaps a channel variable. 

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Re: [asterisk-users] Asterisk 12 - zombie processes

2014-10-29 Thread Paul Belanger 
On Tue, Oct 28, 2014 at 11:10 AM, Yaron Nachum  wrote:
> Mathew,
> When I run 'ps -ef|grep asterisk' the following processes are displayed:
> root  6861 1  0 Aug27 ?00:00:00 /bin/sh
> /ivr/app/asterisk/sbin/safe_asterisk -U asterisk -G asterisk -C
> /ivr/app/asterisk/etc/asterisk/asterisk.conf
> asterisk  8062  6861  3 Oct27 ?00:44:56
> /ivr/app/asterisk/sbin/asterisk -f -U asterisk -G asterisk -C
> /ivr/app/asterisk/etc/asterisk/asterisk.conf -vvvg -c
> root 20776  2200  0 11:20 pts/200:00:33 tail -f asterisk.log
> asterisk 23076  8062  0 17:01 ?00:00:00 [asterisk] 
> asterisk 23897  8062  0 17:03 ?00:00:00 [asterisk] 
>
> also when I run top the same amount of zombie processes are displayed:
> Tasks: 185 total,   1 running, 182 sleeping,   0 stopped,   2 zombie
>
> Regarding the AGI - we are using AGI in order to run php scripts for
> external logic. I have printed the PIDs of the php scripts and none of them
> are related to the PID's of those zombie processes.
> Do you have any idea how to find out what are these processes?
> Yaron.
>
Are you doing anything like:

# asterisk -rx 'core show channels'

via an external process?

-- 
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[asterisk-users] My Asterisk can not send fax via T.38

2014-10-29 Thread Weiqi 
Hello,

I am a newer on asterisk. when I tried to send fax, I can not get success.
After doing a lot of reseach, I decide to ask my question at here.

Instead of using old fax machine, I want my system can send fax via t.38.
Could anyone  give me some idea to correct my configuration?

Thanks a lot!
Here is the detail:

my system: *ubuntu 14.04 + asterisk 11.7*

I use apt-get to install the ubuntu default version of asterisk 11.7. for
easy debug, I just change a few param value with the default version
sip.conf

t38pt_udptl = yes,redundancy,maxdatagram=400
faxdetect = yes

extension.conf

[sendFAX]
exten => s,1,VERBOSE(sending fax...)
exten => s,n,Set(FAXOPT(headerinfo)=Fax from a Demo test)
exten => s,n,SendFAX(/tmp/demo.tiff,f)
;I get demo.tiff file from "$ gs -q -dNOPAUSE -dBATCH -sDEVICE=tiffg4
-sPAPERSIZE=letter -sOutputFile= "
exten => s,n,VERBOSE(ok!)
exten => s,n,Hangup

I use AMI to originate a fax call. In the CLI, everything looks well. I
didn't get any error message. When I use wireshark to check the detail of
this comunication, I found Asterisk used G711 instead of using T.38 which
is expected.

however, at the receiver end, I didn't receive the fax, and I just got a
error "Dcn No Dis"
After a research, I got this:

T.30 Fax Signaling Messages In a Voip fax call, T.38 packets are preceded
and succeeded by T.30 fax signaling messages. These messages include:


   1. DIS: Digital Identification Signal indicating terminating fax
   capabilities (for example, data rate)
   2. DCS: Digital Command Signal indicating transmission mode that will be
   used by originating fax (for example, transfer rate)
   3. TCF: Training Check Sequences (sent for 1.5 seconds)
   4. CFR: Confirmation To Receive indicating the receiving fax is ready to
   receive the document
   5. MPS: MultiPage Signal (sent after each page if more than one page is
   sent)
   6. MCF: Message Confirmation indicating the page was received
   7. EOP: End Of Procedure message indicating there are no more pages to
   be sent
   8. DCN: Disconnect message

 Additional optional messages:

1.CSI: Called Subscriber Identification

2.TSI: Transmitting Subscriber Identification

But I am still confused with what "Dcn No Dis" means what's wrong with my
asterisk system.

I am sure of these:

   1. the receiver is working well.
   2. My ISP provider is fully support fax termination both in g711 and t.38
   3. My testing server is not behind any firewall.

The demo.jpg  is my wireshark's screenshot. form 19
to 1841, all traffic are RTP package.

The t.38 png  diagram illustrates is a typical fax
call.
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Re: [asterisk-users] Astricom 2014 presentations

2014-10-29 Thread David Duffett 
I can confirm that all the videos from AstriCon 2014 will be available at
www.AstriCon.net within about 3 weeks.
On 29 Oct 2014 16:33, "Jeff LaCoursiere"  wrote:

>
> On 10/29/2014 05:50 AM, Bogdan Cristea wrote:
>
>> Hi
>>
>> Will the presentations made at Astricom 2014 be made public as recorded
>> videos ?
>>
>> thanks
>> Bogdan
>>
>
> I'll second the request for that, and also ask if the sessions on Kamailio
> will be similarly available.
>
> Cheers,
>
> j
>
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Re: [asterisk-users] Astricom 2014 presentations

2014-10-29 Thread Jeff LaCoursiere 


On 10/29/2014 05:50 AM, Bogdan Cristea wrote:

Hi

Will the presentations made at Astricom 2014 be made public as recorded videos ?

thanks
Bogdan


I'll second the request for that, and also ask if the sessions on 
Kamailio will be similarly available.


Cheers,

j

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Re: [asterisk-users] Asterisk 12 - zombie processes

2014-10-29 Thread Matthew Jordan 
On Wed, Oct 29, 2014 at 6:59 AM, Yaron Nachum  wrote:
> Hello Mathew and everyone,
> We are still having reboots on our asterisk servers. The latest 12.6.1
> release doesn't fix the issue.
>
> We have the core files of the latest reboots and also debug taken during the
> reboot.
>
> We would like to open an issue. What kind of information you need for the
> issue?
>

Please do open an issue on issues.asterisk.org. Instructions for
generating a backtrace are on the Asterisk wiki:

https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace

Make sure you have DONT_OPTIMIZE and, preferably, BETTER_BACKTRACES
selected in menuselect.

Depending on the nature of the crash, you may be asked for more
information, but we won't know until we see the backtrace.

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Re: [asterisk-users] Query on connecting 3G MSC with Asterisk PBX Server via SIP Interface.

2014-10-29 Thread Chris Bagnall 

On 29/10/14 12:59 pm, A J Stiles wrote:

Imagine what would have happened to the human race if Ugg the Caveman decided
not to share the secret of making fire with everyone freely, but instead went
around demanding shiny beads with menaces from anyone who just wanted to keep
themselves warm .


That's the best analogy I've heard in favour of open development for a 
long time, and something that non-techs can understand.


I thank you sir :-)

Kind regards,

Chris
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Re: [asterisk-users] Query on connecting 3G MSC with Asterisk PBX Server via SIP Interface.

2014-10-29 Thread A J Stiles 
On Tuesday 28 Oct 2014, NACHIAPPAN, SUBBAIAH (SUBBAIAH) wrote:
> Hello,
> 
> I am new to Asterisk forum :).
> 
> I have a requirement of terminating  3G Mobile originated calls (coming
> through 3G-MSC)  to EPBX Phones via Asterisk PBX.
> 
> Setup:
> 
> Mobile  > Mobile Switching Center ( 3G)-SIP interface--->Asterisk
> PBX--->SIP Phone.
> 
> I wanted to know if I require SIP licenses to integrate 3G MSC with my
> Asterisk server.

Of course not  :)  SIP is an Open technology, anybody is free to implement it.

Imagine what would have happened to the human race if Ugg the Caveman decided 
not to share the secret of making fire with everyone freely, but instead went 
around demanding shiny beads with menaces from anyone who just wanted to keep 
themselves warm .

> I know there is a wealth of information in wiki link, but I am unable to
> locate the  required configuration document which will help me in
> integrating MSC with Asterisk EPBX via SIP interface.

You need to contact, in the first instance, the company who are providing your 
SIP interface.  They will have the necessary interoperability documents you 
require.


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list, change address to asterisk1list at earthshod dot co dot uk .

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Re: [asterisk-users] Query on connecting 3G MSC with Asterisk PBX Server via SIP Interface.

2014-10-29 Thread NACHIAPPAN, SUBBAIAH (SUBBAIAH) 
Hello Experts,

Could anybody pl help resolve my query?

Thanks & Regards,
Subbaiah Nachiappan

From: NACHIAPPAN, SUBBAIAH (SUBBAIAH)
Sent: Tuesday, October 28, 2014 6:04 PM
To: 'asterisk-users@lists.digium.com'
Subject: RE: Query on connecting 3G MSC with Asterisk PBX Server via SIP 
Interface.

Hello Folks,

Forgot to mention the software Versions which I am using:

Asterisk: 1.8

Free PBX: 2.11

Asterisk NOW: 5.211.65


Thanks & Regards,
Subbaiah Nachiappan

From: NACHIAPPAN, SUBBAIAH (SUBBAIAH)
Sent: Tuesday, October 28, 2014 5:52 PM
To: 'asterisk-users@lists.digium.com'
Subject: Query on connecting 3G MSC with Asterisk PBX Server via SIP Interface.


Hello,



I am new to Asterisk forum :).



I have a requirement of terminating  3G Mobile originated calls (coming through 
3G-MSC)  to EPBX Phones via Asterisk PBX.





Setup:





Mobile  > Mobile Switching Center ( 3G)-SIP interface--->Asterisk 
PBX--->SIP Phone.



I wanted to know if I require SIP licenses to integrate 3G MSC with my Asterisk 
server.



I know there is a wealth of information in wiki link, but I am unable to locate 
the  required configuration document which will help me in integrating MSC with 
Asterisk EPBX via SIP interface.



Thanks in Advance!!!


Thanks & Regards,
Subbaiah Nachiappan


-Original Message-
From: 
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Sent: Tuesday, October 28, 2014 5:41 PM
To: NACHIAPPAN, SUBBAIAH (SUBBAIAH)
Subject: Welcome to the "asterisk-users" mailing list


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Re: [asterisk-users] Asterisk 12 - zombie processes

2014-10-29 Thread Yaron Nachum 
Hello Mathew and everyone,
We are still having reboots on our asterisk servers. The latest 12.6.1
release doesn't fix the issue.

We have the core files of the latest reboots and also debug taken during
the reboot.

We would like to open an issue. What kind of information you need for the
issue?

Thanks,
Yaron.


On Tue, Oct 28, 2014 at 5:10 PM, Yaron Nachum 
wrote:

> Mathew,
> When I run 'ps -ef|grep asterisk' the following processes are displayed:
> root  6861 1  0 Aug27 ?00:00:00 /bin/sh
> /ivr/app/asterisk/sbin/safe_asterisk -U asterisk -G asterisk -C
> /ivr/app/asterisk/etc/asterisk/asterisk.conf
> asterisk  8062  6861  3 Oct27 ?00:44:56
> /ivr/app/asterisk/sbin/asterisk -f -U asterisk -G asterisk -C
> /ivr/app/asterisk/etc/asterisk/asterisk.conf -vvvg -c
> root 20776  2200  0 11:20 pts/200:00:33 tail -f asterisk.log
> asterisk 23076  8062  0 17:01 ?00:00:00 [asterisk] 
> asterisk 23897  8062  0 17:03 ?00:00:00 [asterisk] 
>
> also when I run top the same amount of zombie processes are displayed:
> Tasks: 185 total,   1 running, 182 sleeping,   0 stopped,   2 zombie
>
> Regarding the AGI - we are using AGI in order to run php scripts for
> external logic. I have printed the PIDs of the php scripts and none of them
> are related to the PID's of those zombie processes.
> Do you have any idea how to find out what are these processes?
> Yaron.
>
> On Tue, Oct 28, 2014 at 4:53 PM, Matthew Jordan 
> wrote:
>
>> On Tue, Oct 28, 2014 at 9:44 AM, Yaron Nachum 
>> wrote:
>> > Hello Mathew,
>> > In the following tutorial it says that channel are marked with ZOMBIE
>> flag.
>> > From your response I assume it has no connection to my problem.
>> > https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Bridging+Project
>> >
>> > Regrading the zombie processes issue we are having. We have debug taken
>> from
>> > the server during such process is invoked. If you want I can attache it.
>>
>> A zombie channel has nothing to do with a process. It was an artefact
>> of an internal process known as a masquerade. While masquerades do
>> sometimes still occur in Asterisk 12+, they are far less frequent and
>> are no longer externally visible.
>>
>> Why do you think you have zombie processes? Asterisk does use a large
>> number of threads, but generally rarely forks processes unless you are
>> using something like original AGI.
>>
>> --
>> Matthew Jordan
>> Digium, Inc. | Engineering Manager
>> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
>> Check us out at: http://digium.com & http://asterisk.org
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
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>> asterisk-users mailing list
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>
>
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[asterisk-users] Astricom 2014 presentations

2014-10-29 Thread Bogdan Cristea 
Hi

Will the presentations made at Astricom 2014 be made public as recorded videos ?

thanks
Bogdan
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