Re: [asterisk-users] [asterisk-app-dev] Asterisk 13 ARI Playback of audio via HTTP

[Matt Riddell]

Answering the below for search engine’s sake.

> On Feb 18, 2019, at 11:23, Matt Riddell  wrote:
> 
> Hey, trying to use ARI with NodeJS - this doesn't work: 
> 
> play(channel, 'sound:http://www.nch.com.au/acm/8k16bitpcm.wav' 
> <http://www.nch.com.au/acm/8k16bitpcm.wav'>);

Problem 1: The url is http but gets redirected to https in the browser
Problem 2: Although the wiki doesn’t state it, you need at least version 14 of 
Asterisk and I was using version 13.  In the process of updating now___
asterisk-app-dev mailing list
asterisk-app-...@lists.digium.com
http://lists.digium.com/cgi-bin/mailman/listinfo/asterisk-app-dev
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

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

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] [asterisk-app-dev] Asterisk 13 ARI Playback of audio via HTTP

[Matt Riddell]

Hey, trying to use ARI with NodeJS - this doesn't work: 

play(channel, 'sound:http://www.nch.com.au/acm/8k16bitpcm.wav');

should it?

https://wiki.asterisk.org/wiki/display/AST/ARI+and+Channels%3A+Simple+Media+Manipulation
 says:

A sound file located on the Asterisk system. You can use the /sounds resource 
to query for available sounds on the system. You can also use specify a media 
file which is consumed via HTTP (e.g sound:http://foo.com/sound.wav)

But I get

[2019-02-18 11:18:29] WARNING[23860][C-0002f9e3]: file.c:774 
ast_openstream_full: File http://www.nch.com.au/acm/8k16bitpcm.wav does not 
exist in any format

Cheers,

Matt___
asterisk-app-dev mailing list
asterisk-app-...@lists.digium.com
http://lists.digium.com/cgi-bin/mailman/listinfo/asterisk-app-dev
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

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

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] [asterisk-app-dev] ARI-client Node.js objects

[Matt Riddell]

> On Jan 11, 2019, at 10:46, Gilles VERRIEZ (SERENEO) 
>  wrote:
> 
> Hi,
> 
> I would like to get the audio resource from a record in order to send it 
> threw AJAX request with my ARI-client Node JS source. I thought 
> Playback.media_uri could help me but it's value is undefined. Any ideas?
> 
> 
I think you’re probably going to be looking for:

ari.recordings.getStoredFile(
  {recordingName: val},
  function (err, binary) {}
);

___
asterisk-app-dev mailing list
asterisk-app-...@lists.digium.com
http://lists.digium.com/cgi-bin/mailman/listinfo/asterisk-app-dev
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

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

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] [asterisk-app-dev] Multiple ChannelDestroyed events for the same channel

[Matt Riddell]

> On Jan 11, 2019, at 11:14, Jean Aunis  wrote:
> 
> Le 11/01/2019 à 16:47, Matt Riddell a écrit :
>> Hiya,
>> 
>> When I hang up on a call to my stasis app I’m getting multiple 
>> channelDestroyed events for the same channel:
> 
> It may happen if several applications subscribed to the channel. In this case 
> you can discriminate the events on the basis of application name.

Yeah you’re right - I was registering 3 applications so get three events - even 
though the channel was only in one application.

Thanks,

Matt Riddell
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

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

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] [asterisk-app-dev] Multiple ChannelDestroyed events for the same channel

[Matt Riddell]

Hiya,

When I hang up on a call to my stasis app I’m getting multiple channelDestroyed 
events for the same channel:

app.js:985:13) Channel was destroyed: 1547220509.77
app.js:1029:17) This was a customer
app.js:1030:17) Checking if this was a customer talking to an agent
app.js:1043:21) Customer was not talking to anyone
app.js:1126:13) 2019-01-11 10:28:29
app.js:985:13) Channel was destroyed: 1547220509.77
app.js:1029:17) This was a customer
app.js:1030:17) Checking if this was a customer talking to an agent
app.js:1043:21) Customer was not talking to anyone
app.js:1126:13) 2019-01-11 10:28:29
app.js:985:13) Channel was destroyed: 1547220509.77
app.js:1029:17) This was a customer
app.js:1030:17) Checking if this was a customer talking to an agent
app.js:1043:21) Customer was not talking to anyone

I’m only registering once to the event:

ari.on('ChannelDestroyed', channelDestroyed);

Is this normal?

I’m writing like a CDR on channel destroyed so don’t want to write it multiple 
times.

Should I keep an array of channels and only write if I haven’t seen the event 
for that channel before?

Cheers,

Matt Riddell
___
asterisk-app-dev mailing list
asterisk-app-...@lists.digium.com
http://lists.digium.com/cgi-bin/mailman/listinfo/asterisk-app-dev
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

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

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] [asterisk-app-dev] ARI Node JS Bridge.addChannel

[Matt Riddell]

> On Jan 7, 2019, at 12:25, Joshua C. Colp  wrote:
> 
> On Mon, Jan 7, 2019, at 1:23 PM, Matt Riddell wrote:
>> Hiya,
>> 
>> I would have expected this to show the channels in the bridge inside 
>> the anonymous function - it shows the bridge is empty though?
>> 
>>var bridge = ari.Bridge();
>>bridge.create({
>>type: 'holding',
>>name: event.application+" bridge"
>>}, function(err, bridge) {
>>bridge.addChannel({
>>channel: incoming.id
>>}, function(err) {
>>console.log("Added to bridge")
>>console.log(bridge.channels).   ; 
>> <——— This Line
>>console.log(err);
>>});
>> 
> 
> I believe you are accessing the snapshot, essentially, of the bridge at the 
> time it was created in which case there would be no channels. You would need 
> to retrieve an up to date snapshot to get the current state.

Yeah cool that worked:

ari.bridges.get({bridgeId: bridge.id}, function (err, newBridge) {
console.log("New Bridge: "+newBridge.channels)
});
___
asterisk-app-dev mailing list
asterisk-app-...@lists.digium.com
http://lists.digium.com/cgi-bin/mailman/listinfo/asterisk-app-dev
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

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

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] [asterisk-app-dev] ARI Node JS Bridge.addChannel

[Matt Riddell]

> On Jan 7, 2019, at 12:25, Joshua C. Colp  wrote:
> 
> On Mon, Jan 7, 2019, at 1:23 PM, Matt Riddell wrote:
>> Hiya,
>> 
>> I would have expected this to show the channels in the bridge inside 
>> the anonymous function - it shows the bridge is empty though?
>> 
>>   var bridge = ari.Bridge();
>>   bridge.create({
>>   type: 'holding',
>>   name: event.application+" bridge"
>>   }, function(err, bridge) {
>>   bridge.addChannel({
>>   channel: incoming.id
>>   }, function(err) {
>>   console.log("Added to bridge")
>>   console.log(bridge.channels).   ; 
>> <——— This Line
>>   console.log(err);
>>   });
>> 
> 
> I believe you are accessing the snapshot, essentially, of the bridge at the 
> time it was created in which case there would be no channels. You would need 
> to retrieve an up to date snapshot to get the current state.

Yeah cool that worked:

ari.bridges.get({bridgeId: bridge.id}, function (err, newBridge) {
console.log("New Bridge: "+newBridge.channels)
});
___
asterisk-app-dev mailing list
asterisk-app-...@lists.digium.com
http://lists.digium.com/cgi-bin/mailman/listinfo/asterisk-app-dev
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

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

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] [asterisk-app-dev] ARI Node JS Bridge.addChannel

[Matt Riddell]

Hiya,

I would have expected this to show the channels in the bridge inside the 
anonymous function - it shows the bridge is empty though?

var bridge = ari.Bridge();
bridge.create({
type: 'holding',
name: event.application+" bridge"
}, function(err, bridge) {
bridge.addChannel({
channel: incoming.id
}, function(err) {
console.log("Added to bridge")
console.log(bridge.channels).   ; <——— This 
Line
console.log(err);
});

Reason being, I’m creating a queue need to move channels between bridges 
depending on agent/customer status etc

Cheers,

Matt Riddell
___
asterisk-app-dev mailing list
asterisk-app-...@lists.digium.com
http://lists.digium.com/cgi-bin/mailman/listinfo/asterisk-app-dev
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

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

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Function CHANNELS

[Matt Hamilton]

I appreciate it if someone can post an an example for function CHANNELS showing 
the usage of the regular expression filter.


Basically I would like to get a count of active channels having a certain 
criteria. Is it possible to search for a channel having a custom variable set 
to specific value (directly in dialplan or thru agi)?


Our Asterisk version is 1.8.
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

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

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] remove

[Matt Fredrickson]

Check the footer at the bottom of this message for instructions on how
to unsubscribe :-)

Matthew Fredrickson

On Fri, Jun 1, 2018 at 12:11 PM, David Mutterer  wrote:
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at: https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users



-- 
Matthew Fredrickson
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

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

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] shell dialplan application blocking

[Matt Riddell (lists)]

Use AGI

Kind regards,

Matt

> On Jun 4, 2018, at 02:28, Benjamin Marty  wrote:
> 
> I'm calling a script which needs to wait a certain time and also hold the 
> call for this time. But the script dialplan application seems to work non 
> blocking. Is there a way to hold the call/dialplan till the shell script is 
> finished?
> 
>same => n,Set(PUSHRESULT=${SHELL(sendpush.sh)})
> 
> -- 
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> 
> Check out the new Asterisk community forum at: https://community.asterisk.org/
> 
> New to Asterisk? Start here:
>  https://wiki.asterisk.org/wiki/display/AST/Getting+Started
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

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

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Testing...

[Matt Fredrickson]

Test from non-digium email.

-- 
Matthew Fredrickson
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

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

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] testing users list

[Matt Ball]

testing users list
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

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

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Test from Digium address

[Matt Fredrickson]

Testing again.

-- 
Matthew Fredrickson
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

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

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] asterisk rules

[Matt Ball]

mailman drools
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

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

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] One more test

[Matt Fredrickson]

I need to send one more test.  Here it is!

-- 
Matthew Fredrickson
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

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

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] More testing

[Matt Fredrickson]

More testing.  Test test test. :-)

-- 
Matthew Fredrickson
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

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

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Test

[Matt Fredrickson]

Testing again :-)

-- 
Matthew Fredrickson
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

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

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] AstriCon Approaching, Super Earlybird Pricing Expires In 3 Days

[Matt Fredrickson]

Hey All,

So one of the jobs that I get to do as head of the Asterisk project is
to help inform people about the yearly conference we have about
Asterisk named Astricon.

For those who are not familiar with it, AstriCon is a fantastic event
for anyone that is serious about Asterisk. This year, it's back in
Orlando, Florida, on October 9-11. Just as a heads up, the Super
Earlybird discount on a full AstriCon pass finishes on April 30, so
it's a good time to register to get the best deal:
https://www.asterisk.org/community/astricon-user-conference/register

AstriCon is a great chance to mix with all your favorite Asterisk
Community members and key members of the Asterisk development team
while you learn the latest developments, watch some crazy Dangerous
Demos and just have a whole bunch of fun! The Expo floor is always
worth a visit too, with many in the Asterisk Ecosystem present to show
your their offerings.

Also, by attending you help to financially support the Asterisk
project, as revenue from attendance is directly attributable to the
project and it makes it easier for me to justify Asterisk related
expenses such as hiring more Asterisk developers.

Hope to see many of you there!

-- 
Matthew Fredrickson
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

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

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Wanted: WebRTC tutorial

[Matt Fredrickson]

On Tue, Apr 24, 2018 at 10:54 AM, Bruce Ferrell  wrote:
> A while back (last year maybe?), there was a Digium blog post on setting up
> WebRTC.
>
> I was never able to get that working.
>
> I was working with Asterisk 15 on a RHEL derived distro and had no idea of
> where to go to shoot the failure.
>
> Has anyone got a tutorial with trouble shooting?

Great question!  I'm assuming you're talking about the SFU blog post -
http://blogs.asterisk.org/2017/09/20/asterisk-15-multi-stream-media-sfu/
?

I'd be curious as to what difficulties you ran into.  We actually need
to try to consolidate the information in that post with the webrtc
setup page on the wiki -
https://wiki.asterisk.org/wiki/display/AST/Asterisk+WebRTC+Support

You might try those two pages if you haven't yet.  If you have
already, perhaps posting your specific challenges that you encountered
here might be helpful.

Thanks!

-- 
Matthew Fredrickson
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

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

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Asterisk Community Services Outages

[Matt Fredrickson]

Dear Asterisk Community,

For the past 24 hours or so, Digium’s upstream provider has had a few
outages that have affected Asterisk community services, including
Asterisk.org, the mailing lists, and potentially other services.  We
apologize for any inconvenience that it has caused.  Hopefully things
are back up and running, but please let me know if you see anything
that’s still down.

Thanks so much for your patience.

-- 
Matthew Fredrickson
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

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

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Strange problem with PRI on 64-bit?

[Matt Fredrickson]

On Tue, Apr 3, 2018 at 4:38 PM, Tony Mountifield  wrote:
> In article 
> 

Re: [asterisk-users] Strange problem with PRI on 64-bit?

[Matt Fredrickson]

On Tue, Apr 3, 2018 at 5:44 AM, Tony Mountifield  wrote:
> I have some more investigation to do on this, but I wanted to see if anyone
> here had any insight into the issue I've run into.
>
> The hardware is a HP DL360 G6 with a TE420 gen 5 4-port T1 PRI card. It is one
> of several systems that have been running without issue since 2010/2011. They
> have all been running CentOS 4 32-bit with Zaptel 1.4.12.1 (with patch for gen
> 5 card), libpri 1.2.8 and asterisk 1.2.32.
>
> Having taken this particular system out of production, I updated it to CentOS
> 6.9 32-bit, with DAHDI 2.11.1, LibPRI 1.6.0 and Asterisk 11.25.3 (this version
> of Asterisk is required at the moment due to custom modifications).
> This appears to work fine.
>
> In order to reduce the number of different versions we support, I reinstalled
> the OS using the 64-bit version of CentOS 6.9 instead, and rebuilt, using
> the same versions as above.
>
> However, for reasons I don't understand, the 64-bit version was logging
> frequent PRI errors every few minutes:
>
> [Apr  1 03:40:52] VERBOSE[8989] chan_dahdi.c: PRI Span: 2 TEI=0 MDL-ERROR 
> (A): Got supervisory frame with F=1 in state 7(Multi-frame established)
> [Apr  1 03:40:58] VERBOSE[8988] chan_dahdi.c: PRI Span: 1 TEI=0 MDL-ERROR 
> (A): Got supervisory frame with F=1 in state 7(Multi-frame established)
> [Apr  1 03:44:06] VERBOSE[8990] chan_dahdi.c: PRI Span: 3 TEI=0 MDL-ERROR 
> (A): Got supervisory frame with F=1 in state 7(Multi-frame established)
> [Apr  1 03:46:38] VERBOSE[8990] chan_dahdi.c: PRI Span: 3 TEI=0 MDL-ERROR 
> (A): Got supervisory frame with F=1 in state 7(Multi-frame established)
> [Apr  1 03:47:20] VERBOSE[8988] chan_dahdi.c: PRI Span: 1 TEI=0 MDL-ERROR 
> (A): Got supervisory frame with F=1 in state 7(Multi-frame established)
> [Apr  1 03:47:24] VERBOSE[8989] chan_dahdi.c: PRI Span: 2 TEI=0 MDL-ERROR 
> (A): Got supervisory frame with F=1 in state 7(Multi-frame established)
>
> This left the PRIs in strange states - trying to make a call failed with 
> cause 101.
>
> So I re-installed the 32-bit OS again, and rebuilt, and the above MDL-ERRORs
> were no longer present, and the system operated normally again.
>
> So my question is: does anyone have any clues why there would be a difference
> in PRI behaviour between 32-bit and 64-bit builds? Has anyone else run into
> anything similar?


That does seem quite odd.  If I remember right, those messages would
come up if it looked like the other end hadn't received a message when
it thought it should have.  I can't think of anything that would
particularly impact 64 bit systems versus 32 bit systems in that
domain (ISDN real time message timing, etc).  Are you sure there's
nothing else different (kernel version or something else like that)?
Maybe also run a patlooptest on the spans in question to make sure
that they're running cleanly.

Matthew Fredrickson

>
> Cheers
> Tony
> --
> Tony Mountifield
> Work: t...@softins.co.uk - http://www.softins.co.uk
> Play: t...@mountifield.org - http://tony.mountifield.org
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at: https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users



-- 
Matthew Fredrickson
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

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

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] More testing - sorry guys

[Matt Fredrickson]

Thanks :-)

On Wed, Mar 28, 2018 at 3:52 PM, Markus Weiler
<markus_wei...@mailworks.org> wrote:
> I received it :-)
>
>
> Am 28.03.2018 um 22:44 schrieb Matt Fredrickson:
>>
>> Just a test.
>>
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>  https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users



-- 
Matthew Fredrickson
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

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

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Sorry for interruption of service

[Matt Fredrickson]

Hey All,

Just as a public service announcement, we had a 12-16 hour window with
mailing list service interruption.

Last night we scheduled a time to update the mailing list server but
today found some problems impacting mailing service after the updates.
Due to this discovery, we quickly reverted the updates to restore
mailing list service.  Things should be back up and running now.

I apologize again for the inconvenience.  You might need to resend any
messages that were sent in the last half day or so.

-- 
Matthew Fredrickson
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

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

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] More testing - sorry guys

[Matt Fredrickson]

Just a test.

-- 
Matthew Fredrickson
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

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

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] AMI potential memory leak

[Matt Fredrickson]

On Wed, Mar 21, 2018 at 4:03 PM, Dan Cropp  wrote:
> We are communicating with Asterisk via AMI.  Running Asterisk version
> 13.18.5 on an Ubuntu box.
>
>
>
> If you look at the event response, the Result field is filled with random
> characters.  I’m not sure what to do because that is a completely random
> result.  It makes no sense.
>
>
>
>
>
> We send the following command to asterisk via AMI
>
> Action: AGI
>
> ActionID: C44415
>
> Channel: SIP/192.168.40.105-1338
>
> CommandID: C44415
>
> Command: asyncagi break
>
>
>
> Asterisk processes it and the Asterisk log shows it sends the following for
> the AsyncAGIEXEC event…
>
> This is coming directly from the Asterisk log so it’s clearly Asterisk
> filling the Result with what appears to be random characters.
>
>
>
> [03/21 15:44:27.793] DEBUG[1213] manager.c: Examining AMI event:
>
> Event: AsyncAGIExec
>
> Privilege: agi,all
>
> Channel: SIP/192.168.40.105-1338
>
> ChannelState: 6
>
> ChannelStateDesc: Up
>
> CallerIDNum: 1234
>
> CallerIDName: 
>
> ConnectedLineNum: 
>
> ConnectedLineName: 
>
> Language: en
>
> AccountCode: 11
>
> Context: ABC
>
> Exten: 3002
>
> Priority: 8
>
> Uniqueid: 1521665055.4920
>
> Linkedid: 1521665055.4920
>
> Result:
> %F7%EF%F0%F4z%7C%EE%EF%F6%FCkWDDLO%5Cm%EF%7B__%60h%FE%E0%D4%D5%D9%DA%DD%E2%E8%E1%DC%DE%E0%F3%EC%EF%7C%ED%ED%EC%FAx~oov%F9%F2%F6%EB%ED%FA%FD%F9%FEvjXGFMKWg%FF%7Cdidew%E8%D9%D9%D8%D6%DA%DF%E3%E3%E2%E5%E9%EA%E1%E6%EF%E9%F8%F8%F6%FD%F8%7C~%7D%FC%F9%F7%ED%F4%F6%F9oh%60JFMKP%5Do%FFgtjch%7B%E3%DF%DB%D6%D8%DB%E6%EC%E7%E6%E5%DF%DD%E0%EA%F6%ED%EC%F7%E8%E4%EB%F2%F8%F9%7Cu%FC%F8%FAwhXIIIHQ%5Dksw%7Fgfl%7F%EA%E2%DA%D8%DC%DE%DE%E2%E4%DF%E0%E3%E9%E4%E8%EC%E6%E3%E1%E4%E0%E3%EE%F4%FAxy%FCj%5BLIJFLU%5Bdm%FFljws%FD%ED%E2%DC%DD%DA%D9%DC%DE%DD%E1%E7%E9%E8%E6%EC%E5%E1%E5%E0%DF%E2%E9%ED%F1xyo%5EOKMHJSU%5Bfvml%7F%7B%7B%F0%EB%EA%E2%DD%DD%DB%DB%DC%DD%E1%E0%E1%E3%E3%DF%DF%DF%DE%E0%E5%E9%F6%7CmcTKMJIOTYboon%7Ctt%FA%F7%F3%E9%DE%E1%DC%D9%DB%DE%E1%DE%E6%E7%DF%E1%DF%DD%DC%DE%E1%E1%EE%FAx%5DRMLHJOPWblo%7C%FA%7B~%F9%FC%F7%E9%E1%E1%DC%D9%DC%DC%DB%DF%E3%E0%DF%E0%DE%DC%DF%E1%E1%EE%F9r%5DQMKHKOQXdmt%F8%F6%FC%FB%FFy%F5%EA%E5%DF%DA%D9%DA%D8%DB%DE%DD%DE%DF%DF%DF%E3%EA%EB%ECrkbSPOLLOST_jk%F9%F3%FB%F9%FF%FE%F9%EF%E9%E2%DF%DA%DA%DA%D9%D9%DB%DC%DC%DF%E3%E4%EB%F4%FFmdYRPLKNOTZdl%7D%EE%F6%EE%EB%EF%EE%EA%E3%E7%DE%DB%DD%D8%D9%D9%D9%D9%DC%DF%E1%EE%FAxf_XROMMNQUYbgx%FE%FB%EC%EE%ED%E4%E4%E0%DD%DD%DB%DA%D9%DA%D9%D9%DD%DD%E2%EC%F9tg%5EZSPONORUZ%5Efmw%FC%F3%ED%E7%E2%E1%DC%DB%DB%D8%D9%D9%D8%DA%DB%DD%E2%E9%F6uh%60%5BVSQOOQSX%5B_jq%F9%ED%E8%E1%DF%DC%DA%DA%DA%DA%D9%DB%DC%DC%E3%E6%EE%FEtha%5EYWVTTUWY%5Eckx%FA%EC%E7%E2%DF%DD%DC%DD%DC%DC%DF%DE%E2%E6%E7%F1%F8%7Boid_%5C%5BZY%5BZ%5B%5E%60flu%FF%F4%EC%E8%E3%E1%E1%DF%DF%E1%E0%E4%E7%E9%EE%F5%FByqmhfca%60%5E__%60bfhlqz%FE%F7%EF%EE%EB%E8%EA%E9%E9%EB%EA%ED%EF%F3%F7%FC%7Cysookkkiijiklmqry%7B%7F%FD%F9%F7%F5%F4%F5%F5%F6%F7%F8%F8%FA%FD%FF%7Dxxtrsooonoposttw%7By%7C%7D~%7F%FE%FF%FE%FD%7F%FE%7F%FE%7F%FF%7F%FE%7F%7D~%7B%7Bzvwwvvwvuwvxwyxxyy%7B%7C%7C%7D%7D%7F%7F%7F%FD%7F%FF~%7D%7D%7B%7Bzywyvwvvvwxxy%7By~%7C%7D%7F%7D~~%7C~%FF%7D%7C%FF~%7F%7F~~%7Dz%7C%7Czxywwxxwxxwy%7Bx%7Bz%7C%7D%7C%FF~%7F%7F~%FF~%7F%7D~%7C%7D%7D~%7D%7D%7C%7B%7B%7B%7Bxzyzyzzz%7Cz%7D~%7C%7C~%7C~%7F~~%FF%7D%7F%7F%7D%7F~%7C%7C%7B%7By%7Byy%7B%7B%7B%7B~%7D%7D~~%7F%7F%FE%7F%FF
>
> CommandId: C44415

That seems kind of yucky.  How reproducible is it?


-- 
Matthew Fredrickson
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

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

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Test

[Matt Fredrickson]

Testing, 1, 2, 3.

-- 
Matthew Fredrickson
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

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

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Test

[Matt Fredrickson]

This is a test.

-- 
Matthew Fredrickson
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

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

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] FOSDEM

[Matt Fredrickson]

Hey All,

For any interested in potentially meeting up to talk about Asterisk
and other fun things, Ben Ford from Digium's Asterisk development team
and myself will be in Brussels for FOSDEM Feb 3-4.

I hope to see many of you there!

-- 
Matthew Fredrickson
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

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

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Recent Video Interview and Upcoming Webinar about Asterisk 15

[Matt Fredrickson]

Hey Everybody,

As a project, we would like to do a better job of getting additional
information about new developments in Asterisk to the community.  I
think this is something I have struggled with in the past (to some
extent) and would like to improve upon in the future.

For anybody interested, there's a fun 15 minute video interview
featuring Matt Jordan and myself discussing Asterisk 15 and what's new
with it.  It can be found at:

https://www.youtube.com/watch?v=0XSDOPftNpM=youtu.be

For those who enjoyed the video or would like to get a bit of a deeper
dive into Asterisk 15, the annual Asterisk 15: Under the Hood
presentation will be this Tuesday, December 5th.  It covers some of
the newer features in Asterisk 15 at a lower level and is intended for
a fairly technical audience.  It's usually very well attended.
Details and registration information can be found at:

http://bit.ly/2BnrpmF

-- 
Matthew Fredrickson
Digium, Inc. | Asterisk Project Lead
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

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

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Asterisk EOL Announcement

[Matt Fredrickson]

Dearly Beloved,

We have gathered here today to mourn the passing of a deeply regarded
branch of Asterisk - Asterisk 11.  As of today, it has officially
reached its end of life.  It was a good branch, having served 5 years
faithfully in the service of its users.  As far as history goes,
11.0.0 was born on November 28th 2012.  It had 1458 commits in its 5
year life, and some will try to use it even after its useful end of
life.  Now, mostly due to the fact that it is no longer with us,
Asterisk 11 will become one of the “great” releases of Asterisk,
joining the ranks of all the other “great” branches such as 1.0, 1.2,
1.4, 1.6, 1.8, and 10.  Please join with me now for a moment of
silence for it, as it passes to the great beyond.

In all seriousness, if you didn’t get the humor in the above message,
today is the day that Asterisk 11 officially goes end of life.  For
the last year, Asterisk 11 has been in security fix only mode, meaning
it stopped receive regular bug fixes a year ago, and has only received
security related fixes.  Today that all ends and not even security
fixes will going into that branch.  If you haven’t gotten off of it
yet, there is no better time than the present.

If you’re curious about the dates and times associated with life
cycles transitions on Asterisk branches, you can read more at
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions

Last of all, thanks to all of you who contribute to the Asterisk
project, whether it be bug reports, bug fixes, new feature
development, or helping other users by answering questions on the
mailing lists, forums, and other venues.  At the end of the day, it’s
the quality of the user and development community that make Asterisk
such a great project.

Best wishes.

-- 
Matthew Fredrickson
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

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

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Asterisk 14 Security Fix Only Mode

[Matt Fredrickson]

Hey all,

For those who may not be aware Asterisk 14 transitioned from bug fix mode
to security-fix-only mode a few weeks ago (Sept 26th). For those of you
that are still on this release, it's a good time to consider building an
upgrade plan for moving to 15.x.x.  I sincerely apologize for the late
notification.  Somehow, this was overlooked in the push to get 15.0.0
released.

You can read more about this process and the relevant dates on the wiki at
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions

Best wishes, and sorry again about the confusion on this.

-- 
Matthew Fredrickson
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

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

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk chan_sip registration attempts

[Matt Riddell (lists)]

Maybe the provider has added an extra gateway and it is not processing accounts 
correctly. 

If they had one before and now two then 40-60% registration fails would show 
that. 

Kind regards,

Matt

> On Oct 10, 2017, at 06:27, Dmitriy Ermakov <demoni...@gmail.com> wrote:
> 
> Hello!
> 
> Could you help me with Asterisk 11.21.2 and AsteriskNow platform.
> 
> The problem is:
> 
> My Asterisk PBX has SIP (chan_sip) trunk to provider.
> 
> Asterisk periodically loses trunk registratrion:
> 
> sip show registry:
> 
> Hostdnsmgr Username   Refresh State   
>  Reg.Time 
> X.X.X.X:5060N  105 Unregistered
>
> 
> This happens sometimes once per 4 hours, sometimes once per a week.
> 
> I don't see any patterns.
> 
> sip.conf:
> 
> registerattempts=0
> 
> registertimeout=20
> 
> peer confifuration:
> [-friend]
> disallow=all
> host=192.168.1.1
> defaultuser=
> fromuser=
> callerid=
> secret=
> type=friend
> qualify=yes
> allow=ulaw
> allow=alaw
> nat=no
> rtpkeepalive=10
> dtmfmode=rfc2833
> insecure=port,invite
> context=from-trunk-ISP1
> fromdomain=
> registration string:
> 
> register=:@/
> 
> where:
> 
>  is our ISP-provided phone number
> 
>  is our ISP-provided SIP secret
> 
>  is our ISP SIP server IP address
> 
> I don't have NAT between the ISP and my server (the ISP server IP address is 
> in Asterisk's sip.conf Localnet scope) but as I can see there is ISP's 
> routers between my Asterisk and the ISP SIP server.
> There is not any firewall between the ISP and my Asterisk. The firewall rules 
> on Asterisk host allow any traffic from host to the ISP and allow 5060/UDP 
> from ISP to Asterisk host.
> 
> 
> When I restart Asterisk I can see successful registration to the ISP. And it 
> works. I can make calls in any direction but after some time I have the 
> "Unregistered" status and calls don't work.
> 
> When I make tcpdump I can see about 40-60% failed REGISTER attempts (401 
> forbidden) and 100% failed OPTIONS attempts (is it qualify packets?) with 401 
> forbidden.
> 
> The interesting thing is: the last REGISTER packet from my Asterisk to the 
> ISP has "OK" response and there was not any REGISTER attempts after this 
> packet, only OPTIONS packets.
> The second interesting thing is: this ISP has been working for about 6 or 
> even 12 months before this problem happened.
> 
> 
> What should I check to understand and solve my problem?
> 
> 
> P.S. I'm sorry for my bad English(
> -- 
> Dmitriy Ermakov
> -- 
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> 
> Check out the new Asterisk community forum at: https://community.asterisk.org/
> 
> New to Asterisk? Start here:
>  https://wiki.asterisk.org/wiki/display/AST/Getting+Started
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

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

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk 15, Jack, streams, speech recognition… so many questions!

[Matt Riddell]

> On 22/09/2017, at 8:08 AM, Jonathan H <lardconce...@gmail.com> wrote:
> 
> Removing the "record in Asterisk/store as file/convert file/upload
> file  <> receive stream/save file/convert file/playback in Asterisk"
> part of the sequence would save vital seconds of silence and caller
> annoyance.


At least in older versions you can use EAGI to get a handle to the audio 
stream.  You can then pipe that stream to something like bluemix using Node.js 
and have a handle to the incoming recognition in realtime too.

--
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)

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

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

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Asterisk 15 Beta Released

[Matt Fredrickson]

It is with great pleasure I wish to inform you of the first beta
release of the new Asterisk 15 branch. It's a very exciting time to be
a user of Asterisk! Asterisk 15 is arguably the biggest release of
Asterisk that has happened in the last 10 or so years. There has been
a lot of work done in the Asterisk core to better support newer
multi-stream video and WebRTC related technologies.  For those who are
interested, much of this will be covered in blog posts at
http://blogs.asterisk.org/ over the next month or two.

Typically, when a new major branch of Asterisk is created (13, 14,
15...), there are a few months of testing on the new branch that
occurs prior to release in order to find regressions and other issues
that may cause a first official release from the branch to be dead on
arrival for a significant number of users. With today's release of
15.0.0-beta1, this process has begun. Please feel free to start
testing this version of Asterisk in as many adverse environments as
possible. Any bugs should be reported on the Asterisk issue tracker at
https://issues.asterisk.org/

As a side note, due to many of the core changes in the 15 branch that
have been made since Asterisk 14 was released, it has been decided
that Asterisk 15 will not be an LTS release. For those of you who are
not familiar with the differences between LTS versus standard
releases, you can find more information here [1].

Thanks to all the many Asterisk community members for providing so
much help and support to make Asterisk the great open source project
that it is.

P.S. Binary codecs and other modules distributed by Digium are not
immediately available for 15.0.0-beta1, but should be shortly.

Best wishes to all, and happy testing!

[1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions

-- 
Matthew Fredrickson
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

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

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Moving call DAHDI from channel X to Y.

[Matt Fredrickson]

On Sun, Jul 30, 2017 at 8:34 PM, Daniel Harper  wrote:
> I am seeing the in the asterisk logs that channels (PRI ISDN)  are
> being moved ..
>
> [Jul 29 16:31:48] VERBOSE[16125] logger.c: -- Moving call
> (DAHDI/57-1) from channel 57 to 58.
>
> I then see the moved channels with a "0:" in front of it.
>
> [Jul 29 16:31:48] VERBOSE[26691] logger.c: -- Hungup 'DAHDI/0:58-1'
>
> Any ideas why this could be happening?
>
> I believe these messages are coming from chan_dahdi.c and the
> "pri_fixup_principle" function.
>

This is a normal thing that can happen when doing B-channel selection
as a part of the call setup process.  If one side disagrees with the
initial choice for B-channel, the other side can request that it be
moved in the reply.  It's more likely to happen on busy PRIs with
bidirectional (ingress and egress) traffic.

Hope that helps!

-- 
Matthew Fredrickson
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

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

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] OT: Want to capture all SIP messages

[Matt Riddell]

> On 31/05/2017, at 3:52 PM, Steve Edwards <asterisk@sedwards.com> wrote:
> 
> On Wed, 31 May 2017, Daniel Tryba wrote:
> 
>> On Wed, May 31, 2017 at 01:39:25PM -0700, Steve Edwards wrote:
>>>> What bugs you about the output format?
>>> 
>>> It's been a while, but as I recollect, it included the date/timestamp in the
>>> file name of the 'ring buffer' which meant that each time the host was
>>> rebooted, dumpcap didn't know the files from the previous run should be
>>> deleted when they 'aged out.'
>> 
>> Solvable by by writing a cleanup script that deletes files over a
>> specific age, just a basic find in the daily crontab:
>> find /path/to/captures -type f -name 'pattern*' -mtime +X -exec rm {} \;
> 
> Been there, done that. Just 1 more thing for me to maintain :)


Easier just to use logrotate no?

That's what it's designed for.

--
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)

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

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

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] softphone instead of desktop phones

[Matt Riddell (lists)]

I use Bria on all of the above. 

Kind regards,

Matt

> On Apr 29, 2017, at 10:35 AM, Thomas <thomasit...@gmail.com> wrote:
> 
> Hello,
> Iam lookong for an Softphone for iPhor oder Android smartphone using togehter 
> with an headset.
> I tried Zoiper and CSipSimple but quality was bad compared to an desktop SIP 
> phone.
> 
> Is there an better softphone?
> 
> Or are there softphone solutions for PC desktop MAC or Android with an 
> headset?
> I want to save cost for desktop phones.
> 
> thanks Thomas
> 
> 
> -- 
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> 
> Check out the new Asterisk community forum at: https://community.asterisk.org/
> 
> New to Asterisk? Start here:
>  https://wiki.asterisk.org/wiki/display/AST/Getting+Started
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

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

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Asterisk 11 EOL 6 Month Notice

[Matt Fredrickson]

Greetings,

This is your friendly 6 month warning that Asterisk 11 will be
reaching an official end of life state on October 25, 2017.  As many
of you know, for the past 6 months Asterisk 11 has been in security
fix only mode.  This means it currently does not receive bug fixes,
but it does receive applicable security fixes and will continue to do
so for the next 6 months.

If you are still running Asterisk 11, this is a great time to start
preparing for a move to the latest LTS branch, 13, as a forward
migration path.  You can read more about Asterisk's branch and
versioning policy as well as relevant version sunset information on
the Asterisk wiki [1].

Best wishes to you all, and happy upgrading. :-)

[1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions

-- 
Matthew Fredrickson
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

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

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk 13.15.0, webrtc, Google chrome 58 beta and "bad media description"

[Matt Fredrickson]

On Sat, Apr 8, 2017 at 7:23 AM, Dan Jenkins  wrote:

>
> On Fri, Apr 7, 2017 at 9:44 PM, Teijo  wrote:
>
>> Hello,
>>
>> I've been using webrtc (Jscommunicator) with Asterisk occasionally. Only
>> problem until now which remained was that if dtls_rekey was set to the
>> value other than 0, call hanged up when using chrome after the time where
>> dtls_rekey was set.
>>
>> I suppose that "bad media description" shown in Chrome's window which
>> causes call to fail, has appeared with Chromes newer versions (currently 58
>> beta installed) or with Asterisk 13.15.0. Audio codec I'm using is Opus.
>>
>> Has somebody else encountered this problem, or more better resolved it?
>>
>> Best regards,
>>
>> Teijo
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> Check out the new Asterisk community forum at:
>> https://community.asterisk.org/
>>
>> New to Asterisk? Start here:
>>  https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
> Hi Teijo
>
> Take a read of https://nimblea.pe/monkey-business/2017/01/19/webrtc-
> asterisk-and-chrome-57/ :)
>

13.15.0 should address rtcp-mux issues.

If there are still issues outstanding, it might be worth reporting a bug on
issues.asterisk.org.

Best wishes :-)

-- 
Matthew Fredrickson
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

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

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] 100% CPU after upgrade.

[Matt Fredrickson]

On Mon, Apr 3, 2017 at 4:45 PM, Mike Diehl  wrote:
> Those are all rational questions, so here we go:
>
> We upgraded from 11.x, though the system was a backup server, so it was never
> actually used.
>
> The system is a 2.4Gh quad-core Xenon with 4G of RAM, so it should have plenty
> of power for what I'm asking it to do.  The system is configured via RT using
> a local Mysql database.
>
> We only use the native SIP channel driver at this time.
>
> I honestly don't see any reason for this server to eat 100% of it's cpu, and
> am hesitant to roll it out to production until I understand why it is.

I don't either.  Is there any Asterisk logging that indicates
something that might be going on?  If you can't see anything, try
increasing the core debug level and core verbose level (core set
verbose 10, core set debug 10) at the Asterisk CLI and see if you get
anything more out of logging to see what's going on.

-- 
Matthew Fredrickson
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

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

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] 100% CPU after upgrade.

[Matt Fredrickson]

One thing you didn't mention was what version you previously upgraded
from...  Also, more information about the system in general would
help.  (Endpoints, is it realtime or flat file configured, if
realtime, what type of database, what channel drivers (SIP or PJSIP,
and others).

Matthew Fredrickson

On Fri, Mar 31, 2017 at 12:08 PM, Mike Diehl  wrote:
> Hi all,
>
> I've upgraded to Asterisk 13.14.0 and now I'm seeing that Asterisk is using 
> 100% CPU.
>
> I have one AMI agent connected that is acting rationally.  I've got a hand 
> full of SIP (RT) registrations.  There is no other call activity.
>
> I've tried to unload various modules; nothing resolved the issue.
>
> Any suggestions?
>
> --
> Mike Diehl
>
>
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at: https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users



-- 
Matthew Fredrickson
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

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

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Bounty on Google Voice

[Matt Fredrickson]

On Wed, Mar 29, 2017 at 11:45 AM, Saint Michael  wrote:
> The channel motif and res_xmpp do not work. But there is one company that
> does make it work and charges $US 6 for a lifetime connection to your own
> free Google Voice number, from SIP. I wonder if anybody would be able to fix
> Asterisks libraries so people of low income would not have to pay a third
> party for this basic translation service.

Hey,

Sorry to hear about your difficulty with this code.  At this time, I'm
not aware of an active maintainer/owner of those modules.  As I'm sure
you're already aware, Asterisk is an open source project worked on by
people with different interests and motivations.  Some work on it
purely for the fun, others work on it because they have a business
interest of some sort.  Bug bounties are a great way to get the
attention of people in both categories, and are usually posted to the
asterisk-dev mailing list.  You can learn more about the guidelines
prior to posting at this wiki page:

https://wiki.asterisk.org/wiki/display/AST/Asterisk+Bug+Bounties

Hope that helps a bit, and best wishes in getting your bug fixed.

-- 
Matthew Fredrickson
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

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

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] UniMRCP and Asterisk 14

[Matt Fredrickson]

On Thu, Mar 23, 2017 at 8:27 PM, Richard Kenner  wrote:
> When I look at the lastest UniMRCP manual, they only mention as high as
> Asterisk 13.  Does anybody know if I need to do anything to allow it
> to work on Asterisk 14 and, if so, what that is?

I can't speak for the MRCP guys, but from a difference perspective,
swapping MRCP from Asterisk 13 to Asterisk 14 shouldn't be too
difficult.  Most of the changes between the two shouldn't affect most
people's use cases, including projects such as MRCP.  I'd definitely
check with their discussion forums though, since it seems that they
don't monitor the asterisk-users mailing lists.

-- 
Matthew Fredrickson
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

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

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Manager events showing in CLI

[Matt Fredrickson]

Try doing a `core set debug 0` at the Asterisk CLI.  That should
disable it.  Or remove debug from your console output in logger.conf.

Best wishes,
Matthew Fredrickson

On Sun, Mar 26, 2017 at 5:35 PM, Telium Technical Support
 wrote:
> I did that too – no debug related settings in there!  That’s why I’m
> stumped.
>
>
>
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Victor
> Villarreal
> Sent: Sunday, March 26, 2017 2:06 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> 
> Subject: Re: [asterisk-users] Manager events showing in CLI
>
>
>
> Ok,
>
>
>
> Please, check your manager.conf and logger.conf for any clue about debugging
> options, into the Asterisk configuration directory.
>
>
>
> El 26 mar. 2017 14:52, "Telium Technical Support" 
> escribió:
>
> I tried that but it had no effect.  Still see things like:
>
>
>
> [2017-03-26 13:49:39] DEBUG[2088]: manager.c:5693 match_filter: Examining
> AMI event:
>
> Event: SuccessfulAuth
>
> Privilege: security,all
>
> EventTV: 2017-03-26T13:49:39.407-0400
>
> Severity: Informational
>
> Service: SIP
>
> EventVersion: 1
>
> AccountID: 221essionID: 0x7fa0cc005cc8
>
> LocalAddress: IPV4/UDP/192.168.67.4/5060
>
> RemoteAddress: IPV4/UDP/192.168.67.26/5060
>
> UsingPassword: 1
>
>
>
>
>
> [2017-03-26 13:49:39] DEBUG[1882]: chan_sip.c:9196 __find_call: = Looking
> for  Call ID: 280f68000ff289291b366a1242530ce8@192.168.67.4:5060 (Checking
> To) --From tag as494dfc4b --To-tag 4155795028
>
> [2017-03-26 13:49:39] DEBUG[1882]: chan_sip.c:4419 __sip_ack: Stopping
> retransmission on '280f68000ff289291b366a1242530ce8@192.168.67.4:5060' of
> Request 102: Match Found
>
> [2017-03-26 13:49:39] DEBUG[1882]: chan_sip.c:6725 sip_destroy: Destroying
> SIP dialog 280f68000ff289291b366a1242530ce8@192.168.67.4:5060
>
> [2017-03-26 13:49:39] DEBUG[1882]: chan_sip.c:4275 __sip_autodestruct: Auto
> destroying SIP dialog 'cbf5d92f6844702b'
>
> [2017-03-26 13:49:39] DEBUG[1882]: chan_sip.c:6725 sip_destroy: Destroying
> SIP dialog cbf5d92f6844702b
>
> [2017-03-26 13:49:39] DEBUG[2088]: manager.c:6138 process_message: Running
> action 'Command'
>
> [2017-03-26 13:49:39] DEBUG[1951]: manager.c:6138 process_message: Running
> action 'Command'
>
>
>
> cli> manager set debug off
>
>
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users



-- 
Matthew Fredrickson
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

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

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Call List Campaign to an IVR

[Matt Riddell]

> On 6/02/2017, at 1:28 PM, Tech Support <aster...@voipbusiness.us> wrote:
> 
> I remember doing the testing and two calls going out at the same time 
> don’t actually have to go out at the *exact* same time. The remote end will 
> pick up one of the two calls, but there is no guarantee which one it will be. 
> Also, if you let the first call ring too long, yes, the second call will go 
> to voicemail,  but the first call will start ringing, which is something we 
> wanted to avoid.
> John
> 

That's the benefit of doing the & thing.

The instant one of them goes to voicemail the other will stop ringing.

Typing calls this happens in a few ms (after post dial delay).

Because they are both going out at the same time with the same provider this is 
super quick.

--
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)

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

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

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Call List Campaign to an IVR

[Matt Riddell]

> On 6/02/2017, at 12:30 PM, Chris Bagnall <aster...@lists.minotaur.cc> wrote:
> 
> On 6/2/17 5:24 pm, Tech Support wrote:
>> Basically,
>> two calls are made.
> ...
>> When the first call is made for
>> such a short period, the remote end still goes off hook, but the call will
>> end before it starts to ring. Then, halfway through the first call, a second
>> call is made. Since the remote end is off hook from the first call, the
>> second call will get sent to voicemail and the message is played there.
> 
> Am I right in thinking call waiting isn't a thing on US mobile networks then? 
> In the UK, call waiting is pretty standard, and almost universally enabled by 
> default on mobile networks. AIUI the same is true for much of Europe.

Heh nah, call waiting still only allows one ringing event to happen at a time.

If your phone is ringing and another comes in it will go to voicemail.

Call waiting allows you to get notified of a call if you're already on one, not 
multiple incoming at the same time.

--
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)

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

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

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Call List Campaign to an IVR

[Matt Riddell (lists)]

Not really, doing the way below you don't even have to worry about it. They 
both go out at the same instant and as soon as it hits voicemail it disconnects 
the other leg. 

If you wanted you could leave it ringing for twenty minutes and it would still 
have the same effect. 

Kind regards,

Matt

> On Feb 6, 2017, at 12:29 PM, Tech Support <aster...@voipbusiness.us> wrote:
> 
> That's the basics, but you have to nail the timing just right. The timing is
> really important to do it the right way.
> 
> 
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
> Sent: Monday, February 06, 2017 12:25 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Call List Campaign to an IVR
> 
> 
>> On Mon, 6 Feb 2017, Tech Support wrote:
>> 
>>  We were able to develop a feature to send the call to voicemail
> about 90% of the time. That way, an end user could (1) not be bothered by
> having to answer the call, (2)
>>  delete the message without listening to it, or (3) listen to the
> message when it was most convenient for them. That way, they were in control
> and things were done on
>>  their terms.
> 
>> On 6/02/2017, at 11:34 AM, Steve Edwards <asterisk@sedwards.com>
>> wrote:
>> 
>> Love the idea. How?
> 
>> On Mon, 6 Feb 2017, Matt Riddell wrote:
>> 
>> exten => 
>> _X.,1,Dial(SIP/0111${EXTEN}@myprovider/1${EXTEN}@myprovider,3)
> 
> Amazing. Who knew?
> 
> So how/why does this work?
> 
> I see 2 calls going out to my cell. Does the first 'busy out' my number at
> my cell provider so the second goes straight to VM? What part does the
> '0111' play?
> 
> --
> Thanks in advance,
> -
> Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
> https://www.linkedin.com/in/steve-edwards-4244281
> 
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> 
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
> 
> New to Asterisk? Start here:
>  https://wiki.asterisk.org/wiki/display/AST/Getting+Started
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> 
> -- 
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> 
> Check out the new Asterisk community forum at: https://community.asterisk.org/
> 
> New to Asterisk? Start here:
>  https://wiki.asterisk.org/wiki/display/AST/Getting+Started
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

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

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Call List Campaign to an IVR

[Matt Riddell]

> On 6/02/2017, at 12:24 PM, Steve Edwards <asterisk@sedwards.com> wrote:
> 
> 
>> On Mon, 6 Feb 2017, Tech Support wrote:
>> 
>>  We were able to develop a feature to send the call to voicemail about 
>> 90% of the time. That way, an end user could (1) not be bothered by having 
>> to answer the call, (2)
>>  delete the message without listening to it, or (3) listen to the 
>> message when it was most convenient for them. That way, they were in control 
>> and things were done on
>>  their terms.
> 
>> On 6/02/2017, at 11:34 AM, Steve Edwards <asterisk@sedwards.com> wrote:
>> Love the idea. How?
> 
> On Mon, 6 Feb 2017, Matt Riddell wrote:
> 
>> exten => _X.,1,Dial(SIP/0111${EXTEN}@myprovider/1${EXTEN}@myprovider,3)
> 
> Amazing. Who knew?
> 
> So how/why does this work?
> 
> I see 2 calls going out to my cell. Does the first 'busy out' my number at my 
> cell provider so the second goes straight to VM? What part does the '0111' 
> play?


Yep, if you have two calls going to the same number at the same time the second 
will be busy.

Meaning it will go to voicemail.

Meaning it will be answered.

Meaning the other leg will be dropped.

The 0111 and the 1 is just so that Asterisk doesn't think that both calls are 
to the same destination and then only make one call.

--
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)

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

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

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Call List Campaign to an IVR

[Matt Riddell]

> 
> On 6/02/2017, at 11:34 AM, Steve Edwards <asterisk@sedwards.com> wrote:
> 
> On Mon, 6 Feb 2017, Tech Support wrote:
> 
>> We were able to develop a feature to send the call to voicemail about 90% of 
>> the time. That way, an end user could (1) not be bothered by having to 
>> answer the call, (2) delete the message without listening to it, or (3) 
>> listen to the message when it was most convenient for them. That way, they 
>> were in control and things were done on their terms.
> 
> Love the idea. How?


exten => _X.,1,Dial(SIP/0111${EXTEN}@myprovider/1${EXTEN}@myprovider,3)

--
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)

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

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

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Issue with handling of 480 DND

[Matt Fredrickson]

Response inline.

On Fri, Jan 6, 2017 at 12:47 PM, Markus Weiler 
wrote:

> Nobody any idea?
>
> It would be really helpful,
>
> Markus
>
>
>
>
> Am 06.01.2017 um 12:07 schrieb Markus Weiler:
>
>> Hi List,
>>
>> we're calling a sip phone from our Asterisk Server, and try to add logic
>> depending on the dialstatus
>>
>> Stripped down example;
>>
>> exten = 494X,n,Dial(SIP/4120089,15,w)
>> exten = 494X,n,Goto(98-${DIALSTATUS},1)
>> exten = 494X,n,Hangup()
>>
>>
>> .
>> exten = 98-BUSY,1,NoOp(Busy)
>> exten = 98-BUSY,n,ExecIf($["${Voicemail}" =
>> "1"]?Playback(/home/4120/mitarbeiter/ab))
>> 
>> exten = 98-NOANSWER,1,NoOp(noanswer)
>> exten = 98-NOANSWER,n,ExecIf($["${Voicemail}" =
>> "1"]?Playback(/home/4120/mitarbeiter/ab))
>> 
>>
>> Íf the phone call times out, the call is sent to 98-NOANSWER and then
>> answered as expected.
>> If the User presses DND on his phone the call is sent to 98-BUSY which
>> is identical but then the call is hung up. This behaviour is
>> unexpected/unwanted.
>>
>> We tried to figure out what the difference is and think it's how
>> Asterisk handles the "480 Do Not Disturb" from the phone
>> (xxx.xxx.xxx.xxx).
>> It is passed to our main incoming server (zzz.zzz.zzz.zzz) as "181 call
>> is being forwarded".
>>
>> Is this a bug or a feature? :-) How could we handle this correctly?
>>
>
I believe that this is a consequence of the fact that when chan_sip
receives a "480 Do Not Disturb" it also queues a redirecting_update frame
on the channel. My guess is that the redirecting update is probably
triggering the "181 Call is being forward as well".  If you add the 'I'
flag to your dial, I believe it should suppress bridging of the redirection
information and I would think that would also cause the 181 not to be
sent.  If it's not that flag, I'd check the documentation for app_dial, as
I'm pretty sure there's a flag that should suppress that redirection from
from being bridged to the calling channel.

Hope that helps.

-- 
Matthew Fredrickson
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

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

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Asterisk 14 web broadcast

[Matt Fredrickson]

Hey All,

Slight interlude from your regularly scheduled programming.

For any interested, I will be giving a web broadcast today about
Asterisk 14 and what's new with Asterisk since the 13 release.  For
those of you that aren't aware, I'm responsible for day to day
management of the Asterisk project now that Matt Jordan has been moved
into the CTO role at Digium.

You can get info about it at:

http://bit.ly/2gDkFrh

It will be live today at 8AM, 2PM, and 9PM CDT.

Hope to see many of you there!

-- 
Matthew Fredrickson
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

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

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Subscribe to events via ARI from node.js without sending to Stasis

[Matt Riddell]

> On 27/11/2016, at 6:44 PM, Joshua Colp <jc...@digium.com> wrote:
> 
> On Wed, Nov 23, 2016, at 06:41 PM, Matt Riddell wrote:
>> 
>> There doesn't appear to be a way to monitor general Asterisk events like
>> you can in the Asterisk manager without polling for channel statuses or
>> sending the channels to the Stasis app and recreating the logic of the
>> Queue application.
> 
> You can subscribe to events for everything, but ARI events are a subset
> of what are available over AMI so you may not get what you want. Events
> raised by applications for example would not be present. You can
> subscribe to all channels by subscribing to "channel:" using the
> /applications//subscription resource, and "bridge:" for all
> bridges.


 is the stasis app name or the Asterisk application name?

--
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)

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

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

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Subscribe to events via ARI from node.js without sending to Stasis

[Matt Riddell]

Hi,

I'm writing a node.js backend to pass events via a websocket to a CRM.

Basically what I want to do is notice when things happen (i.e. new channel, new 
bridge etc) without sending the channels to the Stasis app.

The channels I'm interested in are agents who are in a queue only because they 
are in a realtime MySQL database for the queue_member_table.

There doesn't appear to be a way to monitor general Asterisk events like you 
can in the Asterisk manager without polling for channel statuses or sending the 
channels to the Stasis app and recreating the logic of the Queue application.

Is this a correct assumption?

--
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)

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

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

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Force hangup not working on stuck channel

[Matt Fredrickson]

Also, it looks like in
https://issues.asterisk.org/jira/browse/ASTERISK-21762 there might be
a workaround (see the last comment at the bottom).

Matthew Fredrickson

On Fri, Nov 4, 2016 at 2:01 PM, Matt Fredrickson <cres...@digium.com> wrote:
> On Thu, Nov 3, 2016 at 11:16 AM, Carlos Chavez <cur...@telecomabmex.com> 
> wrote:
>> I am unable to force a hangup on a channel that has been stuck for over two
>> days:
>>
>> IAX2/from-CD-11006   oficina  27701 Up  Dial
>> IAX2/to-CD/2883   3467130007  46:24:59 Sotelo  Sotelo
>> IAX2/to-CD-20713
>>
>> I have tried "hangup request IAX2/from-CD-11006" several times but no joy.
>> I also see the following in the CLI:
>>
>> [Nov  3 10:05:54] WARNING[2879]: chan_iax2.c:4936 handle_call_token: Too
>> much delay in IAX2 calltoken timestamp from address X.X.X.X
>>
>> This is an IAX2 trunk between two Asterisk 1.8 servers (I know it is old but
>> new client so haven't had time yet to upgrade to 13).  Because this channels
>> is stuck
>>  all other calls between servers are not working.  The only way I have found
>> to resolve the problem is to stop and restart Asterisk.  This is obviously a
>> great inconvinience so is there a way for force iax to unload even if there
>> are channels in use?  Or any other way to kill these stubborn channels?
>
> If doing a soft hangup on them doesn't work, the only other way I know
> to do it is to restart Asterisk.  Sorry about the bad news :-(
>
> There's a part of me that's curious as to why the channel is stuck,
> but there's another part of me that says "1.8... run away quickly".
>
> You could try to replicate it with a modern branch (i.e. 13/14) and
> see if it still exists.  At the very least that'd leave you the option
> of posting a bug on the issue tracker about it.  Also, a lot of bugs
> have been fixed since 1.8, so it's quite possible that this issue is
> resolved as well.
>
> --
> Matthew Fredrickson
> Digium, Inc. | Engineering Manager
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA



-- 
Matthew Fredrickson
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

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

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Force hangup not working on stuck channel

[Matt Fredrickson]

On Thu, Nov 3, 2016 at 11:16 AM, Carlos Chavez  wrote:
> I am unable to force a hangup on a channel that has been stuck for over two
> days:
>
> IAX2/from-CD-11006   oficina  27701 Up  Dial
> IAX2/to-CD/2883   3467130007  46:24:59 Sotelo  Sotelo
> IAX2/to-CD-20713
>
> I have tried "hangup request IAX2/from-CD-11006" several times but no joy.
> I also see the following in the CLI:
>
> [Nov  3 10:05:54] WARNING[2879]: chan_iax2.c:4936 handle_call_token: Too
> much delay in IAX2 calltoken timestamp from address X.X.X.X
>
> This is an IAX2 trunk between two Asterisk 1.8 servers (I know it is old but
> new client so haven't had time yet to upgrade to 13).  Because this channels
> is stuck
>  all other calls between servers are not working.  The only way I have found
> to resolve the problem is to stop and restart Asterisk.  This is obviously a
> great inconvinience so is there a way for force iax to unload even if there
> are channels in use?  Or any other way to kill these stubborn channels?

If doing a soft hangup on them doesn't work, the only other way I know
to do it is to restart Asterisk.  Sorry about the bad news :-(

There's a part of me that's curious as to why the channel is stuck,
but there's another part of me that says "1.8... run away quickly".

You could try to replicate it with a modern branch (i.e. 13/14) and
see if it still exists.  At the very least that'd leave you the option
of posting a bug on the issue tracker about it.  Also, a lot of bugs
have been fixed since 1.8, so it's quite possible that this issue is
resolved as well.

-- 
Matthew Fredrickson
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

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

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] What's the smallest, lightest Asterisk you can build? Does size even matter?

[Matt Fredrickson]

On Tue, Nov 1, 2016 at 6:00 PM, Jonathan H  wrote:
> All I need is PJSIP, ulaw, alaw, wav, astdb and all the dialplan functions.
>
> I don't need any other DB layer, I have no hardware, and I was wondering
> what the smallest build possible was.
>
> I experimented, but everything relied on other things. And then I
> wondered... is there actually any point? Is there anything to be gained?
>
> Will it matter more when there are lots of concurrent calls, or should I
> just not worry, leave all the options in makemenu, make it easy on myself
> and build the full thing each time?

For most modules there isn't a big point to disabling them as there
isn't an lot of ongoing CPU activity for a module not being used.
That being said, there are some things you can disable that can
improve your performance, but they're going to be application
dependent.  I believe that disabling CDRs, for example, can make a big
difference on heavily loaded systems and some people don't use them.
CDRs take some amount of work at the end to stitch together the notion
of a call from the various call related events that occur.  HEP also
is a big offender.  CELs and AMI can also make some difference when
disabled, but nothing on the order of CDRs and HEP.

Hope that helps!

-- 
Matthew Fredrickson
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

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

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] RS485 Audio device

[Matt Fredrickson]

On Fri, Oct 28, 2016 at 7:09 PM, Jerry Geis  wrote:
> Hi All,
>
> Is there any devices or pair of devices that do audio over RS485
> and then convert to SIP for us in asterisk?
> Of course a speaker and push button at the other end.
>
> Is there anything like that out there?

Ok, I'll bite.  How does one do audio over RS485?

I've never worked with RS485, but from some brief googling it looks
like it's a fancy version of RS232.  I'm not sure where you'd get
(analog) audio from on RS232.

-- 
Matthew Fredrickson
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

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

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Asterisk 11 - Security Fix Only Notice

[Matt Fredrickson]

Hey All,

This is a friendly notice that as of today Asterisk 11 has entered
security fix only mode.  From this point onward additional releases of
Asterisk 11 will no longer be made unless there is a security fix
being applied to the branch.  Users of Asterisk 11 are encouraged to
move to one of the newer major versions, Asterisk 13 (LTS) or Asterisk
14, as soon as possible.

For more information on Asterisk versions and their supported lifetimes,
please see the following wiki page:

https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions

Thank you for your continued support of Asterisk!

-- 
Matthew Fredrickson
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
  http://www.asterisk.org/community/astricon-user-conference

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Got bitten by the 255 char variable limit - how best to work around it?

[Matt Fredrickson]

On Sat, Oct 22, 2016 at 8:05 PM, Jonathan H  wrote:
> I loop through a list in Asterisk which is generated by a Python AGI
> and I've just been bitten by a variable limit I didn't realise existed
> before.
>
> The only way I can think of working around this is to get Python to
> write the list to file, then do a FILE_COUNT_LINE to get the number of
> items (needed first) and then iterate through them by doing FILE to
> read one line at a time.
>
> Would rather stay away from polluting ASTDB, and don't want to install
> MySQL as I don't need it for anything else.
>
> Ideally, though, is there a workaround for the variable limit? Thanks!

I was curious about this and started looking through the code to
remember how this all worked.  I don't see anything specifically that
would truncate setting a channel variable internally, but perhaps it's
a limitation due to how AGI is processed or buffers used for
parsing/interpreting there.  There are a number of fixed sized buffers
used in the AGI parsing code.  I'm curious, what version of Asterisk
are you using?

I'm not aware of any workarounds other than those you mentioned.  If
you wanted to get really into it, you could poke around in
res/res_agi.c and see if you can figure out which buffer is limiting.
YMMV there though.

-- 
Matthew Fredrickson
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
  http://www.asterisk.org/community/astricon-user-conference

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Audio when card is in condition yellow

[Matt Fredrickson]

Usually a card is supposed to send yellow alarm (so it's transmitted)
when it detects LOS (loss of signal) on the T1, or essentially a red
alarm condition is detected.  So if yellow is being sent, it means
that at least one end is not able to sync up the line, which means
you'll have junk/garbled audio potentially.

Hope that helps,
Matthew Fredrickson

On Fri, Oct 21, 2016 at 1:51 PM, Jerry Geis  wrote:
> With asterisk 11.23.0 I have about 121 SIP devices connected.
> normally things sound fine when speaking a message on these devices (using
> conference bridge).
>
> Currently the TE122 card is in condition yellow normally it is not.
> I sent a message to all devices and it was garbled on many of them.
> Is this a result of card in COND YELLOW?
>
> If so what can I do about it while the card is in cond yellow.
> The other end is currently down, no ETA on coming back.
>
> Thanks,
>
> Jerry
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
>   http://www.asterisk.org/community/astricon-user-conference
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users



-- 
Matthew Fredrickson
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
  http://www.asterisk.org/community/astricon-user-conference

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] queue_log/cel sqlite

[Matt Fredrickson]

On Thu, Oct 20, 2016 at 9:45 AM, marek cervenka <cerva...@gmail.com> wrote:
>
> Dne 20/10/2016 v 16:32 Matt Fredrickson napsal(a):
>>
>> On Thu, Oct 20, 2016 at 4:50 AM, marek cervenka <cerva...@gmail.com>
>> wrote:
>>>
>>> i tested this
>>>
>>> # cat /etc/asterisk/extconfig.conf
>>> [settings]
>>> queue_log => sqlite3,cdrDb
>>>
>>> # cat /etc/asterisk/res_config_sqlite3.conf
>>> [cdrDb]
>>> dbfile = /var/lib/asterisk/realtime.sqlite3
>>>
>>> sqlite3 /var/lib/asterisk/realtime.sqlite3
>>>
>>> CREATE TABLE "queue_log" ("time" TEXT, "data1" TEXT, "data2" TEXT,
>>> "data3"
>>> TEXT, "data4" TEXT, "data5" TEXT, "event" TEXT, "agent" TEXT, "queuename"
>>> TEXT, "callid" TEXT);
>>>
>>> and it works
>>>
>>> sqlite> select * from queue_log;
>>> 2016-10-20 11:40:36.628804||QUEUESTART|NONE|NONE|NONE
>>> 2016-10-20 11:40:36.690313||CONFIGRELOAD|NONE|NONE|NONE
>>>
>>> column types needs modification to something more appropriate
>>>
>>> can someone with confluence access ad info to
>>>
>>>
>>> https://wiki.asterisk.org/wiki/display/AST/Realtime+Database+Configuration ?
>>
>> Which info are you referring to?  The table schema?
>>
>
> ideally add "correct" sql schema for sqlite to asterisk repo and link it to
>
> https://wiki.asterisk.org/wiki/display/AST/Realtime+Database+Configuration
>
>
> it was hard for me to find if queue_log can be logged with sqlite. imho it
> will be usefull document the example configuration for others
> but i'm not sure where is the best place
> maybe https://wiki.asterisk.org/wiki/display/AST/Queue+Logs ?

My suggestion would be to add a comment to the page with your proposed
changes.  That would be the best place to start, for the next time
someone works on that page.

-- 
Matthew Fredrickson
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
  http://www.asterisk.org/community/astricon-user-conference

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] queue_log/cel sqlite

[Matt Fredrickson]

On Thu, Oct 20, 2016 at 4:50 AM, marek cervenka  wrote:
> i tested this
>
> # cat /etc/asterisk/extconfig.conf
> [settings]
> queue_log => sqlite3,cdrDb
>
> # cat /etc/asterisk/res_config_sqlite3.conf
> [cdrDb]
> dbfile = /var/lib/asterisk/realtime.sqlite3
>
> sqlite3 /var/lib/asterisk/realtime.sqlite3
>
> CREATE TABLE "queue_log" ("time" TEXT, "data1" TEXT, "data2" TEXT, "data3"
> TEXT, "data4" TEXT, "data5" TEXT, "event" TEXT, "agent" TEXT, "queuename"
> TEXT, "callid" TEXT);
>
> and it works
>
> sqlite> select * from queue_log;
> 2016-10-20 11:40:36.628804||QUEUESTART|NONE|NONE|NONE
> 2016-10-20 11:40:36.690313||CONFIGRELOAD|NONE|NONE|NONE
>
> column types needs modification to something more appropriate
>
> can someone with confluence access ad info to
>
> https://wiki.asterisk.org/wiki/display/AST/Realtime+Database+Configuration ?

Which info are you referring to?  The table schema?

Matthew Fredrickson


>
>
> is there somebody using it in production?
> thanks
>
> Dne 20/10/2016 v 10:16 marek cervenka napsal(a):
>
>> hi,
>>
>> is it possible log cel/queue_log to sqlite?
>>
>> via odbc?
>>
>> any experience?
>>
>> marek
>>
>>
>>
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
>  http://www.asterisk.org/community/astricon-user-conference
>
> New to Asterisk? Start here:
>  https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users



-- 
Matthew Fredrickson
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
  http://www.asterisk.org/community/astricon-user-conference

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Streaming for ASR

[Matt Riddell]

> On 19/10/2016, at 2:32 AM, Luca Pradovera <luca.pradov...@gmail.com> wrote:
> 
> Would UnicastRTP be able to output u-law frames directly? If so, I think that 
> is all I need.

Joshua Colp did a great writeup that may work for your situation:

http://www.joshua-colp.com/broadcasting-asterisk-conferences/ 
<http://www.joshua-colp.com/broadcasting-asterisk-conferences/>

I'm still working on mine :-)

--
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)

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
  http://www.asterisk.org/community/astricon-user-conference

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Streaming for ASR

[Matt Riddell]

> On 17/10/2016, at 4:07 PM, Joshua Colp <jc...@digium.com> wrote:
> 
> Matt Riddell wrote:
>> 
>>> On 17/10/2016, at 3:43 PM, Luca Pradovera <luca.pradov...@gmail.com
>>> <mailto:luca.pradov...@gmail.com>> wrote:
>>> 
>>> I have been working on designs for two different projects, where both
>>> of them would need to use the IBM Watson streaming ASR service.
>>> 
>>> Would it be possible to write out the audio frames as they get
>>> recorded? Watson supports 16 bit signed little endian audio, IIRC, but
>>> there are a few other raw formats available.
>> 
>> That's the main problem I was discussing at AstriDevCon. There's no way
>> currently to stream the frames to a service.
>> 
>> At the moment I'm saving to a file and then sending that file once
>> recording has finished.
> 
> The UnicastRTP channel driver allows you to send RTP to a specific target 
> address with media. Combined with Chanspy (or Snoop channels in ARI) you can 
> duplicate audio from a channel and send it off to where you want.

So originate a new channel, make one leg a UnicastRTP and the other a chanspy 
to spy on the channel you're interested in transcribing?

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)

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
  http://www.asterisk.org/community/astricon-user-conference

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Streaming for ASR

[Matt Riddell]

> On 17/10/2016, at 3:43 PM, Luca Pradovera <luca.pradov...@gmail.com> wrote:
> 
> I have been working on designs for two different projects, where both of them 
> would need to use the IBM Watson streaming ASR service.
> 
> Would it be possible to write out the audio frames as they get recorded? 
> Watson supports 16 bit signed little endian audio, IIRC, but there are a few 
> other raw formats available.

That's the main problem I was discussing at AstriDevCon.  There's no way 
currently to stream the frames to a service.

At the moment I'm saving to a file and then sending that file once recording 
has finished.

--
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)

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
  http://www.asterisk.org/community/astricon-user-conference

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Surfing the web via Asterisk.

[Matt Riddell]

I'm guessing you're going to be wanting something closer to this:

https://www.npmjs.com/package/speech-rule-engine 
<https://www.npmjs.com/package/speech-rule-engine>

--
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)

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
  http://www.asterisk.org/community/astricon-user-conference

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Surfing the web via Asterisk.

[Matt Riddell]

> On 17/10/2016, at 9:51 AM, Jonathan H <lardconce...@gmail.com> wrote:
> 
> Ah, no, you misunderstand. Asterisk wouldn't care one little bit what
> is on the page - Chromevox would do all that.
> A screenreader usually tabs or arrows their way about, selecting
> headings to read content.
> 
> Thus, Asterisk ONLY needs to be able to hear content FROM the browser
> and pipe it to the channel, and pass keypresses back TO the browser.
> 
> The human is the parser, if that makes sense?


Right, so you're using a prebuilt browser to do the parsing.

You'd really want to see if you can get ChromeVox as a library rather than as a 
browser though - otherwise you're going to be limiting yourself to using one 
concurrent channel and hacks like jack audio to move the audio from the browser 
to the channel.

--
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)

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
  http://www.asterisk.org/community/astricon-user-conference

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Surfing the web via Asterisk.

[Matt Riddell (lists)]

This is a really interesting project but I think it's going to be seriously 
hard. You're going to need to parse meaning from a site, and that's not an easy 
thing to do. 

If you're focused on a few of the bigger sites then it might be easier. 

You almost want a middle layer that can parse meaning from a site into xml or 
something. 

Then I'd work on creating objects out of each kind of tag. The problem is that 
navigation may not be the same as you'd see when visiting a site. You're not 
really going to be moving left and right. It would be more like tab works. Next 
item kinda thing. And items wouldn't necessarily be in the same order as you 
see. Pull right/left classes for bootstrap etc would make layout different. 

I would maybe check if there are any libraries that can parse HTML into objects 
first and if not then start building on yourself. The telephony side will be 
easy. You'd use agi or something to navigate the object you create and tts to 
describe current position. The hard part will be parsing the HTML even though 
most HTML is broken :-)

Kind regards,

Matt Riddell

> On Oct 17, 2016, at 9:00 AM, Jonathan H <lardconce...@gmail.com> wrote:
> 
> Has anyone attempted making the web phone accessible? I can only find one 
> company which operated between 1996 and 2000. 
> 
> I was thinking, install Chrome with Chromevox, headless, on a server, and use 
> something like an AGI to send basic keyboard commands to navigate a page, as 
> a screenreader user would, and pipe the audio back to a channel, to be 
> streamed by Asterisk. 
> 
> (Bear with me here - it's a project for blind people involving a telephone 
> and some lateral thinking!)
> 
> And yes, I mean more than just CURL a page, tts it and then read. I'm talking 
> about using the keypad to navigate the headers and landmarks. There are just 
> enough keys to make it viable.
> 
> Of particular interest is the very high quality of the Chromevox screenreader 
> voice from Google.
> 
> Does such a framework exist? I'm aware of a project called Chromium Headless, 
> but some of the links are broken and it doesn't seem to have the 
> audio/extension part
> 
> Failing headless, what about running it on a vps in an x-window environment? 
> Not sure how I'd pass the keyboard presses to it, without using a keyboard...
> 
> Any ideas, or is the whole idea complete madness? Thanks!
> -- 
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> 
> Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
>  http://www.asterisk.org/community/astricon-user-conference
> 
> New to Asterisk? Start here:
>  https://wiki.asterisk.org/wiki/display/AST/Getting+Started
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
  http://www.asterisk.org/community/astricon-user-conference

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Installing Asterisk on MAC native

[Matt Riddell (lists)]

It pretty much just works the same way as Linux. you might need to use brew to 
install a few prerequisites but I've got it running on my MacBook Pro without 
any major problems. 

It's good for testing things but I wouldn't use a MacBook as an office server 
or anything. 

And to be fair most of the time I just use virtualization and spin up Debian in 
parallels. 

Kind regards,

Matt

> On Sep 20, 2016, at 4:07 PM, Glenn Geller (VDOPh) <ggel...@vdo-ph.com> wrote:
> 
> If you're looking for installing on a MAC, best to start searching for MAC 
> OSX install
> 
> See here: 
> http://www.voip-info.org/wiki/view/Asterisk+Getting+Started+on+MacOSX
> 
> I don't know how old this is, or if it directly applies to your task at hand, 
> but it may be a start.
> 
> Also, if you're just looking for a simple PBX for light usage, there may be 
> other options out there for MAC OS as well.
> 
> Good hunting!
> 
> Glenn Geller
> 
>> On Tue, Sep 20, 2016 at 1:58 PM, Saint Michael <vene...@gmail.com> wrote:
>> ​I need to install Asterisk on a MAC, native, no virtualization.
>> Has anybody done this? Are there documents on the Internet?
>> I googled it and all web sites that claimed to help installing Asterisk on
>> a MAC have disappeared. Is it possible at all?
>> Digium should actually have a MAC app in the Apple store with a PBX. It 
>> should be a paid app. I would buy it right away.
>> 
>> ​
>> 
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> 
>> Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
>>   http://www.asterisk.org/community/astricon-user-conference
>> 
>> New to Asterisk? Start here:
>>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>> 
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> -- 
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> 
> Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
>  http://www.asterisk.org/community/astricon-user-conference
> 
> New to Asterisk? Start here:
>  https://wiki.asterisk.org/wiki/display/AST/Getting+Started
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
  http://www.asterisk.org/community/astricon-user-conference

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Asterisk 11 - Security Fix Mode

[Matt Fredrickson]

Hello Everyone,

As many of you are already aware, we are rapidly approaching the time
when the Asterisk 11 branch will go into what is known as security fix
only mode.  Up to this point, bug fixes have been included and merged
into the 11 branch.  For Asterisk 11, this new phase of life shall
begin October 25th of this year.

This means that from a development perspective, the Asterisk
development team will not be putting effort into bug fixes for the 11
branch after the 25th of October.  Security related patches will be
merged for another year before completely putting the branch to rest.
During the course of that year, releases will be made as needed and as
security related patches are merged.

For any questions, you can either reply to this message or look at the
Asterisk Versioning Policy wiki page [1] as it explains most of this
process in greater detail.

Thanks so much again for all of your support and effort.  Asterisk
would not be what it is if it were not for all the great contributors
that are and have been involved.

[1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions

-- 
Matthew Fredrickson
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
  http://www.asterisk.org/community/astricon-user-conference

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Farewell

[Matt Fredrickson]

Best of wishes to you in your retirement!  It's been a great 10 years, and
I'm personally looking forward to the great things coming in the next 10.

Matthew Fredrickson

On Wed, Aug 17, 2016 at 8:37 AM, Vincent Medina  wrote:

> I just wanted to wish all of you good luck I'm officially retired and will
> be removing my name from the list. I can attest that this list has been a
> great help throughout my career. I have deployed probably over 100
> installations over a 10-year period.
>
> Any of you newcomers this list the most valuable tool you can have.
>
>
>
> Sincerely,
>
> Vincent Medina
> Information Systems Director
> APCN, Inc.
>
> (305)785-3355
>
> Sent using www.apcn.net Internet Services.
>
>
>  Original message 
> From: Dario Estupinan 
> Date: 08/17/2016 8:53 AM (GMT-05:00)
> To: Asterisk Users Mailing List - Non-Commercial Discussion <
> asterisk-users@lists.digium.com>
> Subject: Re: [asterisk-users] Realtime SIP peers do not register any more
> after upgrade to Asterisk 13
>
> REMOVE ME please.
>
> 2016-08-15 15:16 GMT-05:00 Jonas Kellens :
>
>> Hello
>>
>> after I have upgraded from Asterisk 12 to asterisk-certified-13.8-cert1
>> none of my realtime SIP peers (saved in MySQL DB) register anymore.
>>
>>
>> [Aug 15 22:03:43] NOTICE[30098]: chan_sip.c:28451
>> handle_request_register: Registration from ''
>> failed for '78.119.140.190:5076' - Wrong password
>> [Aug 15 22:04:13] NOTICE[30098]: chan_sip.c:28451
>> handle_request_register: Registration from ''
>> failed for '78.119.140.190:5072' - Wrong password
>> [Aug 15 22:04:43] NOTICE[30098]: chan_sip.c:28451
>> handle_request_register: Registration from ''
>> failed for '78.119.140.190:5062' - Wrong password
>> [Aug 15 22:04:46] NOTICE[30098]: chan_sip.c:28451
>> handle_request_register: Registration from ''
>> failed for '78.119.140.190:5060' - Wrong password
>> [Aug 15 22:04:53] NOTICE[30098]: chan_sip.c:28451
>> handle_request_register: Registration from ''
>> failed for '78.119.140.190:5060' - Wrong password
>>
>>
>> Is this a known problem ??
>>
>>
>> Second question I have : can I get the complete list of columns that can
>> be used in realtime database for sip peers somewhere (update for Ast 13) ?
>> Are columns like dtlsenable, dtlsverify, dtlscertfile, dtlscafile,
>> dtlssetup possible ??
>>
>>
>>
>>
>> Thanks for the help.
>>
>>
>> Kind regards.
>>
>> Jonas.
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>   http://www.asterisk.org/hello
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
>
> --
>
> *DARIO ESTUPIÑAN V.*
> *Líder de NOC+*
> *Cel: 3008832295*
> *E-Mail: darioestupi...@soygenial.co *
>
>
>
>
> Antes de imprimir este mensaje, asegúrese de que es necesario. Proteger el
> medio ambiente está también en sus manos.
>
>
> AVISO LEGAL: Este mensaje es confidencial, puede contener información
> privilegiada y no puede ser usado ni divulgado por personas distintas de su
> destinatario. Si recibe este correo por error, por favor elimínelo y avise
> a su remitente. Está prohibida su retención, grabación, utilización,
> aprovechamiento o divulgación con cualquier propósito. La Corporación
> Politécnica Nacional de Colombia no asume ninguna responsabilidad por
> eventuales daños generados por el recibo y el uso de este material, siendo
> responsabilidad del destinatario verificar con sus propios medios la
> existencia de virus u otros defectos. El presente correo electrónico solo
> refleja la opinión de su Remitente y no representa necesariamente la
> opinión oficial de la Corporación o de sus Directivos.
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 
Matthew Fredrickson
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   

Re: [asterisk-users] Asterisk 11.23.0 on CentOS6 : how to get ICE support ?

[Matt Fredrickson]

On Thu, Aug 11, 2016 at 9:40 AM, Jonas Kellens <jonas.kell...@telenet.be> wrote:
> My main reason not to upgrade to Ast 13 is because I'm afraid of losing
> functionality as there are certain functions deprecated/replaced. This can
> also cause headache :-)
>
> I will do so if there is no other option.
>
> But still, I don't see why Ast 13 would differ so much in this case ? If ICE
> and NAT is working (not causing problems) why should Ast 13 bring me audio
> and Ast 12 don't ??

If you want to minimize grief, start with 13 - WebRTC has been a
moving target for the last 5 years, it is not an old, mature standard
like ISDN or SIP.  If you find interop problems in an older version of
Asterisk with WebRTC, it's likely that it has been fixed in 13, and if
it hasn't the most likely place to obtain the fix will be in 13.

After you get the WebRTC part working, then you can move back the
versions of Asterisk you're using to see if it still works.

As far as ICE not working goes, if the browser you're talking to is
not on the same network as the Asterisk server, it's *possible* you
might need a true TURN server as well, instead of just an ICE server.

Matthew Fredrickson

>
>
>
> On 11-08-16 16:25, Jonathan H wrote:
>
> I'm genuinely fascinated why you are insisting on using a version of
> Asterisk almost 3 years old, for which EOL support ended last year.
>
> Is there any particular reason you cannot or will not use the current
> version as others have suggested?
>
> Also, I see you are using Doubango and WebRTC, but in the logs, I see WS and
> WSS.
>
> You NEED to be using 100% WSS otherwise you've not got a hope in hell of
> anything working with WEBRTC.
> Check the console of the web browser you are trying to make the call from
> (CTRL-SHIFT-I in Chrome on Windows, for example).
>
> Also, you'll need to be using valid certificates - self-signed certificates
> won't work for any current implementation of WebRTC that I know of,
> certainly not if anything involves current versions of Chrome or Firefox.
> That said, LetsEncrypt certs work fine for this, so no need to spend out on
> one.
>
> Switch to Asterisk 13.10 and save yourself a whole lotta headache.
>
> On 11 August 2016 at 15:09, Jonas Kellens <jonas.kell...@telenet.be> wrote:
>>
>> Hello
>>
>> Using Asterisk 12.8.2.
>>
>
>
>>
>> On 10-08-16 22:03, Matt Fredrickson wrote:
>>>
>>> My suggestion is to verify and debug against Asterisk 13 first, and
>>> then you can try backing down versions, rather than reverse.  WebRTC
>>> is a rapidly moving target, and has required ongoing changes that may
>>> not have made it into older and feature frozen versions of Asterisk.
>
>
>
>
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users



-- 
Matthew Fredrickson
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk & Vitelity Invite issues

[Matt Fredrickson]

Wait a second, I thought in your original email that you said that
Asterisk was generating reinvites.  It sounds now like you're saying
that the remote side is initiating reinvites instead.

My understanding is that the canreinvite/directmedia option only
influences Asterisk's behavior with regards to generating reinivites.
If it receives a reinvite, I don't think these options will do
anything about that.  In fact, I'd guess that not properly responding
to a received reinvite is going to potentially break things from the
SIP perspective.

Matthew Fredrickson


On Wed, Aug 10, 2016 at 4:53 PM, Tammy Firefly <tammy-li...@wiztech.biz> wrote:
>
>
> On 8/9/16 12:40 PM, Matt Fredrickson wrote:
>> On Mon, Aug 8, 2016 at 9:25 AM, Tammy Firefly <tammy-li...@wiztech.biz> 
>> wrote:
>>> Hi All,
>>>
>>> We have asterisk 11.23 running sip to vitelity and from there IAX trunks
>>> split off to where they need to go.  We are having a problem getting
>>> chan_sip to quit ignoring re-invites from Vitelity.  Our side ends up
>>> sending a reinvite which their side & they do not support us sending a
>>> reinvite.  Ive tried:
>>>
>>> canreinvite=no which was supposedly replaced by:
>>>
>>> directmedia=no
>>>
>>> Can anyone shed any light on this matter?  I'd love to get this fixed.
>>>
>>
>> Those options *should* influence chan_sip's reinvite behavior - at
>> least they have from my experiences working with chan_sip.  Do you
>> know what is triggering the reinvite in the first place, or does it
>> look like a normal media reinvite?
>>
>
>
> every 15 minutes vitelity sends a re-invite to keep the call going.  I
> have a packet capture from it if you'd like it feel free to email me off
> list @ tamara.wis...@wiztech.biz
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users



-- 
Matthew Fredrickson
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Original Callerid on transfer in asterisk 13

[Matt Fredrickson]

How are you attempting to view the original CallerId?

Matthew Fredrickson

On Wed, Aug 10, 2016 at 2:59 PM, Israel Gottlieb  wrote:
> Hi
> Is there any configuration change in asterisk 13.9.1 to show original
> callerid on a transfer
> In asterisk 11.21 it works as expected
>
> Thanks
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users



-- 
Matthew Fredrickson
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk 11.23.0 on CentOS6 : how to get ICE support ?

[Matt Fredrickson]

My suggestion is to verify and debug against Asterisk 13 first, and
then you can try backing down versions, rather than reverse.  WebRTC
is a rapidly moving target, and has required ongoing changes that may
not have made it into older and feature frozen versions of Asterisk.

Matthew Fredrickson

On Wed, Aug 10, 2016 at 3:01 PM, Jonas Kellens <jonas.kell...@telenet.be> wrote:
> Hello
>
> thank you for your answer.
>
> I don't understand how there are many tutorials and examples on the web
> where every time the outcome is a working setup. Very strange I feel now
> after my personal experience with Asterisk 11 and webRTC.
>
> You also say Asterisk 13. How about Asterisk 12 then ??
>
>
>
> Kind regards.
>
>
>
> On 10-08-16 21:53, Matt Fredrickson wrote:
>
> I don't see an ice-ufrag or ice-pwd line in the response from
> Asterisk, correlating with your suspicion that there is no ICE.  Are
> you sure that the stun server you're using (the google one) still
> works?  I haven't tried that server in a while, but I distantly seem
> to recall that maybe they shut it down.
>
> Asterisk 13 is a better place to be as well.  Asterisk 11 hasn't been
> feature updated in a while, and it could be that it could be a number
> of patches/fixes behind with regards to webrtc support, particularly
> with regards to interoperating with a modern browser version.
>
> Hope that helps,
> Matthew Fredrickson
>
> On Wed, Aug 10, 2016 at 5:02 AM, Jonas Kellens <jonas.kell...@telenet.be>
> wrote:
>
> On 10-08-16 08:52, Ludovic Gasc wrote:
>
> For WebRTC, I recommend you to use Asterisk 13+.
>
> Have a nice day.
>
> Ludovic Gasc (GMLudo)
> http://www.gmludo.eu/
>
>
>
>
> Hello
>
> then why is there an option in sip.conf and rtp.conf " icesupport=yes" ??
>
> This is no answer to my question.
>
> So again : what am I missing to get ICE support on my Asterisk 11.23.0 ??
>
>
>
> Kind regards.
>
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users



-- 
Matthew Fredrickson
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk 11.23.0 on CentOS6 : how to get ICE support ?

[Matt Fredrickson]

I don't see an ice-ufrag or ice-pwd line in the response from
Asterisk, correlating with your suspicion that there is no ICE.  Are
you sure that the stun server you're using (the google one) still
works?  I haven't tried that server in a while, but I distantly seem
to recall that maybe they shut it down.

Asterisk 13 is a better place to be as well.  Asterisk 11 hasn't been
feature updated in a while, and it could be that it could be a number
of patches/fixes behind with regards to webrtc support, particularly
with regards to interoperating with a modern browser version.

Hope that helps,
Matthew Fredrickson

On Wed, Aug 10, 2016 at 5:02 AM, Jonas Kellens  wrote:
>
> On 10-08-16 08:52, Ludovic Gasc wrote:
>
> For WebRTC, I recommend you to use Asterisk 13+.
>
> Have a nice day.
>
> Ludovic Gasc (GMLudo)
> http://www.gmludo.eu/
>
>
>
>
> Hello
>
> then why is there an option in sip.conf and rtp.conf " icesupport=yes" ??
>
> This is no answer to my question.
>
> So again : what am I missing to get ICE support on my Asterisk 11.23.0 ??
>
>
>
> Kind regards.
>
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users



-- 
Matthew Fredrickson
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] EAGI script with missing audio on /dev/fd/3

[Matt Fredrickson]

On Tue, Aug 2, 2016 at 11:42 AM, nik600  wrote:
> Dear all
>
> i'm trying to access to the input audio raw stream with a very basic EAGI
> script:
>
>
> #!/bin/sh
> echo "EXEC Queue 2001"
> cat  /dev/fd/3 > /tmp/pippo
>
> This is my dialplan:
>
> exten => 001,NoOp(test)
> exten => 001,n,Answer
> exten => 001,n,EAGI(/tmp/my-eagi.agi)
>
>
> When i call, the script is executed and the call goes in queue, i can hear
> the MOH, the file /tmp/pippo is created but it is empty.
>
> Any idea or suggestion?

If you take out the "echo "EXEC Queue 2001" part of it, do you get
audio in the file?

-- 
Matthew Fredrickson
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk & Vitelity Invite issues

[Matt Fredrickson]

On Mon, Aug 8, 2016 at 9:25 AM, Tammy Firefly  wrote:
> Hi All,
>
> We have asterisk 11.23 running sip to vitelity and from there IAX trunks
> split off to where they need to go.  We are having a problem getting
> chan_sip to quit ignoring re-invites from Vitelity.  Our side ends up
> sending a reinvite which their side & they do not support us sending a
> reinvite.  Ive tried:
>
> canreinvite=no which was supposedly replaced by:
>
> directmedia=no
>
> Can anyone shed any light on this matter?  I'd love to get this fixed.
>

Those options *should* influence chan_sip's reinvite behavior - at
least they have from my experiences working with chan_sip.  Do you
know what is triggering the reinvite in the first place, or does it
look like a normal media reinvite?

-- 
Matthew Fredrickson
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] AstriCon 2016 - XMPP and Asterisk

[Matt Fredrickson]

Looking forward to seeing you there, and hopefully to seeing your talk!

Matthew Fredrickson

On Tue, Aug 2, 2016 at 11:02 AM, Marcelo Terres  wrote:
> Going to AstriCon 2016 ?
>
> Don't miss my talk about how to use XMPP and Asterisk to improve the
> user experience.
>
> https://astricon2016.sched.org/event/7Zje/using-asterisk-and-xmpp-to-provide-greater-tools-to-your-customers-and-your-users
>
> Regards,
>
> Marcelo H. Terres 
> IM: mhter...@jabber.mundoopensource.com.br
> https://www.mundoopensource.com.br
> https://twitter.com/mhterres
> https://linkedin.com/in/marceloterres
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users



-- 
Matthew Fredrickson
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Force out-bond call to specific CIC

[Matt Fredrickson]

Yes, as far as I remember, in your dial string, simply use a
Dial(DAHDI/X/1234567) where X is the dahdi device channel number.

Hope that helps.
Matthew Fredrickson

On Wed, Jul 13, 2016 at 5:22 AM, Mehdi Shirazi  wrote:
> Hi
>
> How is it possible to use Dial application to force out-bond call use
> "specified channel" number in one E1 or specified CIC (SS7) ?
>
> Regards
> M.Shirazi
>
>
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users



-- 
Matthew Fredrickson
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Unable to create channel DAHDI

[Matt Fredrickson]

Looks like the hookstate is listed as offhook.  I don't think
chan_dahdi will attempt to make a call out a device that is offhook.

Hope that helps,
Matthew Fredrickson

On Tue, Jun 7, 2016 at 1:36 PM, Brent Davidson
 wrote:
> In trying to troubleshoot the Delay after Answer problem I had before (which
> seems to be fixed), I have somehow created a new problem:
>
> Outgoing calls are now failing with the following message:
>
> [Jun  7 13:28:09] WARNING[9247][C-]: app_dial.c:2429 dial_exec_full:
> Unable to create channel of type 'DAHDI' (cause 0 - Unknown)
>
> But I DO have working dahdi as incoming calls are working correctly.
>
> CLI> dahdi show channels
>Chan Extension   Context Language   MOH Interpret
> BlockedIn Service Description
>  pseudo defaultdefault
> Yes
>   3 mainmenu   default
> Yes
>   4 mainmenu   default
> Yes
> CLI> dahdi show status
> Description  Alarms  IRQbpviol CRCFra
> Codi Options  LBO
> Wildcard AEX410  OK  0  0  0  CAS
> Unk   0 db (CSU)/0-133 feet (DSX-1)
> CLI> dahdi show channel 3
> Channel: 3
> Description:
> File Descriptor: 14
> Span: 1
> Extension:
> Dialing: no
> Context: mainmenu
> Caller ID:
> Calling TON: 0
> Caller ID name:
> Mailbox: none
> Destroy: 0
> InAlarm: 0
> Signalling Type: FXS Kewlstart
> Radio: 0
> Owner: 
> Real: 
> Callwait: 
> Threeway: 
> Confno: -1
> Propagated Conference: -1
> Real in conference: 0
> DSP: no
> Busy Detection: yes
> Busy Count: 8
> Busy Pattern: 0,0,0,0
> TDD: no
> Relax DTMF: yes
> Dialing/CallwaitCAS: 0/0
> Default law: ulaw
> Fax Handled: no
> Pulse phone: no
> HW Gains (RX/TX): Disabled/Disabled
> SW Gains (RX/TX): 0.00/0.00
> Dynamic Range Compression (RX/TX): 0.00/0.00
> DND: no
> Echo Cancellation:
> 128 taps
> (unless TDM bridged) currently OFF
> Wait for dialtone: 0ms
> Actual Confinfo: Num/0, Mode/0x
> Actual Confmute: No
> Hookstate (FXS only): Offhook
> CLI>
>
> dahdi show channel 4
> Channel: 4
> Description:
> File Descriptor: 15
> Span: 1
> Extension:
> Dialing: no
> Context: mainmenu
> Caller ID:
> Calling TON: 0
> Caller ID name:
> Mailbox: none
> Destroy: 0
> InAlarm: 0
> Signalling Type: FXS Kewlstart
> Radio: 0
> Owner: 
> Real: 
> Callwait: 
> Threeway: 
> Confno: -1
> Propagated Conference: -1
> Real in conference: 0
> DSP: no
> Busy Detection: yes
> Busy Count: 8
> Busy Pattern: 0,0,0,0
> TDD: no
> Relax DTMF: yes
> Dialing/CallwaitCAS: 0/0
> Default law: ulaw
> Fax Handled: no
> Pulse phone: no
> HW Gains (RX/TX): Disabled/Disabled
> SW Gains (RX/TX): 0.00/0.00
> Dynamic Range Compression (RX/TX): 0.00/0.00
> DND: no
> Echo Cancellation:
> 128 taps
> (unless TDM bridged) currently OFF
> Wait for dialtone: 0ms
> Actual Confinfo: Num/0, Mode/0x
> Actual Confmute: No
> Hookstate (FXS only): Offhook
>
> The Hookstates always say offhook for some reason, though I'm not sure why.
>
> My setup:
>
> Server running Asterisk 13.9.1, Dahdi 2.11.1 w/ OSLEC
> Server is CentOS 7
> Quad core CPU with 16GB Ram
> 2 Snom 300 phones.
> NO NAT.  Server and phone are on the same subnet with only a gigabit switch
> between them.
> Digium AEX410P analog card with 2 incoming analog PSTN lines
>
> Any ideas?
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users



-- 
Matthew Fredrickson
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Advices on how to evaluate voice quality in a mixed Dahdi/SIP environment ?

[Matt Fredrickson]

On Wed, May 18, 2016 at 9:44 AM, Olivier  wrote:
> I've got the following setup:
>
> PSTN  ITSP  SDSL Modem-Router  Gateway -
> Asterisk with B410P --- SIP Phones

Wow.

> Both SDSL Modem-Router and Gateway are managed by my ITSP.
>
> Some calls coming from PSTN and forwarded to an other PSTN number have a
> poor voice quality.

How are you forwarding them?  Is it in such a way that you remain in
the audio path, or do you get out of the audio path in the forward?

> How can I best illustrate this ?

It depends on what let has the bad audio.  If it's on the SIP side
(RTP to RTP) a pcap file will show you your perspective of audio
losses.  Received RTCP reports should show you the other side's
perspective of audio losses as well.

> A friend advised me to simply record incoming DAHDI channel, for instance.
> How can I then translate record WAV file into meaningful figures ?

If DAHDI is still in the picture in the forward scenario, that would
be another place to monitor the audio.

> More generaly, what would you suggest ?

Try to capture each leg (IP side, using tcpdump/wireshark) and on
DAHDI using dahdi_monitor or something equivalent.  Figure out if any
of your legs of audio quality issues.  If you don't see anything, it's
something at their end.

-- 
Matthew Fredrickson
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Avaya Phones and Asterisk

[Matt Fredrickson]

On Fri, May 20, 2016 at 8:54 AM, Diogo Cosito  wrote:
> Dear gentlemen, how are you?
> I wonder if anyone has experience with Avaya devices, 9608G and 9641GS
> models, running on SIP and using TCP transport.
> The calls work well, but the callerid only "pass" number of the extension or
> external number, without the name (configured correctly in sip.conf and
> testing with other devices UDP works fine, like Yealink, Eyebeam, etc),
> device contacts list does not work and also the hint does not work 
> can anybody help me?

So if you use UDP transport everythng works fine?

Can you post a packet capture of this happening?  Also, what version
of Asterisk and your sip.conf?

-- 
Matthew Fredrickson
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] pjsip segfault problem

[Matt Fredrickson]

Have you tried updating to pjproject version 2.5.x?  It should have
the patch that you listed in your other email, which I believe should
be included in that branch.

Hope that helps, and best of luck.

Matthew Fredrickson

On Thu, May 26, 2016 at 4:11 AM, Marek Červenka  wrote:
> hi,
>
> after switch from 13.7 + pjproject 2.4.5 to 13.9.1 pjproject bundled i have
> problem with segfault  (centos 6)
>
> Program terminated with signal 11, Segmentation fault.
> #0  0xb7665695 in check_cached_response (sess=0xafbd688c, packet=0xb07676d8,
> pkt_size=132, options=1, token=0xafecc2bc, parsed_len=0x0,
> src_addr=0xb0e47a20, src_addr_len=16)
> at ../src/pjnath/stun_session.c:1287
> 1287if (t->msg_magic == msg->hdr.magic &&
>
>
> it was only once after 2 days.  i dont know how to repeat it now :(
>
> any similiar experience?
>
>
> --
> ---
> Marek Cervenka
> ===
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users



-- 
Matthew Fredrickson
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] T.38 with Audiocodes gateway

[Matt Fredrickson]

On Fri, Apr 29, 2016 at 1:34 AM, Olivier  wrote:
> Hello,
>
> I'm helping a colleague (*) which has the following setup:
>
> ITSP ---  --- Asterisk 13 ---  --
> Audiocodes MP-112 ---   --- Fax machine
>
> My issue is the following :
> Audiocodes gateway reject INVITEs with 488 Not Acceptable Here
>
> It seems this gateway requires t38 settings to be present in SDP body in the
> very first INVITE.
>
> My questions are the following:
>
> 1. I expected T.38 to exclusively work with reINVITE where calls are
> established as normal voice calls (PCMA/PCMU in SDP, for instance) and then
> upgraded to T.38 (when CNG is detected, for instance).
> Have you ever heard of T.38 sessions being established right from the start
> (ie with T.38 settings in the first INVITE) ?

No.  It would seem to be extremely broken if it denies a call based on
a lack of T.38 sdp parameters on the initial INVITE.

> 2. Is it possible to configure Asterisk to pass T.38 settings in SDP in the
> first INVITE it sends ?
>
> 3. Any suggestion with Audiocodes gateway ?

Look for T.38 settings maybe?  See if there is something keeping you
from sending an initial invite with non-T.38 SDP?

-- 
Matthew Fredrickson
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] PRI error "ROSE REJECT"

[Matt Fredrickson]

Perhaps it's taking a bit longer in the network for the media path to
open after the CONNECT, which would explain why the first digit is not
being detected.

Also, what changed last week?

I don't see the ROSE REJECT message anywhere in the pri debug -
perhaps you didn't catch it.

Matthew Fredrickson

On Fri, Mar 25, 2016 at 9:15 PM, Carlos Chavez <cur...@telecomabmex.com> wrote:
> On 2016-03-25 16:02, Matt Fredrickson wrote:
>>
>> PRI debug of the entire call would be great, also, switchtype would be
>> awesome as well.
>>
>> Thanks!
>>
>> Matthew Fredrickson
>>
>> On Thu, Mar 24, 2016 at 4:07 PM, Carlos Rojas <crt.ro...@gmail.com> wrote:
>>>
>>> Hi
>>>
>>> Did you activate the pri debug on the cli asterisk?
>>>
>>> On Thu, Mar 24, 2016 at 12:59 PM, Carlos Chavez <cur...@telecomabmex.com>
>>> wrote:
>>>>
>>>>
>>>> We've been having some problems with an E1 PRI line for a few days.  We
>>>> get the following errors:
>>>>
>>>> [Mar 24 10:13:39] ERROR[22009] chan_dahdi.c: PRI Span: 2 ROSE REJECT:
>>>> [Mar 24 10:13:39] ERROR[22009] chan_dahdi.c: PRI Span: 2INVOKE
>>>> ID:
>>>> 316
>>>> [Mar 24 10:13:39] ERROR[22009] chan_dahdi.c: PRI Span: 2PROBLEM:
>>>> Invoke: Unrecognized Operation
>>>>
>>>> The telephone company says that everything is fine on their side,
>>>> obviously.  The problems started a few days ago when a user reported
>>>> that
>>>> incoming calls get dropped when you try to dial a particular extension
>>>> from
>>>> the main IVR.  We are using Asterisk 1.8.15-cert2 on a CentOS 6.7
>>>> server,
>>>> DAHDI 2.6.1 and libpri 1.4.  Any recommendations?
>>>>
>>>> --
>
>
> system.conf:
> # Span 1: TE2/0/1 "T2XXP (PCI) Card 0 Span 1" (MASTER)
> span=1,2,0,cas,hdb3
> cas=1-15:1101
> cas=17-31:1101
>
> # Span 2: TE2/0/2 "T2XXP (PCI) Card 0 Span 2"
> span=2,1,0,ccs,hdb3
> # termtype: te
> bchan=32-46
> dchan=47
> bchan=48-62
>
> loadzone= mx
> defaultzone = mx
>
> chan_dahdi.conf:
> language=es
> context=e1-incoming
> usecallerid=yes
> hidecallerid=no
> callwaiting=no
> canpark=no
> usecallingpres=no
> callwaitingcallerid=no
> threewaycalling=no
> transfer=yes
> cancallforward=no
> callreturn=no
> echocancel=yes
> echocancelwhenbridged=no
> echotraining=yes
> rxgain=0.0
> txgain=0.0
> accountcode=E1
> amaflags=default
> signalling=pri_cpe
> pridialplan=unknown
> prilocaldialplan=unknown
> switchtype=euroisdn
> overlapdial=no
> immediate=no
> group=2
> faxdetect=no
> callerid=asreceived
> mohinterpret=default
> mohsuggest=default
> dahdichan=32-46,48-62
>
> Here is the pri debug:
> -- Accepting call from '55' to '5732' on channel 0/1, span 2
> -- Executing [5732@e1-incoming:1] Goto("DAHDI/i2/55-3c",
> "menu-gci,s,1") in new stack
> -- Goto (menu-gci,s,1)
> -- Executing [s@menu-gci:1] Wait("DAHDI/i2/55-3c", "2") in new
> stack
> -- Executing [s@menu-gci:2] Answer("DAHDI/i2/55-3c", "") in new
> stack
> PRI Span: 2 q931.c:4683 q931_connect: Call 48 enters state 8 (Connect
> Request).  Hold state: Idle
> PRI Span: 2
> PRI Span: 2 > DL-DATA request
> PRI Span: 2 > Protocol Discriminator: Q.931 (8)  len=14
> PRI Span: 2 > TEI=0 Call Ref: len= 2 (reference 48/0x30) (Sent to
> originator)
> PRI Span: 2 > Message Type: CONNECT (7)
> PRI Span: 2 TEI=0 Transmitting N(S)=98, window is open V(A)=98 K=7
> PRI Span: 2
> PRI Span: 2 > Protocol Discriminator: Q.931 (8)  len=14
> PRI Span: 2 > TEI=0 Call Ref: len= 2 (reference 48/0x30) (Sent to
> originator)
> PRI Span: 2 > Message Type: CONNECT (7)
> PRI Span: 2 > [18 03 a9 83 81]
> PRI Span: 2 > Channel ID (len= 5) [ Ext: 1  IntID: Implicit  Other(PRI)
> Spare: 0  Exclusive  Dchan: 0
> PRI Span: 2 >   ChanSel: As indicated in following
> octets
> PRI Span: 2 >   Ext: 1  Coding: 0  Number Specified
> Channel Type: 3
> PRI Span: 2 >   Ext: 1  Channel: 1 Type: CPE]
> PRI Span: 2 > [1e 02 81 82]
> PRI Span: 2 > Progress Indicator (len= 4) [ Ext: 1  Coding: CCITT (ITU)
> standard (0)  0: 0  Location: Private network serving the local user (1)
> PRI Span: 2 >   Ext: 1  Progress Description:
> Called equipment is non-ISDN. (2) ]
> 

Re: [asterisk-users] PRI error "ROSE REJECT"

[Matt Fredrickson]

PRI debug of the entire call would be great, also, switchtype would be
awesome as well.

Thanks!

Matthew Fredrickson

On Thu, Mar 24, 2016 at 4:07 PM, Carlos Rojas  wrote:
> Hi
>
> Did you activate the pri debug on the cli asterisk?
>
> On Thu, Mar 24, 2016 at 12:59 PM, Carlos Chavez 
> wrote:
>>
>> We've been having some problems with an E1 PRI line for a few days.  We
>> get the following errors:
>>
>> [Mar 24 10:13:39] ERROR[22009] chan_dahdi.c: PRI Span: 2 ROSE REJECT:
>> [Mar 24 10:13:39] ERROR[22009] chan_dahdi.c: PRI Span: 2INVOKE ID:
>> 316
>> [Mar 24 10:13:39] ERROR[22009] chan_dahdi.c: PRI Span: 2PROBLEM:
>> Invoke: Unrecognized Operation
>>
>> The telephone company says that everything is fine on their side,
>> obviously.  The problems started a few days ago when a user reported that
>> incoming calls get dropped when you try to dial a particular extension from
>> the main IVR.  We are using Asterisk 1.8.15-cert2 on a CentOS 6.7 server,
>> DAHDI 2.6.1 and libpri 1.4.  Any recommendations?
>>
>> --
>> Telecomunicaciones Abiertas de México S.A. de C.V.
>> Carlos Chávez
>> dCAP #1349
>> +52 (55)9116-91161
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>   http://www.asterisk.org/hello
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users



-- 
Matthew Fredrickson
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] OPUS support in Asterisk 13

[Matt Fredrickson]

On Thu, Mar 24, 2016 at 9:09 AM, Chirag Desai  wrote:
> Hi all,
>
> Sorry if this has been asked before. I searched a lot, but found conflicting
> answers, so hoping for some clarification.
>
> My question is does Asterisk 13 support OPUS? If so which version exactly?

Sort of - it supports OPUS pass through officially, but does not
support OPUS transcoding.  I'm not certain when pass through support
went in though, but I believe it was prior to cutting of the 13
branch.

There are some unofficial patches that add OPUS support to Asterisk
but I cannot point you to which ones are best to use, unfortunately.

Hope that helps!

-- 
Matthew Fredrickson
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] 1000 analogue lines with asterisk

[Matt Riddell (lists)]

There is definitely no way you should put 1000 lines on a single box. To be 
honest I do wonder what you want to do with 1000 lines as your description 
probably changes the recommendations. 

Kind regards,

Matt

> On Feb 17, 2016, at 5:09 PM, Goke Aruna <gok...@gmail.com> wrote:
> 
> Thanks Harry.
> I will check and revert. I hope it work perfectly with asterisk.
> Regards
> 
>> On Wed, Feb 17, 2016 at 8:32 AM, Harry McGregor <hmcgre...@biggeeks.org> 
>> wrote:
>> Hi,
>> 
>> For analog, I really like telco grade channel banks.
>> 
>> I would recommend the adit 600, there is a good market on Ebay, and you can 
>> do 48 channels per adit 600, with 2 T1 interfaces.  Having onsite spares 
>> would not be an issue (cost is low).  You can put two next to each other in 
>> a rack, taking up about 2U of space per 2 channel banks.
>> 
>> You could service this with six eight port T1 cards, or with eleven/twelve 
>> quad T1 cards.  I would distribute across two, three, or even four servers 
>> for redundancy/resiliency and load balancing.
>> 
>> -Harry
>> 
>> 
>>> On 02/17/2016 12:16 AM, Goke Aruna wrote:
>>> 
>>>> On Wed, Feb 17, 2016 at 8:14 AM, Mitul Limbani <mi...@enterux.in> wrote:
>>>> Sangoma 50 port FXS
>>> 
>>> 
>>> Thanks.
>>> Will I now stack 20 boxes in order to achieve the 1000 FXS lines?
>>> Regards
>>> 
>>> 
>> 
>> 
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>http://www.asterisk.org/hello
>> 
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> -- 
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>   http://www.asterisk.org/hello
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] accept DMTF tone during ringing

[Matt Fredrickson]

For what channel driver, and what use case?

It's my understanding that in the traditional telephone network
(ISDN/SS7/analog), prior to a call being answered, you were not necessarily
guaranteed a two way media path.  Sometimes it was available (there are few
stories of large companies who somehow talked their telco provider into
allowing it so users could traverse their IVR prior to a billable answer)
but I don't *think* it's guaranteed.

>From a SIP standpoint, there are many ways it could not work as well,
depending on the type of DTMF being sent.  If you're using inband DTMF of
any sort (audio inband or RFC2833-ish) if you don't have a bidirectional
media path established through NATs and such, you could miss digits as well.

Matthew Fredrickson


On Sun, Nov 8, 2015 at 3:50 PM, hadi  wrote:

>
> Hi,
>
>
> How to accept DMTF tone during ringing mode? Its possible.
>
> Regards
>
> -Hadi.Salem
>
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 
Matthew Fredrickson
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Looking for PRI Card with automatic fail over

[Matt Riddell (lists)]

There was a product called something like red box or similar that I saw around 
5 years ago. Probably not entirely helpful but maybe Google will help. 

Kind regards,

Matt

 On Aug 3, 2015, at 9:50 AM, Eric Klein eric.kl...@greenfieldtech.net wrote:
 
 Hi all,
 
 Strange request, I have a customer where we are putting an Asterisk PBX in 
 front of a legacy (non-VoIP) PBX. One of the requirements it that the 
 Asterisk PBX have 2 PRI ports (on towards the legacy PBX and one towards the 
 carrier) with the ability to go to pass through should the Asterisk PBX 
 (software or hardware level) fail.
 
 I did not see this feature in the Digium, Sangoma, Allo, or OpenVox cards.
 
 Does anyone know of a card that will do this? I know that Digium has an 
 external box (the r850) that does something similar for 2 PBXs making them 
 high availability, but in this case I only have the 1 Asterisk box acting as 
 a gateway and passing some calls out over SIP and IAX2. 
 
 Any suggestions would be appreciated.
 
 Thanks
 Eric
 -- 
 _
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --
 New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello
 
 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Looking for PRI Card with automatic fail over

[Matt Riddell]

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)

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk 11 and pulseaudio setup as local user

[Matt Riddell]

On 3Jul, 2015, at 06:17, Jerry Geis ge...@pagestation.com wrote:
 
 alsa_card_init^[[0m: snd_pcm_open failed: Connection refused
 soundcard_init^[[0m: Problem opening alsa capture device
 

If the problem is a permission thing I’d confirm that.

Run Asterisk as root and see if the problem goes away.

If so then you’ll have to work your way through devices and drivers and check 
who owns them and is able access them.

I’d also try temporarily disabling SELinux to see if that helps.

As with the first option this would be temporary - go back to running Asterisk 
as myuser or whatever and reenable SELinux regardless of the outcome.

--
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)

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Action Originate in Asterisk 13 creates 2 calls in core show channels

[Matt Riddell]

 On 3Jul, 2015, at 13:46, Alonso Genis abge...@gmail.com wrote:
 
 Hello,
 
 I am migrating a PABX system based in Asterisk 1.4 to Asterisk 13, with 
 success.
 
 I have an application that sends an action Originate to AMI for
 calling, it's working well, but when i see to Asterisk's CLI, i see 2
 calls for just one originate:


It looks like you’re originating with local channels.

Try with /n at the end (not sure if that helps anymore) and also send us your 
originate line.

--
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)


-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk 11 and pulseaudio setup as local user

[Matt Riddell]

 I am trying to get Asterisk 11 to co-exist with a centos 7 box that has pulse 
 audio running as a local user.
 
 Has anyone done that?
 What is the trick?
 
 I changed directories /var/run/asterisk, /var/spool/asterisk, 
 /var/log/asterisk,
 /usr/lib/asterisk, /etc/dahdi and all that stuff.
 
 The console channel gets errors on opening.

What errors are you getting?

--
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)

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Branch based on call volume

[Matt Riddell]

 On 27Jun, 2015, at 15:34, Michelle Dupuis mdup...@ocg.ca wrote:
 
 Is there a simple way to get call volume from a particular trunk within the 
 dialplan (for conditional branching)? 


Do you mean large number of calls or how loud the call is?

--
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)-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Branch based on call volume

[Matt Riddell]

 On 28Jun, 2015, at 10:19, Michelle Dupuis mdup...@ocg.ca wrote:
 
 ​I meant how many calls are in progress on a particular trunk.  (Sorry - I 
 didn't even think of the other interpretation).


Then yeah group and group count functions:

https://wiki.asterisk.org/wiki/display/AST/Function_GROUP 
https://wiki.asterisk.org/wiki/display/AST/Function_GROUP

https://wiki.asterisk.org/wiki/display/AST/Function_GROUP_COUNT 
https://wiki.asterisk.org/wiki/display/AST/Function_GROUP_COUNT

Just in case you’re wondering, the function to change the loudness of a channel 
is volume :-)

https://wiki.asterisk.org/wiki/display/AST/Function_VOLUME 
https://wiki.asterisk.org/wiki/display/AST/Function_VOLUME

--
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)-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Product CDR/Queue/Meetme

[Matt Riddell]

I will post this to the Asterisk news for you.

--
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)-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users