From infos at madovsky.org  Tue Jan  1 11:36:22 2019
From: infos at madovsky.org (Madovsky)
Date: Tue, 1 Jan 2019 03:36:22 -0800
Subject: [Freeswitch-users] about mod_av and ffmpeg/libav
Message-ID: <b82ff9a7-2e2b-934b-5279-e7f8af09294c@madovsky.org>

Hi,

what is today the compatibility status fo FS last git and libav version? 
Also is it compatible with ffmpeg too?

thanks



From babak.freeswitch at gmail.com  Wed Jan  2 12:52:59 2019
From: babak.freeswitch at gmail.com (Babak Yakhchali)
Date: Wed, 2 Jan 2019 16:22:59 +0330
Subject: [Freeswitch-users] mod_http_cache and mod_httapi Corrupted wav file
	after download
Message-ID: <CAEVfcab3rLkrT=z0E_895tAshHNxwGU8ORLgfF9q5MMq0Xm1eA@mail.gmail.com>

Hi
I'm using freeswitch 1.8.4 64 bit build using visual studio 2017 on win 10.
When using playback with the file on disk the file plays ok. but if using
http based sound urls which use http_cache or httapi the file playback is
distorted.
I attached the file and the downloaded cached file. The sizes are
different!!
everything works fine on debian
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From social at bohboh.info  Wed Jan  2 15:00:46 2019
From: social at bohboh.info (Social Boh)
Date: Wed, 2 Jan 2019 10:00:46 -0500
Subject: [Freeswitch-users] FreeSWITCH sources
Message-ID: <a2f1bd87-f094-1e50-3975-b6b5cf9c3575@bohboh.info>

Hello,

right now I can not download the sources of FreeSWITCH

git clone -b v1.6 https://freeswitch.org/stash/scm/fs/freeswitch.git

fatal: unable to access 
'https://freeswitch.org/stash/scm/fs/freeswitch.git/': Failed to connect 
to 2803:d000:fffe::151: Network is unreachable

Thank you

Regards

-- 
---
I'm SoCIaL, MayBe



From david.villasmil.work at gmail.com  Wed Jan  2 16:04:49 2019
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Wed, 2 Jan 2019 17:04:49 +0100
Subject: [Freeswitch-users] Module's architecture image
In-Reply-To: <DM6PR12MB2843FEB40A71F0D3117D055E97B20@DM6PR12MB2843.namprd12.prod.outlook.com>
References: <CAFGRPVrANh0duNvfrhkDeRfT2NTygOwpBo1Nfm+6Sgnix43K8g@mail.gmail.com>
 <DM6PR12MB2843FEB40A71F0D3117D055E97B20@DM6PR12MB2843.namprd12.prod.outlook.com>
Message-ID: <CAFGRPVoz4vtURgpNafLkgAanvFpaJ9poR0icjdUA-kx8NP9igA@mail.gmail.com>

Found it, it’s this one.

On Mon, 31 Dec 2018 at 16:33, bjordan at e-teleco.com <bjordan at e-teleco.com>
wrote:

> I do not know the specific image you are talking about but could it be
> this one?
>
>
>
> [image:
> https://www.packtpub.com/sites/default/files/Article-Images/1004_01_01.png]
>
>
>
> *From:* FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> *On
> Behalf Of *David Villasmil
> *Sent:* Friday, December 28, 2018 1:12 PM
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Subject:* [Freeswitch-users] Module's architecture image
>
>
>
> Hello all,
>
>
>
> Years ago, Anthony sent out an image explaining FS' module architecture. I
> had it somewhere but can't find it now...
>
>
>
> Anyone got it?
>
>
> Regards,
>
>
>
> David Villasmil
>
> email: david.villasmil.work at gmail.com
>
> phone: +34669448337
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From s.safarov at gmail.com  Wed Jan  2 17:39:47 2019
From: s.safarov at gmail.com (Sergey Safarov)
Date: Wed, 2 Jan 2019 20:39:47 +0300
Subject: [Freeswitch-users] about mod_av and ffmpeg/libav
In-Reply-To: <b82ff9a7-2e2b-934b-5279-e7f8af09294c@madovsky.org>
References: <b82ff9a7-2e2b-934b-5279-e7f8af09294c@madovsky.org>
Message-ID: <CAHtxdDdV3DHiVWwpDJXahxmAPHBPqTvPJKLQN6-4behHexxhoA@mail.gmail.com>

mod_av is compiled successful with ffmpeg from rpmfusion repo. But during
mod_av load shared lib linking is failed.
Looks as need package properly one of dependence package.

ср, 2 янв. 2019 г., 17:37 Madovsky <infos at madovsky.org>:

> Hi,
>
> what is today the compatibility status fo FS last git and libav version?
> Also is it compatible with ffmpeg too?
>
> thanks
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From s.safarov at gmail.com  Wed Jan  2 17:42:04 2019
From: s.safarov at gmail.com (Sergey Safarov)
Date: Wed, 2 Jan 2019 20:42:04 +0300
Subject: [Freeswitch-users] mod_http_cache and mod_httapi Corrupted wav
 file after download
In-Reply-To: <CAEVfcab3rLkrT=z0E_895tAshHNxwGU8ORLgfF9q5MMq0Xm1eA@mail.gmail.com>
References: <CAEVfcab3rLkrT=z0E_895tAshHNxwGU8ORLgfF9q5MMq0Xm1eA@mail.gmail.com>
Message-ID: <CAHtxdDemRUymW92X2duXDc39zpnfJwLfys6wjze2gNE1=AJWjA@mail.gmail.com>

Need to check downloaded file size using wireshark or web server logs.


ср, 2 янв. 2019 г., 17:46 Babak Yakhchali <babak.freeswitch at gmail.com>:

> Hi
> I'm using freeswitch 1.8.4 64 bit build using visual studio 2017 on win
> 10. When using playback with the file on disk the file plays ok. but if
> using http based sound urls which use http_cache or httapi the file
> playback is distorted.
> I attached the file and the downloaded cached file. The sizes are
> different!!
> everything works fine on debian
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From Alejandro.Castera at on24.com  Wed Jan  2 17:53:41 2019
From: Alejandro.Castera at on24.com (Alejandro Castera)
Date: Wed, 2 Jan 2019 17:53:41 +0000
Subject: [Freeswitch-users] How can we make Freeswitch work with a verto
	client running on Edge?
Message-ID: <em4b12e718-dbb0-470f-ae73-522b1bc2c30a@desktop-e7483b9>

Hello all,

We can use the verto communicator client to join a conference handled by our own Freeswitch server.
It works on Chrome, Firefox and Opera, on which we can connect ok and get audio+video.
But it does NOT work on Edge.

The candidates Edge provides are the following:

candidate:1 1 UDP 2130706431 192.168.0.7 63832 typ host
candidate:2 1 TCP 1684798975 192.168.0.7 63832 typ srflx raddr 192.168.0.7 rport 63832 tcptype active

So for for every local IP, Edge generates 2 candidates: 1 host candidate, and 1 srflx candidate.
The srvflx candidate is built wrong on purpose (transport address == related address).
I guess they do that so that when the ICE connection checks take place, the server side can "learn" a new prflx candidate (see RFC5245, section 7.2.1.3)

I have tested the same scenario using a server which makes use of the libnice library for the ICE implementation, and it works!!!
No TURN server is being used, no STUN server is being used, the candidates Edge sends are the 2 above (which do not contain a public IP) and still the server generates a prflx candidate and the connection is established perfectly.

But when using our Freeswitch server (which does not make use of the libnice library for ICE) the connection does not work.
Then again, the funny thing is that if we use the cantina.freeswitch.org Freeswitch server, then it DOES work!
So there must be something in the code/configuration that could be done so that the Freeswitch server can "learn" a prflx candidate from those Edge is sending out.

Could anyone help us?
Alex
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From s.safarov at gmail.com  Wed Jan  2 18:07:06 2019
From: s.safarov at gmail.com (Sergey Safarov)
Date: Wed, 2 Jan 2019 21:07:06 +0300
Subject: [Freeswitch-users] FreeSWITCH sources
In-Reply-To: <a2f1bd87-f094-1e50-3975-b6b5cf9c3575@bohboh.info>
References: <a2f1bd87-f094-1e50-3975-b6b5cf9c3575@bohboh.info>
Message-ID: <CAHtxdDdxOanR3A=Lo=FsW_FOwB-X4rA4VVAWTpT2uNmYb7gKfA@mail.gmail.com>

You can try disable ipv6 on your station and then clone repo.

ср, 2 янв. 2019 г., 20:48 Social Boh <social at bohboh.info>:

> Hello,
>
> right now I can not download the sources of FreeSWITCH
>
> git clone -b v1.6 https://freeswitch.org/stash/scm/fs/freeswitch.git
>
> fatal: unable to access
> 'https://freeswitch.org/stash/scm/fs/freeswitch.git/': Failed to connect
> to 2803:d000:fffe::151: Network is unreachable
>
> Thank you
>
> Regards
>
> --
> ---
> I'm SoCIaL, MayBe
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From mike at freeswitch.org  Wed Jan  2 20:24:04 2019
From: mike at freeswitch.org (Mike Jerris)
Date: Wed, 2 Jan 2019 15:24:04 -0500
Subject: [Freeswitch-users] about mod_av and ffmpeg/libav
In-Reply-To: <b82ff9a7-2e2b-934b-5279-e7f8af09294c@madovsky.org>
References: <b82ff9a7-2e2b-934b-5279-e7f8af09294c@madovsky.org>
Message-ID: <2E1A63DE-0FEE-475E-BF05-33D398D9FCAE@freeswitch.org>

We recently added some support for newer ffmpeg versions, should work now with latest release.  You would have to test and confirm anything unreleased.  You will want to use the latest FreeSWITCH release to test as that has the patches to support newer versions.

https://freeswitch.org/jira/browse/FS-11523



> On Jan 1, 2019, at 6:36 AM, Madovsky <infos at madovsky.org> wrote:
> 
> Hi,
> 
> what is today the compatibility status fo FS last git and libav version? Also is it compatible with ffmpeg too?
> 
> thanks
> 



From infos at madovsky.org  Wed Jan  2 20:31:43 2019
From: infos at madovsky.org (Madovsky)
Date: Wed, 2 Jan 2019 12:31:43 -0800
Subject: [Freeswitch-users] about mod_av and ffmpeg/libav
In-Reply-To: <CAHtxdDdV3DHiVWwpDJXahxmAPHBPqTvPJKLQN6-4behHexxhoA@mail.gmail.com>
References: <b82ff9a7-2e2b-934b-5279-e7f8af09294c@madovsky.org>
 <CAHtxdDdV3DHiVWwpDJXahxmAPHBPqTvPJKLQN6-4behHexxhoA@mail.gmail.com>
Message-ID: <e8ea7c89-5ef7-edfd-6a0f-3ded32b17e54@madovsky.org>

I'm compiling ffmpeg and libav myself,

that's why I asked, for now I have libav-11.12 especiall for FS

but would like to "harmonize" it with the the last version 12.x and 
ffmpeg-4.1.x

On 1/2/2019 9:39 AM, Sergey Safarov wrote:
> mod_av is compiled successful with ffmpeg from rpmfusion repo. But 
> during mod_av load shared lib linking is failed.
> Looks as need package properly one of dependence package.
>
> ср, 2 янв. 2019 г., 17:37 Madovsky <infos at madovsky.org 
> <mailto:infos at madovsky.org>>:
>
>     Hi,
>
>     what is today the compatibility status fo FS last git and libav
>     version?
>     Also is it compatible with ffmpeg too?
>
>     thanks
>
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Services
>     sales at freeswitch.com <mailto:sales at freeswitch.com>
>     https://freeswitch.com
>
>     Official FreeSWITCH Sites
>     https://freeswitch.com/oss
>     https://freeswitch.org/confluence
>     https://cluecon.com
>
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     https://freeswitch.com
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com

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From infos at madovsky.org  Wed Jan  2 21:34:57 2019
From: infos at madovsky.org (Madovsky)
Date: Wed, 2 Jan 2019 13:34:57 -0800
Subject: [Freeswitch-users] about mod_av and ffmpeg/libav
In-Reply-To: <2E1A63DE-0FEE-475E-BF05-33D398D9FCAE@freeswitch.org>
References: <b82ff9a7-2e2b-934b-5279-e7f8af09294c@madovsky.org>
 <2E1A63DE-0FEE-475E-BF05-33D398D9FCAE@freeswitch.org>
Message-ID: <632e7d5b-b380-e70c-8c28-b06e640780f7@madovsky.org>

Ok Mike,

will test asap

thanks!

On 1/2/2019 12:24 PM, Mike Jerris wrote:
> We recently added some support for newer ffmpeg versions, should work now with latest release.  You would have to test and confirm anything unreleased.  You will want to use the latest FreeSWITCH release to test as that has the patches to support newer versions.
>
> https://freeswitch.org/jira/browse/FS-11523
>
>
>
>> On Jan 1, 2019, at 6:36 AM, Madovsky <infos at madovsky.org> wrote:
>>
>> Hi,
>>
>> what is today the compatibility status fo FS last git and libav version? Also is it compatible with ffmpeg too?
>>
>> thanks
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
>



From mike at freeswitch.org  Wed Jan  2 22:08:25 2019
From: mike at freeswitch.org (Mike Jerris)
Date: Wed, 2 Jan 2019 17:08:25 -0500
Subject: [Freeswitch-users] about mod_av and ffmpeg/libav
In-Reply-To: <e8ea7c89-5ef7-edfd-6a0f-3ded32b17e54@madovsky.org>
References: <b82ff9a7-2e2b-934b-5279-e7f8af09294c@madovsky.org>
 <CAHtxdDdV3DHiVWwpDJXahxmAPHBPqTvPJKLQN6-4behHexxhoA@mail.gmail.com>
 <e8ea7c89-5ef7-edfd-6a0f-3ded32b17e54@madovsky.org>
Message-ID: <CDEE7E2C-101B-4E55-BE4D-197158DF2752@freeswitch.org>

It might work with the latest FreeSWITCH tree.  Ffmpeg 4.1 and later will not work without the patches that just went in.  Try it out and if there are issues in the build we would need ifdefed patches to address them, but there is a good chance that it will build fine.



> On Jan 2, 2019, at 3:31 PM, Madovsky <infos at madovsky.org> wrote:
> 
> I'm compiling ffmpeg and libav myself,
> 
> that's why I asked, for now I have libav-11.12 especiall for FS
> 
> but would like to "harmonize" it with the the last version 12.x and ffmpeg-4.1.x
> 
> On 1/2/2019 9:39 AM, Sergey Safarov wrote:
>> mod_av is compiled successful with ffmpeg from rpmfusion repo. But during mod_av load shared lib linking is failed.
>> Looks as need package properly one of dependence package.
>> 
>> ср, 2 янв. 2019 г., 17:37 Madovsky <infos at madovsky.org <mailto:infos at madovsky.org>>:
>> Hi,
>> 
>> what is today the compatibility status fo FS last git and libav version? 
>> Also is it compatible with ffmpeg too?
>> 
>> thanks
>> 

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From alex at freeswitch.com  Wed Jan  2 23:55:08 2019
From: alex at freeswitch.com (Alexey Sibyakin)
Date: Thu, 3 Jan 2019 08:55:08 +0900
Subject: [Freeswitch-users] How can we make Freeswitch work with a verto
 client running on Edge?
In-Reply-To: <em4b12e718-dbb0-470f-ae73-522b1bc2c30a@desktop-e7483b9>
References: <em4b12e718-dbb0-470f-ae73-522b1bc2c30a@desktop-e7483b9>
Message-ID: <CAGJLwFimw4U8V7ik+DdfLWarAR28DcoSCEE3yoCLqOpULEMZ3w@mail.gmail.com>

Hello,

Please do not repost your questions. Try to use Chrome or wait while Edge
will become Chrome too as Anthony suggested.

Regards,

Alex

On Thu, Jan 3, 2019 at 6:10 AM Alejandro Castera <Alejandro.Castera at on24.com>
wrote:

> Hello all,
>
> We can use the verto communicator client to join a conference handled by
> our own Freeswitch server.
> It works on Chrome, Firefox and Opera, on which we can connect ok and get
> audio+video.
> But it does NOT work on Edge.
>
> The candidates Edge provides are the following:
>
> candidate:1 1 UDP 2130706431 192.168.0.7 63832 typ host
> candidate:2 1 TCP 1684798975 192.168.0.7 63832 typ srflx raddr 192.168.0.7
> rport 63832 tcptype active
>
> So for for every local IP, Edge generates 2 candidates: 1 host candidate,
> and 1 srflx candidate.
> The srvflx candidate is built wrong on purpose (transport address ==
> related address).
> I guess they do that so that when the ICE connection checks take place,
> the server side can "learn" a new prflx candidate (see RFC5245, section
> 7.2.1.3)
>
> I have tested the same scenario using a server which makes use of the
> libnice library for the ICE implementation, and it works!!!
> No TURN server is being used, no STUN server is being used, the candidates
> Edge sends are the 2 above (which do not contain a public IP) and still the
> server generates a prflx candidate and the connection is established
> perfectly.
>
> But when using our Freeswitch server (which does not make use of the
> libnice library for ICE) the connection does not work.
> Then again, the funny thing is that if we use the cantina.freeswitch.org
> Freeswitch server, then it DOES work!
> So there must be something in the code/configuration that could be done so
> that the Freeswitch server can "learn" a prflx candidate from those Edge is
> sending out.
>
> Could anyone help us?
> Alex
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com



-- 
Alex Sibyakin | Support Engineer
FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045
<https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g>
Email: alex at freeswitch.com
Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/>
Need commercial support? Contact sales at freeswitch.com for details.
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From infos at madovsky.org  Thu Jan  3 00:33:42 2019
From: infos at madovsky.org (Madovsky)
Date: Wed, 2 Jan 2019 16:33:42 -0800
Subject: [Freeswitch-users] about mod_av and ffmpeg/libav
In-Reply-To: <2E1A63DE-0FEE-475E-BF05-33D398D9FCAE@freeswitch.org>
References: <b82ff9a7-2e2b-934b-5279-e7f8af09294c@madovsky.org>
 <2E1A63DE-0FEE-475E-BF05-33D398D9FCAE@freeswitch.org>
Message-ID: <a1ce7561-b7dd-2fd7-f6a9-87049baf342c@madovsky.org>

Unfortunately I cannot test mod_av with the las FS git

after a git pull I still cannot compile FS with gcc 8

src/switch_core_sqldb.c: In function ‘switch_cache_db_status’:
src/switch_core_sqldb.c:3781:4: error: ‘strncpy’ output truncated before 
termina ting nul copying as many bytes from a string as its length 
[-Werror=stringop-tru ncation]
     strncpy(cleankey_str, dbh->name, strlen(dbh->name));
     ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
cc1: all warnings being treated as errors

although this thread said it was fixed

https://freeswitch.org/jira/browse/FS-11345


On 1/2/2019 12:24 PM, Mike Jerris wrote:
> We recently added some support for newer ffmpeg versions, should work now with latest release.  You would have to test and confirm anything unreleased.  You will want to use the latest FreeSWITCH release to test as that has the patches to support newer versions.
>
> https://freeswitch.org/jira/browse/FS-11523
>
>
>
>> On Jan 1, 2019, at 6:36 AM, Madovsky <infos at madovsky.org> wrote:
>>
>> Hi,
>>
>> what is today the compatibility status fo FS last git and libav version? Also is it compatible with ffmpeg too?
>>
>> thanks
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
>



From s.safarov at gmail.com  Thu Jan  3 08:48:47 2019
From: s.safarov at gmail.com (Sergey Safarov)
Date: Thu, 3 Jan 2019 11:48:47 +0300
Subject: [Freeswitch-users] about mod_av and ffmpeg/libav
In-Reply-To: <2E1A63DE-0FEE-475E-BF05-33D398D9FCAE@freeswitch.org>
References: <b82ff9a7-2e2b-934b-5279-e7f8af09294c@madovsky.org>
 <2E1A63DE-0FEE-475E-BF05-33D398D9FCAE@freeswitch.org>
Message-ID: <CAHtxdDcpzjhH3d3q_a+zgkVaY+ZWN90TGhv7nD8oqFK65hmcNw@mail.gmail.com>

I have properly loaded compiled from source mod_av after
updating glib2-2.56.1-2.el7.x86_64
more info https://bugs.centos.org/view.php?id=15495

чт, 3 янв. 2019 г. в 01:03, Mike Jerris <mike at freeswitch.org>:

> We recently added some support for newer ffmpeg versions, should work now
> with latest release.  You would have to test and confirm anything
> unreleased.  You will want to use the latest FreeSWITCH release to test as
> that has the patches to support newer versions.
>
> https://freeswitch.org/jira/browse/FS-11523
>
>
>
> > On Jan 1, 2019, at 6:36 AM, Madovsky <infos at madovsky.org> wrote:
> >
> > Hi,
> >
> > what is today the compatibility status fo FS last git and libav version?
> Also is it compatible with ffmpeg too?
> >
> > thanks
> >
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From harangozo.laszlo at tct.hu  Thu Jan  3 12:13:47 2019
From: harangozo.laszlo at tct.hu (=?UTF-8?B?SGFyYW5nb3rDsywgTMOhc3psw7M=?=)
Date: Thu, 3 Jan 2019 13:13:47 +0100
Subject: [Freeswitch-users] spandsp_start_dtmf - same inband DTMF is
 detected multiple times if current playback is stopped with uuid_break
Message-ID: <CAFXzTyg7R1-Fe3hia5VAdvefaK2v7YJmZGXc6ePSB_5=cqS0-g@mail.gmail.com>

Hi,

Any suggestions would be welcome with the following scenario:

External app managing calls trough ESL
We have to detect inband DTMF digits (we generally try to use a
configuration where we don't have to deal with inband DTMF digits, but in
some rare cases it cannot be avoided). There are cases when a playback has
to be stopped when a certain digit is received. Inband DTMFs are detected
with spandsp_start_dtmf. So the following happens:

   1. Inbound call answered
   2. spandsp_start_dtmf called
   3. A file is being played
   4. Client starts to send an inband DTMF
   5. After a very short time FreeSwitch issues the DTMF event
   6. External app stops the playback with uuid_break
   7. External app starts to play a different file
   8. Freeswitch detects the same digit again (which is still present in
   the RTP, the DTMF is on for roughly 400ms, and only ~190ms has elapsed
   since step 5)

The scenario is reproducible and happens in a consistent way. The inband
DTMF is continuous. FreeSwitch reports only 1 DTMF event if the play is not
interrupted with uuid_break.

Is it possible to avoid this sort of multiple detection? Or do I have to
handle this in the external app level?

Is this an issue I should report to JIRA?

*Hari*
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From mike at freeswitch.com  Thu Jan  3 15:31:20 2019
From: mike at freeswitch.com (Mike Jerris)
Date: Thu, 3 Jan 2019 10:31:20 -0500
Subject: [Freeswitch-users] about mod_av and ffmpeg/libav
In-Reply-To: <a1ce7561-b7dd-2fd7-f6a9-87049baf342c@madovsky.org>
References: <b82ff9a7-2e2b-934b-5279-e7f8af09294c@madovsky.org>
 <2E1A63DE-0FEE-475E-BF05-33D398D9FCAE@freeswitch.org>
 <a1ce7561-b7dd-2fd7-f6a9-87049baf342c@madovsky.org>
Message-ID: <CADrq_yGpio1zxd8pUqEbjhBYtCVrhZR2NkhnNuyEom0peHMacA@mail.gmail.com>

i’m happy to take patches that fix these issues but we don’t spend much
time trying to fix issues with bleeding edge distributions, it’s just not
an interest or priority for us.  If you want to use them you will need to
be responsible for fixing the issues and supplying us patches, which we are
happy to review and merge.  If you are unable to make these fixes then i
suggest you use a not so bleeding edge distribution.

On Thu, Jan 3, 2019 at 10:26 AM Madovsky <infos at madovsky.org> wrote:

> Unfortunately I cannot test mod_av with the las FS git
>
> after a git pull I still cannot compile FS with gcc 8
>
> src/switch_core_sqldb.c: In function ‘switch_cache_db_status’:
> src/switch_core_sqldb.c:3781:4: error: ‘strncpy’ output truncated before
> termina ting nul copying as many bytes from a string as its length
> [-Werror=stringop-tru ncation]
>      strncpy(cleankey_str, dbh->name, strlen(dbh->name));
>      ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
> cc1: all warnings being treated as errors
>
> although this thread said it was fixed
>
> https://freeswitch.org/jira/browse/FS-11345
>
>
> On 1/2/2019 12:24 PM, Mike Jerris wrote:
> > We recently added some support for newer ffmpeg versions, should work
> now with latest release.  You would have to test and confirm anything
> unreleased.  You will want to use the latest FreeSWITCH release to test as
> that has the patches to support newer versions.
> >
> > https://freeswitch.org/jira/browse/FS-11523
> >
> >
> >
> >> On Jan 1, 2019, at 6:36 AM, Madovsky <infos at madovsky.org> wrote:
> >>
> >> Hi,
> >>
> >> what is today the compatibility status fo FS last git and libav
> version? Also is it compatible with ffmpeg too?
> >>
> >> thanks
> >>
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Services
> > sales at freeswitch.com
> > https://freeswitch.com
> >
> > Official FreeSWITCH Sites
> > https://freeswitch.com/oss
> > https://freeswitch.org/confluence
> > https://cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > https://freeswitch.com
> >
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From Alejandro.Castera at on24.com  Thu Jan  3 18:08:26 2019
From: Alejandro.Castera at on24.com (Alejandro Castera)
Date: Thu, 3 Jan 2019 18:08:26 +0000
Subject: [Freeswitch-users] How can we make Freeswitch work with a verto
 client running on Edge?
In-Reply-To: <CAGJLwFimw4U8V7ik+DdfLWarAR28DcoSCEE3yoCLqOpULEMZ3w@mail.gmail.com>
References: <em4b12e718-dbb0-470f-ae73-522b1bc2c30a@desktop-e7483b9>
 <CAGJLwFimw4U8V7ik+DdfLWarAR28DcoSCEE3yoCLqOpULEMZ3w@mail.gmail.com>
Message-ID: <emd598dd3c-dbdf-4fb9-a594-92d1354e0fd6@desktop-e7483b9>

Sorry but it is perfectly ok to post my question on the users' list, since the devs were of no help whatsoever.
Maybe a user has run into the same problem and has found a solution to it.
Thus, with all due respect, your remark is inappropriate.



------ Original Message ------
From: "Alexey Sibyakin" <alex at freeswitch.com<mailto:alex at freeswitch.com>>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
Sent: 02-Jan-19 6:55:08 PM
Subject: Re: [Freeswitch-users] How can we make Freeswitch work with a verto client running on Edge?

Hello,

Please do not repost your questions. Try to use Chrome or wait while Edge will become Chrome too as Anthony suggested.

Regards,

Alex

On Thu, Jan 3, 2019 at 6:10 AM Alejandro Castera <Alejandro.Castera at on24.com<mailto:Alejandro.Castera at on24.com>> wrote:
Hello all,

We can use the verto communicator client to join a conference handled by our own Freeswitch server.
It works on Chrome, Firefox and Opera, on which we can connect ok and get audio+video.
But it does NOT work on Edge.

The candidates Edge provides are the following:

candidate:1 1 UDP 2130706431 192.168.0.7 63832 typ host
candidate:2 1 TCP 1684798975 192.168.0.7 63832 typ srflx raddr 192.168.0.7 rport 63832 tcptype active

So for for every local IP, Edge generates 2 candidates: 1 host candidate, and 1 srflx candidate.
The srvflx candidate is built wrong on purpose (transport address == related address).
I guess they do that so that when the ICE connection checks take place, the server side can "learn" a new prflx candidate (see RFC5245, section 7.2.1.3)

I have tested the same scenario using a server which makes use of the libnice library for the ICE implementation, and it works!!!
No TURN server is being used, no STUN server is being used, the candidates Edge sends are the 2 above (which do not contain a public IP) and still the server generates a prflx candidate and the connection is established perfectly.

But when using our Freeswitch server (which does not make use of the libnice library for ICE) the connection does not work.
Then again, the funny thing is that if we use the cantina.freeswitch.org<http://cantina.freeswitch.org> Freeswitch server, then it DOES work!
So there must be something in the code/configuration that could be done so that the Freeswitch server can "learn" a prflx candidate from those Edge is sending out.

Could anyone help us?
Alex
_________________________________________________________________________
Professional FreeSWITCH Services
sales at freeswitch.com<mailto:sales at freeswitch.com>
https://freeswitch.com

Official FreeSWITCH Sites
https://freeswitch.com/oss
https://freeswitch.org/confluence
https://cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
https://freeswitch.com


--
Alex Sibyakin | Support Engineer
FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045<https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g>
Email: alex at freeswitch.com<mailto:alex at freeswitch.com>
Website: https://www.FreeSWITCH.com<https://www.freeswitch.com/>
Need commercial support? Contact sales at freeswitch.com<mailto:sales at freeswitch.com> for details.

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From infos at madovsky.org  Thu Jan  3 18:26:21 2019
From: infos at madovsky.org (Madovsky)
Date: Thu, 3 Jan 2019 10:26:21 -0800
Subject: [Freeswitch-users] about mod_av and ffmpeg/libav
In-Reply-To: <CAHtxdDcpzjhH3d3q_a+zgkVaY+ZWN90TGhv7nD8oqFK65hmcNw@mail.gmail.com>
References: <b82ff9a7-2e2b-934b-5279-e7f8af09294c@madovsky.org>
 <2E1A63DE-0FEE-475E-BF05-33D398D9FCAE@freeswitch.org>
 <CAHtxdDcpzjhH3d3q_a+zgkVaY+ZWN90TGhv7nD8oqFK65hmcNw@mail.gmail.com>
Message-ID: <9380750f-7ccb-b64f-93d9-6cb561f318e9@madovsky.org>

I'm not confident to upgrade glib on my OS which has 
glib-1.2.10-52.fc28.x86_64

On 1/3/2019 12:48 AM, Sergey Safarov wrote:
> I have properly loaded compiled from source mod_av after 
> updating glib2-2.56.1-2.el7.x86_64
> more info https://bugs.centos.org/view.php?id=15495
>
> чт, 3 янв. 2019 г. в 01:03, Mike Jerris <mike at freeswitch.org 
> <mailto:mike at freeswitch.org>>:
>
>     We recently added some support for newer ffmpeg versions, should
>     work now with latest release.  You would have to test and confirm
>     anything unreleased.  You will want to use the latest FreeSWITCH
>     release to test as that has the patches to support newer versions.
>
>     https://freeswitch.org/jira/browse/FS-11523
>
>
>
>     > On Jan 1, 2019, at 6:36 AM, Madovsky <infos at madovsky.org
>     <mailto:infos at madovsky.org>> wrote:
>     >
>     > Hi,
>     >
>     > what is today the compatibility status fo FS last git and libav
>     version? Also is it compatible with ffmpeg too?
>     >
>     > thanks
>     >
>
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Services
>     sales at freeswitch.com <mailto:sales at freeswitch.com>
>     https://freeswitch.com
>
>     Official FreeSWITCH Sites
>     https://freeswitch.com/oss
>     https://freeswitch.org/confluence
>     https://cluecon.com
>
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     https://freeswitch.com
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com

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From nathan at robotics.net  Thu Jan  3 19:49:37 2019
From: nathan at robotics.net (Nathan Stratton)
Date: Thu, 3 Jan 2019 14:49:37 -0500
Subject: [Freeswitch-users] Fwd: switch_core_session.c:538 Could not locate
	channel type $
In-Reply-To: <CAHubkyP3Zu=BMsFk6CosP+X9e3h87+iaKu4crOh6WSG54mGikA@mail.gmail.com>
References: <CAHubkyP3Zu=BMsFk6CosP+X9e3h87+iaKu4crOh6WSG54mGikA@mail.gmail.com>
Message-ID: <CAHubkyPDx0FBqxEA3xpai4W1Z54r=+jJ5tXi3inPAJRbLeWu7g@mail.gmail.com>

New install, I have two users registered and I am trying to call between
them. The error I am getting is:

2019-01-03 19:32:16.148530 [DEBUG] switch_ivr_originate.c:2204 Parsing
global variables
2019-01-03 19:32:16.148530 [ERR] switch_core_session.c:538 Could not locate
channel type $
                        {sofia_contact(*
2019-01-03 19:32:16.148530 [NOTICE] switch_ivr_originate.c:2944 Cannot
create outgoing channel of type [$
                        {sofia_contact(*] cause: [CHAN_NOT_IMPLEMENTED]
2019-01-03 19:32:16.148530 [DEBUG] switch_ivr_originate.c:3941 Originate
Resulted in Error Cause: 66 [CHAN_NOT_IMPLEMENTED]
2019-01-03 19:32:16.148530 [NOTICE] switch_ivr_originate.c:2944 Cannot
create outgoing channel of type [user] cause: [CHAN_NOT_IMPLEMENTED]
2019-01-03 19:32:16.148530 [DEBUG] switch_ivr_originate.c:3941 Originate
Resulted in Error Cause: 66 [CHAN_NOT_IMPLEMENTED]
2019-01-03 19:32:16.148530 [INFO] mod_dptools.c:3518 Originate Failed.
Cause: CHAN_NOT_IMPLEMENTED
EXECUTE sofia/internal/1000 at illumy1.com answer()

Any ideas? This has been driving me crazy.

><>
nathan stratton
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From infos at madovsky.org  Thu Jan  3 19:56:56 2019
From: infos at madovsky.org (Madovsky)
Date: Thu, 3 Jan 2019 11:56:56 -0800
Subject: [Freeswitch-users] about mod_av and ffmpeg/libav
In-Reply-To: <CADrq_yGpio1zxd8pUqEbjhBYtCVrhZR2NkhnNuyEom0peHMacA@mail.gmail.com>
References: <b82ff9a7-2e2b-934b-5279-e7f8af09294c@madovsky.org>
 <2E1A63DE-0FEE-475E-BF05-33D398D9FCAE@freeswitch.org>
 <a1ce7561-b7dd-2fd7-f6a9-87049baf342c@madovsky.org>
 <CADrq_yGpio1zxd8pUqEbjhBYtCVrhZR2NkhnNuyEom0peHMacA@mail.gmail.com>
Message-ID: <24896420-2e89-d29f-b121-22a678271b8d@madovsky.org>

I'm using Fedora since 20 years and won't change all my server just for 
one piece of software.

Fedora is not a  bleeding edge distribution, it's used by institutions 
like governments, companies and so on.

also this error below has nothing to do with the OS, but with GCC 
version 8.1.2 (correct me if I'm wrong)

CentOS has exactly the same architecture than Fedora, so I guess the 
problem is elsewhere.

I would love to make patches, but my knowledge in C language is near zero.

On 1/3/2019 7:31 AM, Mike Jerris wrote:
> i’m happy to take patches that fix these issues but we don’t spend 
> much time trying to fix issues with bleeding edge distributions, it’s 
> just not an interest or priority for us.  If you want to use them you 
> will need to be responsible for fixing the issues and supplying us 
> patches, which we are happy to review and merge.  If you are unable to 
> make these fixes then i suggest you use a not so bleeding edge 
> distribution.
>
> On Thu, Jan 3, 2019 at 10:26 AM Madovsky <infos at madovsky.org 
> <mailto:infos at madovsky.org>> wrote:
>
>     Unfortunately I cannot test mod_av with the las FS git
>
>     after a git pull I still cannot compile FS with gcc 8
>
>     src/switch_core_sqldb.c: In function ‘switch_cache_db_status’:
>     src/switch_core_sqldb.c:3781:4: error: ‘strncpy’ output truncated
>     before
>     termina ting nul copying as many bytes from a string as its length
>     [-Werror=stringop-tru ncation]
>          strncpy(cleankey_str, dbh->name, strlen(dbh->name));
>          ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
>     cc1: all warnings being treated as errors
>
>     although this thread said it was fixed
>
>     https://freeswitch.org/jira/browse/FS-11345
>
>
>     On 1/2/2019 12:24 PM, Mike Jerris wrote:
>     > We recently added some support for newer ffmpeg versions, should
>     work now with latest release.  You would have to test and confirm
>     anything unreleased.  You will want to use the latest FreeSWITCH
>     release to test as that has the patches to support newer versions.
>     >
>     > https://freeswitch.org/jira/browse/FS-11523
>     >
>     >
>     >
>     >> On Jan 1, 2019, at 6:36 AM, Madovsky <infos at madovsky.org
>     <mailto:infos at madovsky.org>> wrote:
>     >>
>     >> Hi,
>     >>
>     >> what is today the compatibility status fo FS last git and libav
>     version? Also is it compatible with ffmpeg too?
>     >>
>     >> thanks
>     >>
>     >
>     >
>     _________________________________________________________________________
>     > Professional FreeSWITCH Services
>     > sales at freeswitch.com <mailto:sales at freeswitch.com>
>     > https://freeswitch.com
>     >
>     > Official FreeSWITCH Sites
>     > https://freeswitch.com/oss
>     > https://freeswitch.org/confluence
>     > https://cluecon.com
>     >
>     > FreeSWITCH-users mailing list
>     > FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     >
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     > https://freeswitch.com
>     >
>
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Services
>     sales at freeswitch.com <mailto:sales at freeswitch.com>
>     https://freeswitch.com
>
>     Official FreeSWITCH Sites
>     https://freeswitch.com/oss
>     https://freeswitch.org/confluence
>     https://cluecon.com
>
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     https://freeswitch.com
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From s.safarov at gmail.com  Thu Jan  3 20:55:13 2019
From: s.safarov at gmail.com (Sergey Safarov)
Date: Thu, 3 Jan 2019 23:55:13 +0300
Subject: [Freeswitch-users] about mod_av and ffmpeg/libav
In-Reply-To: <CADrq_yGpio1zxd8pUqEbjhBYtCVrhZR2NkhnNuyEom0peHMacA@mail.gmail.com>
References: <b82ff9a7-2e2b-934b-5279-e7f8af09294c@madovsky.org>
 <2E1A63DE-0FEE-475E-BF05-33D398D9FCAE@freeswitch.org>
 <a1ce7561-b7dd-2fd7-f6a9-87049baf342c@madovsky.org>
 <CADrq_yGpio1zxd8pUqEbjhBYtCVrhZR2NkhnNuyEom0peHMacA@mail.gmail.com>
Message-ID: <CAHtxdDe_R2BnvAf9qWLmt5GA6EdPiieRtDfTaC33YYuUMjGhsg@mail.gmail.com>

You can looks this PRs
https://freeswitch.org/stash/projects/FS/repos/freeswitch/pull-requests/1561
https://freeswitch.org/stash/projects/FS/repos/freeswitch/pull-requests/1638

I have tested on Fedora 29 and all works well.
Used rpmfusion repo.


чт, 3 янв. 2019 г. в 23:46, Mike Jerris <mike at freeswitch.com>:

> i’m happy to take patches that fix these issues but we don’t spend much
> time trying to fix issues with bleeding edge distributions, it’s just not
> an interest or priority for us.  If you want to use them you will need to
> be responsible for fixing the issues and supplying us patches, which we are
> happy to review and merge.  If you are unable to make these fixes then i
> suggest you use a not so bleeding edge distribution.
>
> On Thu, Jan 3, 2019 at 10:26 AM Madovsky <infos at madovsky.org> wrote:
>
>> Unfortunately I cannot test mod_av with the las FS git
>>
>> after a git pull I still cannot compile FS with gcc 8
>>
>> src/switch_core_sqldb.c: In function ‘switch_cache_db_status’:
>> src/switch_core_sqldb.c:3781:4: error: ‘strncpy’ output truncated before
>> termina ting nul copying as many bytes from a string as its length
>> [-Werror=stringop-tru ncation]
>>      strncpy(cleankey_str, dbh->name, strlen(dbh->name));
>>      ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
>> cc1: all warnings being treated as errors
>>
>> although this thread said it was fixed
>>
>> https://freeswitch.org/jira/browse/FS-11345
>>
>>
>> On 1/2/2019 12:24 PM, Mike Jerris wrote:
>> > We recently added some support for newer ffmpeg versions, should work
>> now with latest release.  You would have to test and confirm anything
>> unreleased.  You will want to use the latest FreeSWITCH release to test as
>> that has the patches to support newer versions.
>> >
>> > https://freeswitch.org/jira/browse/FS-11523
>> >
>> >
>> >
>> >> On Jan 1, 2019, at 6:36 AM, Madovsky <infos at madovsky.org> wrote:
>> >>
>> >> Hi,
>> >>
>> >> what is today the compatibility status fo FS last git and libav
>> version? Also is it compatible with ffmpeg too?
>> >>
>> >> thanks
>> >>
>> >
>> >
>> _________________________________________________________________________
>> > Professional FreeSWITCH Services
>> > sales at freeswitch.com
>> > https://freeswitch.com
>> >
>> > Official FreeSWITCH Sites
>> > https://freeswitch.com/oss
>> > https://freeswitch.org/confluence
>> > https://cluecon.com
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > https://freeswitch.com
>> >
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From infos at madovsky.org  Thu Jan  3 21:03:08 2019
From: infos at madovsky.org (Madovsky)
Date: Thu, 3 Jan 2019 13:03:08 -0800
Subject: [Freeswitch-users] about mod_av and ffmpeg/libav
In-Reply-To: <CAHtxdDcpzjhH3d3q_a+zgkVaY+ZWN90TGhv7nD8oqFK65hmcNw@mail.gmail.com>
References: <b82ff9a7-2e2b-934b-5279-e7f8af09294c@madovsky.org>
 <2E1A63DE-0FEE-475E-BF05-33D398D9FCAE@freeswitch.org>
 <CAHtxdDcpzjhH3d3q_a+zgkVaY+ZWN90TGhv7nD8oqFK65hmcNw@mail.gmail.com>
Message-ID: <3e4899cd-5672-876d-549d-4d51b9286ca4@madovsky.org>

sorry I provided glib and not glib2 package

the actual version for glib2 on Fedora 28 is glib2-2.56.4-1.fc28.x86

On 1/3/2019 12:48 AM, Sergey Safarov wrote:
> I have properly loaded compiled from source mod_av after 
> updating glib2-2.56.1-2.el7.x86_64
> more info https://bugs.centos.org/view.php?id=15495
>
> чт, 3 янв. 2019 г. в 01:03, Mike Jerris <mike at freeswitch.org 
> <mailto:mike at freeswitch.org>>:
>
>     We recently added some support for newer ffmpeg versions, should
>     work now with latest release.  You would have to test and confirm
>     anything unreleased.  You will want to use the latest FreeSWITCH
>     release to test as that has the patches to support newer versions.
>
>     https://freeswitch.org/jira/browse/FS-11523
>
>
>
>     > On Jan 1, 2019, at 6:36 AM, Madovsky <infos at madovsky.org
>     <mailto:infos at madovsky.org>> wrote:
>     >
>     > Hi,
>     >
>     > what is today the compatibility status fo FS last git and libav
>     version? Also is it compatible with ffmpeg too?
>     >
>     > thanks
>     >
>
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Services
>     sales at freeswitch.com <mailto:sales at freeswitch.com>
>     https://freeswitch.com
>
>     Official FreeSWITCH Sites
>     https://freeswitch.com/oss
>     https://freeswitch.org/confluence
>     https://cluecon.com
>
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     https://freeswitch.com
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com

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From mayamatakeshi at gmail.com  Thu Jan  3 21:53:00 2019
From: mayamatakeshi at gmail.com (mayamatakeshi)
Date: Fri, 4 Jan 2019 06:53:00 +0900
Subject: [Freeswitch-users] spandsp_start_dtmf - same inband DTMF is
 detected multiple times if current playback is stopped with uuid_break
In-Reply-To: <CAFXzTyg7R1-Fe3hia5VAdvefaK2v7YJmZGXc6ePSB_5=cqS0-g@mail.gmail.com>
References: <CAFXzTyg7R1-Fe3hia5VAdvefaK2v7YJmZGXc6ePSB_5=cqS0-g@mail.gmail.com>
Message-ID: <CABaNFCYk=3LM4U0keqoGdOmxXzPECKO7XBJBnKdsVBoOFdTsiA@mail.gmail.com>

On Fri, Jan 4, 2019 at 1:09 AM Harangozó, László <harangozo.laszlo at tct.hu>
wrote:

> Hi,
>
> Any suggestions would be welcome with the following scenario:
>
> External app managing calls trough ESL
> We have to detect inband DTMF digits (we generally try to use a
> configuration where we don't have to deal with inband DTMF digits, but in
> some rare cases it cannot be avoided). There are cases when a playback has
> to be stopped when a certain digit is received. Inband DTMFs are detected
> with spandsp_start_dtmf. So the following happens:
>
>    1. Inbound call answered
>    2. spandsp_start_dtmf called
>    3. A file is being played
>    4. Client starts to send an inband DTMF
>    5. After a very short time FreeSwitch issues the DTMF event
>    6. External app stops the playback with uuid_break
>    7. External app starts to play a different file
>    8. Freeswitch detects the same digit again (which is still present in
>    the RTP, the DTMF is on for roughly 400ms, and only ~190ms has elapsed
>    since step 5)
>
> The scenario is reproducible and happens in a consistent way. The inband
> DTMF is continuous. FreeSwitch reports only 1 DTMF event if the play is not
> interrupted with uuid_break.
>
> Is it possible to avoid this sort of multiple detection? Or do I have to
> handle this in the external app level?
>
>
Try setting min_dup_digit_spacing_ms :

https://freeswitch.org/confluence/display/FREESWITCH/min_dup_digit_spacing_ms


I got into this digit duplication issue and solved it using the above.
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From gregor at infomedia.si  Thu Jan  3 21:01:49 2019
From: gregor at infomedia.si (Gregor Nanger)
Date: Thu, 3 Jan 2019 22:01:49 +0100
Subject: [Freeswitch-users] How can we make Freeswitch work with a verto
 client running on Edge?
In-Reply-To: <emd598dd3c-dbdf-4fb9-a594-92d1354e0fd6@desktop-e7483b9>
References: <em4b12e718-dbb0-470f-ae73-522b1bc2c30a@desktop-e7483b9>
 <CAGJLwFimw4U8V7ik+DdfLWarAR28DcoSCEE3yoCLqOpULEMZ3w@mail.gmail.com>
 <emd598dd3c-dbdf-4fb9-a594-92d1354e0fd6@desktop-e7483b9>
Message-ID: <CAC7ttFGO9SFSMcTfsCXnA5iK0AEm8LuvFuQSBsf0Y33q88Mx_Q@mail.gmail.com>

Alejandro, Edge is not fully compliant with Webrtc and hence verto is not
working. I admire your eagerness and welcome you and I think someone just
have to tell you this. But if you are in doubt, try to find solution and
community will be thankfull. Don't take my post as negative, but just to
spare you some time.

Br, Gregor

On Thu, 3 Jan 2019, 20:49 Alejandro Castera <Alejandro.Castera at on24.com
wrote:

> Sorry but it is perfectly ok to post my question on the users' list, since
> the devs were of no help whatsoever.
> Maybe a user has run into the same problem and has found a solution to it.
> Thus, with all due respect, your remark is inappropriate.
>
>
>
> ------ Original Message ------
> From: "Alexey Sibyakin" <alex at freeswitch.com>
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Sent: 02-Jan-19 6:55:08 PM
> Subject: Re: [Freeswitch-users] How can we make Freeswitch work with a
> verto client running on Edge?
>
> Hello,
>
> Please do not repost your questions. Try to use Chrome or wait while Edge
> will become Chrome too as Anthony suggested.
>
> Regards,
>
> Alex
>
> On Thu, Jan 3, 2019 at 6:10 AM Alejandro Castera <
> Alejandro.Castera at on24.com> wrote:
>
>> Hello all,
>>
>> We can use the verto communicator client to join a conference handled by
>> our own Freeswitch server.
>> It works on Chrome, Firefox and Opera, on which we can connect ok and get
>> audio+video.
>> But it does NOT work on Edge.
>>
>> The candidates Edge provides are the following:
>>
>> candidate:1 1 UDP 2130706431 192.168.0.7 63832 typ host
>> candidate:2 1 TCP 1684798975 192.168.0.7 63832 typ srflx raddr
>> 192.168.0.7 rport 63832 tcptype active
>>
>> So for for every local IP, Edge generates 2 candidates: 1 host candidate,
>> and 1 srflx candidate.
>> The srvflx candidate is built wrong on purpose (transport address ==
>> related address).
>> I guess they do that so that when the ICE connection checks take place,
>> the server side can "learn" a new prflx candidate (see RFC5245, section
>> 7.2.1.3)
>>
>> I have tested the same scenario using a server which makes use of the
>> libnice library for the ICE implementation, and it works!!!
>> No TURN server is being used, no STUN server is being used, the
>> candidates Edge sends are the 2 above (which do not contain a public IP)
>> and still the server generates a prflx candidate and the connection is
>> established perfectly.
>>
>> But when using our Freeswitch server (which does not make use of the
>> libnice library for ICE) the connection does not work.
>> Then again, the funny thing is that if we use the cantina.freeswitch.org
>> Freeswitch server, then it DOES work!
>> So there must be something in the code/configuration that could be done
>> so that the Freeswitch server can "learn" a prflx candidate from those Edge
>> is sending out.
>>
>> Could anyone help us?
>> Alex
>> _________________________________________________________________________
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
>
>
> --
> Alex Sibyakin | Support Engineer
> FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045
> <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g>
> Email: alex at freeswitch.com
> Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/>
> Need commercial support? Contact sales at freeswitch.com for details.
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From infos at madovsky.org  Fri Jan  4 13:14:07 2019
From: infos at madovsky.org (Madovsky)
Date: Fri, 4 Jan 2019 05:14:07 -0800
Subject: [Freeswitch-users] about mod_av and ffmpeg/libav
In-Reply-To: <CAHtxdDe_R2BnvAf9qWLmt5GA6EdPiieRtDfTaC33YYuUMjGhsg@mail.gmail.com>
References: <b82ff9a7-2e2b-934b-5279-e7f8af09294c@madovsky.org>
 <2E1A63DE-0FEE-475E-BF05-33D398D9FCAE@freeswitch.org>
 <a1ce7561-b7dd-2fd7-f6a9-87049baf342c@madovsky.org>
 <CADrq_yGpio1zxd8pUqEbjhBYtCVrhZR2NkhnNuyEom0peHMacA@mail.gmail.com>
 <CAHtxdDe_R2BnvAf9qWLmt5GA6EdPiieRtDfTaC33YYuUMjGhsg@mail.gmail.com>
Message-ID: <203f18c5-7088-ac57-6d58-88d01c6b45ea@madovsky.org>

I really don't understand why it does not compile on Fedora 28 since

all the versions with 29 are identical. I also cloned a fresh FS without 
success

On 1/3/2019 12:55 PM, Sergey Safarov wrote:
> You can looks this PRs
> https://freeswitch.org/stash/projects/FS/repos/freeswitch/pull-requests/1561
> https://freeswitch.org/stash/projects/FS/repos/freeswitch/pull-requests/1638
>
> I have tested on Fedora 29 and all works well.
> Used rpmfusion repo.
>
>
> чт, 3 янв. 2019 г. в 23:46, Mike Jerris <mike at freeswitch.com 
> <mailto:mike at freeswitch.com>>:
>
>     i’m happy to take patches that fix these issues but we don’t spend
>     much time trying to fix issues with bleeding edge distributions,
>     it’s just not an interest or priority for us.  If you want to use
>     them you will need to be responsible for fixing the issues and
>     supplying us patches, which we are happy to review and merge.  If
>     you are unable to make these fixes then i suggest you use a not so
>     bleeding edge distribution.
>
>     On Thu, Jan 3, 2019 at 10:26 AM Madovsky <infos at madovsky.org
>     <mailto:infos at madovsky.org>> wrote:
>
>         Unfortunately I cannot test mod_av with the las FS git
>
>         after a git pull I still cannot compile FS with gcc 8
>
>         src/switch_core_sqldb.c: In function ‘switch_cache_db_status’:
>         src/switch_core_sqldb.c:3781:4: error: ‘strncpy’ output
>         truncated before
>         termina ting nul copying as many bytes from a string as its
>         length
>         [-Werror=stringop-tru ncation]
>              strncpy(cleankey_str, dbh->name, strlen(dbh->name));
>              ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
>         cc1: all warnings being treated as errors
>
>         although this thread said it was fixed
>
>         https://freeswitch.org/jira/browse/FS-11345
>
>
>         On 1/2/2019 12:24 PM, Mike Jerris wrote:
>         > We recently added some support for newer ffmpeg versions,
>         should work now with latest release.  You would have to test
>         and confirm anything unreleased.  You will want to use the
>         latest FreeSWITCH release to test as that has the patches to
>         support newer versions.
>         >
>         > https://freeswitch.org/jira/browse/FS-11523
>         >
>         >
>         >
>         >> On Jan 1, 2019, at 6:36 AM, Madovsky <infos at madovsky.org
>         <mailto:infos at madovsky.org>> wrote:
>         >>
>         >> Hi,
>         >>
>         >> what is today the compatibility status fo FS last git and
>         libav version? Also is it compatible with ffmpeg too?
>         >>
>         >> thanks
>         >>
>         >
>         >
>         _________________________________________________________________________
>         > Professional FreeSWITCH Services
>         > sales at freeswitch.com <mailto:sales at freeswitch.com>
>         > https://freeswitch.com
>         >
>         > Official FreeSWITCH Sites
>         > https://freeswitch.com/oss
>         > https://freeswitch.org/confluence
>         > https://cluecon.com
>         >
>         > FreeSWITCH-users mailing list
>         > FreeSWITCH-users at lists.freeswitch.org
>         <mailto:FreeSWITCH-users at lists.freeswitch.org>
>         > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         >
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         > https://freeswitch.com
>         >
>
>
>         _________________________________________________________________________
>         Professional FreeSWITCH Services
>         sales at freeswitch.com <mailto:sales at freeswitch.com>
>         https://freeswitch.com
>
>         Official FreeSWITCH Sites
>         https://freeswitch.com/oss
>         https://freeswitch.org/confluence
>         https://cluecon.com
>
>         FreeSWITCH-users mailing list
>         FreeSWITCH-users at lists.freeswitch.org
>         <mailto:FreeSWITCH-users at lists.freeswitch.org>
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         https://freeswitch.com
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Services
>     sales at freeswitch.com <mailto:sales at freeswitch.com>
>     https://freeswitch.com
>
>     Official FreeSWITCH Sites
>     https://freeswitch.com/oss
>     https://freeswitch.org/confluence
>     https://cluecon.com
>
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     https://freeswitch.com
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com

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From rtraverso86 at gmail.com  Fri Jan  4 13:20:03 2019
From: rtraverso86 at gmail.com (Riccardo Traverso)
Date: Fri, 4 Jan 2019 14:20:03 +0100
Subject: [Freeswitch-users] Missing audio in bridged call
In-Reply-To: <CAHJeUE-FD=VuGumM+vuYwH7nQZ+4xDDhD4Js-qSuqeaz24fUCw@mail.gmail.com>
References: <CAHJeUE-FD=VuGumM+vuYwH7nQZ+4xDDhD4Js-qSuqeaz24fUCw@mail.gmail.com>
Message-ID: <CAHJeUE8yAZDp8JKCZXsCFOxx4g-kQVDx4GUk8vWKGH9TEqgsQw@mail.gmail.com>

Hi everyone,

I'm totally new to FreeSWITCH (and PBX software in general) and I'm trying
to build a small directory IVR to start getting the hang of it. I'm
currently stuck, and I need your help please.

This is the setup. I've got a FreeSWITCH on the first machine with a Lua
extension that contains a basic IVR application. When a user calls the IVR
and performs a choice, I want to connect him to a specific remote SIP
address. I tried with the transfer command at first, but in my
understanding I could only use that if I wanted to connect him to another
user/extension on the same server, which isn't my case. So I tried with
bridging instead.
Right now, the IVR doesn't even ask for a choice, it just plays an audio
file and tries to connect to a fixed remote address for test, say
user at example.com, which in turn again just plays a different prompt.

When I call the Lua extension I correctly hear the audio of the first
prompt, but nothing from the second. I also tried starting voice
recognition on user at example.com, which works instead. Yet, even though
apparently user at example.com can hear me through the bridge, I cannot hear
his prompt. If i call directly user at example.com I hear the prompt correctly.

I do not even know where to start from, any advices, please? Which kind of
configs or logs would be useful to diagnose the problem? Thanks in advance!

Below is the code to my Lua extension.

Best regards,
Riccardo


    session:answer()
    session:setAutoHangup(false);

    playback("/some/welcome/audio.wav")

    local session1 = freeswitch.Session("sofia/external/user at example.com");
    freeswitch.bridge(session, session2);
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From nathan at robotics.net  Fri Jan  4 14:46:20 2019
From: nathan at robotics.net (Nathan Stratton)
Date: Fri, 4 Jan 2019 09:46:20 -0500
Subject: [Freeswitch-users] WSS certs
Message-ID: <CAHubkyN1F1kP1f0amJBWq7JED2xNTF_YVwpZnoHEkGpHB4awgg@mail.gmail.com>

I created my wss.pem from my .crt .key and .ca-bundle from my wildcard cert
from COMODO. I have my internal.xml file configured to use wss on 443, and
in vars.xml I point ca_certs to /etc/freeswitch/certs where my wss.pem file
lives owned by freeswitch:daemon.

However, when I try to test it, I get back the default self signed cert,
not my cert from wss.pem.


nathan at marge cert $ openssl s_client -connect as1-east.illumy1.com:443
CONNECTED(00000003)
depth=0 C = US, CN = FreeSWITCH
verify error:num=18:self signed certificate
verify return:1
depth=0 C = US, CN = FreeSWITCH
verify return:1
---
Certificate chain
 0 s:/C=US/CN=FreeSWITCH
   i:/C=US/CN=FreeSWITCH
---
Server certificate
-----BEGIN CERTIFICATE-----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-----END CERTIFICATE-----
subject=/C=US/CN=FreeSWITCH
issuer=/C=US/CN=FreeSWITCH
---
No client certificate CA names sent
---
SSL handshake has read 1527 bytes and written 863 bytes
---
New, TLSv1/SSLv3, Cipher is AES256-GCM-SHA384
Server public key is 4096 bit
Secure Renegotiation IS supported
Compression: NONE
Expansion: NONE
No ALPN negotiated
SSL-Session:
    Protocol  : TLSv1.2
    Cipher    : AES256-GCM-SHA384
    Session-ID:
99DD9750EDDD173E6E41606FB834F0F5AA4B27AA0CCF8284F8D87F47E607D9A9
    Session-ID-ctx:
    Master-Key:
D1F6C0AD00EB7151098BF0DD68670DE9D4631ACED00CE97EAD684B2670BDE283D34FC85CF7D6CED82FB79C68A150988A
    Key-Arg   : None
    Krb5 Principal: None
    PSK identity: None
    PSK identity hint: None
    TLS session ticket lifetime hint: 300 (seconds)
    TLS session ticket:
    0000 - 9a 90 ee 94 ba 4d da e1-d7 c9 6d f1 bb 86 0b 74
 .....M....m....t
    0010 - 53 d3 62 eb ca 6b 3e 2b-c4 36 f4 34 ff 73 e0 6a
 S.b..k>+.6.4.s.j
    0020 - 79 f7 72 d7 ca 24 fa 60-bb 37 c8 b9 cd df 71 74
 y.r..$.`.7....qt
    0030 - 00 d8 37 c6 a2 ef dc 49-08 15 36 04 45 58 f5 af
 ..7....I..6.EX..
    0040 - 0c 09 66 36 98 34 6f d0-6d cb 4a 6e 9e 2a 67 d1
 ..f6.4o.m.Jn.*g.
    0050 - b2 84 a1 f2 ff 6f 7a 89-6f 92 5f ca 8b 6a 96 d1
 .....oz.o._..j..
    0060 - 7a 18 f4 b8 50 8e 31 d1-d0 9f 52 d0 01 43 ba eb
 z...P.1...R..C..
    0070 - 6b 89 bb 9e 7c 60 dd 16-ce 2e 14 c4 44 ca 32 74
 k...|`......D.2t
    0080 - da 66 fc 17 ac a3 04 29-3d f6 b8 39 c4 c2 48 81
 .f.....)=..9..H.
    0090 - 75 a1 2e 93 bc 2d 23 c5-5d 35 1b 88 1e 75 97 ee
 u....-#.]5...u..

    Start Time: 1546611916
    Timeout   : 300 (sec)
    Verify return code: 18 (self signed certificate)
---
read:errno=0


><>
nathan stratton
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From bjordan at e-teleco.com  Fri Jan  4 20:48:13 2019
From: bjordan at e-teleco.com (bjordan at e-teleco.com)
Date: Fri, 4 Jan 2019 20:48:13 +0000
Subject: [Freeswitch-users] Fwd: switch_core_session.c:538 Could not
 locate	channel type $
In-Reply-To: <CAHubkyPDx0FBqxEA3xpai4W1Z54r=+jJ5tXi3inPAJRbLeWu7g@mail.gmail.com>
References: <CAHubkyP3Zu=BMsFk6CosP+X9e3h87+iaKu4crOh6WSG54mGikA@mail.gmail.com>
 <CAHubkyPDx0FBqxEA3xpai4W1Z54r=+jJ5tXi3inPAJRbLeWu7g@mail.gmail.com>
Message-ID: <DM6PR12MB28434196D1CA4B50B366679B978E0@DM6PR12MB2843.namprd12.prod.outlook.com>

Could you post your dialplans? It looks like you may have something in the wrong spot.

From: FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> On Behalf Of Nathan Stratton
Sent: Thursday, January 3, 2019 11:50 AM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Fwd: switch_core_session.c:538 Could not locate channel type $


New install, I have two users registered and I am trying to call between them. The error I am getting is:

2019-01-03 19:32:16.148530 [DEBUG] switch_ivr_originate.c:2204 Parsing global variables
2019-01-03 19:32:16.148530 [ERR] switch_core_session.c:538 Could not locate channel type $
                        {sofia_contact(*
2019-01-03 19:32:16.148530 [NOTICE] switch_ivr_originate.c:2944 Cannot create outgoing channel of type [$
                        {sofia_contact(*] cause: [CHAN_NOT_IMPLEMENTED]
2019-01-03 19:32:16.148530 [DEBUG] switch_ivr_originate.c:3941 Originate Resulted in Error Cause: 66 [CHAN_NOT_IMPLEMENTED]
2019-01-03 19:32:16.148530 [NOTICE] switch_ivr_originate.c:2944 Cannot create outgoing channel of type [user] cause: [CHAN_NOT_IMPLEMENTED]
2019-01-03 19:32:16.148530 [DEBUG] switch_ivr_originate.c:3941 Originate Resulted in Error Cause: 66 [CHAN_NOT_IMPLEMENTED]
2019-01-03 19:32:16.148530 [INFO] mod_dptools.c:3518 Originate Failed.  Cause: CHAN_NOT_IMPLEMENTED
EXECUTE sofia/internal/1000 at illumy1.com<mailto:1000 at illumy1.com> answer()

Any ideas? This has been driving me crazy.

><>
nathan stratton
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From vma at vallimamod.org  Sun Jan  6 13:54:41 2019
From: vma at vallimamod.org (Vallimamod Abdullah)
Date: Sun, 6 Jan 2019 14:54:41 +0100
Subject: [Freeswitch-users] Fwd: switch_core_session.c:538 Could not
 locate channel type $
In-Reply-To: <CAHubkyPDx0FBqxEA3xpai4W1Z54r=+jJ5tXi3inPAJRbLeWu7g@mail.gmail.com>
References: <CAHubkyP3Zu=BMsFk6CosP+X9e3h87+iaKu4crOh6WSG54mGikA@mail.gmail.com>
 <CAHubkyPDx0FBqxEA3xpai4W1Z54r=+jJ5tXi3inPAJRbLeWu7g@mail.gmail.com>
Message-ID: <E9F599AB-43DD-453C-BC28-AAB820A7D84D@vallimamod.org>

Hi,

It looks like a typo in your config. Check if you don't have a newline between "$" and "{sofia_contact(*" somewhere in your dialplan or directory config files.

Hope this helps.


Best Regards,
-- 
Vallimamod Abdullah
SIP Solutions
vma at sip.solutions
linkedin.com/in/vallimamod
.


> On 3 Jan 2019, at 20:49, Nathan Stratton <nathan at robotics.net> wrote:
> 
> 
> New install, I have two users registered and I am trying to call between them. The error I am getting is:
> 
> 2019-01-03 19:32:16.148530 [DEBUG] switch_ivr_originate.c:2204 Parsing global variables
> 2019-01-03 19:32:16.148530 [ERR] switch_core_session.c:538 Could not locate channel type $
>                         {sofia_contact(*
> 2019-01-03 19:32:16.148530 [NOTICE] switch_ivr_originate.c:2944 Cannot create outgoing channel of type [$
>                         {sofia_contact(*] cause: [CHAN_NOT_IMPLEMENTED]
> 2019-01-03 19:32:16.148530 [DEBUG] switch_ivr_originate.c:3941 Originate Resulted in Error Cause: 66 [CHAN_NOT_IMPLEMENTED]
> 2019-01-03 19:32:16.148530 [NOTICE] switch_ivr_originate.c:2944 Cannot create outgoing channel of type [user] cause: [CHAN_NOT_IMPLEMENTED]
> 2019-01-03 19:32:16.148530 [DEBUG] switch_ivr_originate.c:3941 Originate Resulted in Error Cause: 66 [CHAN_NOT_IMPLEMENTED]
> 2019-01-03 19:32:16.148530 [INFO] mod_dptools.c:3518 Originate Failed.  Cause: CHAN_NOT_IMPLEMENTED
> EXECUTE sofia/internal/1000 at illumy1.com <mailto:1000 at illumy1.com> answer()
> 
> Any ideas? This has been driving me crazy. 
> 
> ><>
> nathan stratton
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
> 
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com

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From mj310792 at gmail.com  Mon Jan  7 09:32:50 2019
From: mj310792 at gmail.com (mittali jangid)
Date: Mon, 7 Jan 2019 15:02:50 +0530
Subject: [Freeswitch-users] Presence not working on resubscribe
In-Reply-To: <CADnqS3Oq27kJTs2WCsOMp9HFcujs6gqmh0o0onTcdU7UsDZw9g@mail.gmail.com>
References: <CADnqS3Oq27kJTs2WCsOMp9HFcujs6gqmh0o0onTcdU7UsDZw9g@mail.gmail.com>
Message-ID: <CADnqS3MV9tbdnSHugsXrzJDXgb7t2E8k9a-o5Zy17r2Rx7C4DQ@mail.gmail.com>

Hi

This is a gentle reminder for help required.

On Mon, Dec 24, 2018 at 11:50 AM mittali jangid <mj310792 at gmail.com> wrote:

> Hi,
>
> I am working on presence in polycom phones. I have BLF key for
> fifo-orbits. Presence is working fine when call is parked in orbit. But if
> phone has lower subscription-expiry and phone resubscribes, then freeswitch
> responds with "Terminated" state notify.
>
> Same is happening for user-directory extension number set as BLF in
> polycom phones.
>
> Also if i add BLF for either fifo-orbit or extension when any call is
> parked or extension is attending call respectively, presence is not working
> as per expectation.
>
> I verified this in latest freeswitch version 1.8.2 and also Master pull.
>
> I would like to know why it is behaving this way. Below link also faced
> similar issue, but it is still not resolved :
>
> http://lists.freeswitch.org/pipermail/freeswitch-users/2013-October/100598.html
>
> Your help is much appreciated. Let me know what information is required
> from my end.
>
>
> Thanks and regards,
> Mittali Jangid
>
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From grcamauer at gmail.com  Fri Jan  4 22:48:27 2019
From: grcamauer at gmail.com (Guillermo Ruiz Camauer)
Date: Fri, 4 Jan 2019 19:48:27 -0300
Subject: [Freeswitch-users] DOS against FreeSwitch
Message-ID: <CAPNTnnq-Wd8LywtvJ9q07LAU2sD3PwfcYZhNB-T559WEaa573w@mail.gmail.com>

Is the core team aware of this?  Any solutions?

http://startrinity.com/VS2/FreeswitchPenetrationTests.aspx#freeswitch_181118



-- 
Guillermo Ruiz Camauer
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From alex at freeswitch.com  Fri Jan  4 23:47:37 2019
From: alex at freeswitch.com (Alexey Sibyakin)
Date: Sat, 5 Jan 2019 08:47:37 +0900
Subject: [Freeswitch-users] Fwd: switch_core_session.c:538 Could not
 locate channel type $
In-Reply-To: <CAHubkyPDx0FBqxEA3xpai4W1Z54r=+jJ5tXi3inPAJRbLeWu7g@mail.gmail.com>
References: <CAHubkyP3Zu=BMsFk6CosP+X9e3h87+iaKu4crOh6WSG54mGikA@mail.gmail.com>
 <CAHubkyPDx0FBqxEA3xpai4W1Z54r=+jJ5tXi3inPAJRbLeWu7g@mail.gmail.com>
Message-ID: <CAGJLwFhMzfwkO3t8kQ50M7cKz9J5stfGaAjj+PGq8PUk95zxPw@mail.gmail.com>

What's your dial-string and corresponding dialplan?

On Fri, Jan 4, 2019 at 10:44 PM Nathan Stratton <nathan at robotics.net> wrote:

>
> New install, I have two users registered and I am trying to call between
> them. The error I am getting is:
>
> 2019-01-03 19:32:16.148530 [DEBUG] switch_ivr_originate.c:2204 Parsing
> global variables
> 2019-01-03 19:32:16.148530 [ERR] switch_core_session.c:538 Could not
> locate channel type $
>                         {sofia_contact(*
> 2019-01-03 19:32:16.148530 [NOTICE] switch_ivr_originate.c:2944 Cannot
> create outgoing channel of type [$
>                         {sofia_contact(*] cause: [CHAN_NOT_IMPLEMENTED]
> 2019-01-03 19:32:16.148530 [DEBUG] switch_ivr_originate.c:3941 Originate
> Resulted in Error Cause: 66 [CHAN_NOT_IMPLEMENTED]
> 2019-01-03 19:32:16.148530 [NOTICE] switch_ivr_originate.c:2944 Cannot
> create outgoing channel of type [user] cause: [CHAN_NOT_IMPLEMENTED]
> 2019-01-03 19:32:16.148530 [DEBUG] switch_ivr_originate.c:3941 Originate
> Resulted in Error Cause: 66 [CHAN_NOT_IMPLEMENTED]
> 2019-01-03 19:32:16.148530 [INFO] mod_dptools.c:3518 Originate Failed.
> Cause: CHAN_NOT_IMPLEMENTED
> EXECUTE sofia/internal/1000 at illumy1.com answer()
>
> Any ideas? This has been driving me crazy.
>
> ><>
> nathan stratton
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com



-- 
Alex Sibyakin | Support Engineer
FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045
<https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g>
Email: alex at freeswitch.com
Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/>
Need commercial support? Contact sales at freeswitch.com for details.
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From Rex.Lin at quantatw.com  Mon Jan  7 02:22:24 2019
From: Rex.Lin at quantatw.com (=?big5?B?UmV4IExpbiAoqkysUr5lKQ==?=)
Date: Mon, 7 Jan 2019 02:22:24 +0000
Subject: [Freeswitch-users] Cipher Suite not working
Message-ID: <HK0PR04MB2772A7C60CFDBB75FA8B008C82890@HK0PR04MB2772.apcprd04.prod.outlook.com>

Currently we are using TLS for client/server communication, I would like to do debug using wireshark but got problem in decrypting.
I have seen Server Hello with only one single Cipher Suite of ECDHE, I’ve tried changing “vars” into AES256-SHA, though with the same result;
I wonder if anything had been missed or I’m in a wrong way?

Below is the detail, TKS for any advice.

Scenario: Client --> Kamailio PSTN --> External in FreeSWITCH
FreeSWITCH version: 1.6.20-37-987c9b9~64bit (-37-987c9b9 64bit)

----------------------------Wireshark debug----------------------------
private key: agent.pem
FreeSWITCH to Kamailio
Cipher Suite : TLS_ECDHE_RSA_WITH_AES_256_CBC_SHA (0xc014)

----------------------------vars.xml---------------------------------------
<X-PRE-PROCESS cmd="set" data="sip_tls_ciphers=AES256-SHA"/>


Best Regards,
Rex Lin


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From Rex.Lin at quantatw.com  Tue Jan  8 02:33:48 2019
From: Rex.Lin at quantatw.com (=?big5?B?UmV4IExpbiAoqkysUr5lKQ==?=)
Date: Tue, 8 Jan 2019 02:33:48 +0000
Subject: [Freeswitch-users] Cipher Suite not working
Message-ID: <HK0PR04MB2772DB7A6A53E4F19A0A0063828A0@HK0PR04MB2772.apcprd04.prod.outlook.com>

Hello,

Currently we are using TLS for client/server communication, I would like to do debug using wireshark but got problem in decrypting.
I have seen Server Hello with only one single Cipher Suite of ECDHE, I've tried changing "vars" into AES256-SHA, though with the same result;
I wonder if anything had been missed or I'm in a wrong way?

Below is the detail, TKS for any advice.

Scenario: Client --> Kamailio PSTN --> External in FreeSWITCH
FreeSWITCH version: 1.6.20-37-987c9b9~64bit (-37-987c9b9 64bit)
 
----------------------------Wireshark debug----------------------------
private key: agent.pem
FreeSWITCH to Kamailio
Cipher Suite : TLS_ECDHE_RSA_WITH_AES_256_CBC_SHA (0xc014)
 
----------------------------vars.xml---------------------------------------
<X-PRE-PROCESS cmd="set" data="sip_tls_ciphers=AES256-SHA"/>



Best Regards,
Rex Lin

From gmaruzz at gmail.com  Tue Jan  8 11:23:01 2019
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Tue, 8 Jan 2019 12:23:01 +0100
Subject: [Freeswitch-users] Missing audio in bridged call
In-Reply-To: <CAHJeUE8yAZDp8JKCZXsCFOxx4g-kQVDx4GUk8vWKGH9TEqgsQw@mail.gmail.com>
References: <CAHJeUE-FD=VuGumM+vuYwH7nQZ+4xDDhD4Js-qSuqeaz24fUCw@mail.gmail.com>
 <CAHJeUE8yAZDp8JKCZXsCFOxx4g-kQVDx4GUk8vWKGH9TEqgsQw@mail.gmail.com>
Message-ID: <CALXCt0rsgy3fYMmSN1XKt-A4d4cJQVPbvgNWn8cR3KotnJhTjQ@mail.gmail.com>

before bridging, wait until session1 is connected

eg, after creating session1, insert line

if (session1:ready()) then

and put a line with just

end

affter bridging

For an old example, don't know if is still working, look at:

https://freeswitch.org/confluence/display/FREESWITCH/Lua+example+Bridging+two+calls+with+retry



On Tue, Jan 8, 2019 at 11:41 AM Riccardo Traverso <rtraverso86 at gmail.com>
wrote:

>
> Hi everyone,
>
> I'm totally new to FreeSWITCH (and PBX software in general) and I'm trying
> to build a small directory IVR to start getting the hang of it. I'm
> currently stuck, and I need your help please.
>
> This is the setup. I've got a FreeSWITCH on the first machine with a Lua
> extension that contains a basic IVR application. When a user calls the IVR
> and performs a choice, I want to connect him to a specific remote SIP
> address. I tried with the transfer command at first, but in my
> understanding I could only use that if I wanted to connect him to another
> user/extension on the same server, which isn't my case. So I tried with
> bridging instead.
> Right now, the IVR doesn't even ask for a choice, it just plays an audio
> file and tries to connect to a fixed remote address for test, say
> user at example.com, which in turn again just plays a different prompt.
>
> When I call the Lua extension I correctly hear the audio of the first
> prompt, but nothing from the second. I also tried starting voice
> recognition on user at example.com, which works instead. Yet, even though
> apparently user at example.com can hear me through the bridge, I cannot hear
> his prompt. If i call directly user at example.com I hear the prompt
> correctly.
>
> I do not even know where to start from, any advices, please? Which kind of
> configs or logs would be useful to diagnose the problem? Thanks in advance!
>
> Below is the code to my Lua extension.
>
> Best regards,
> Riccardo
>
>
>     session:answer()
>     session:setAutoHangup(false);
>
>     playback("/some/welcome/audio.wav")
>
>     local session1 = freeswitch.Session("sofia/external/user at example.com
> ");
>     freeswitch.bridge(session, session2);
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com



-- 
Sincerely,

Giovanni Maruzzelli
OpenTelecom.IT
cell: +39 347 266 56 18
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From nathan at robotics.net  Tue Jan  8 13:38:16 2019
From: nathan at robotics.net (Nathan Stratton)
Date: Tue, 8 Jan 2019 08:38:16 -0500
Subject: [Freeswitch-users] Fwd: switch_core_session.c:538 Could not
 locate channel type $
In-Reply-To: <E9F599AB-43DD-453C-BC28-AAB820A7D84D@vallimamod.org>
References: <CAHubkyP3Zu=BMsFk6CosP+X9e3h87+iaKu4crOh6WSG54mGikA@mail.gmail.com>
 <CAHubkyPDx0FBqxEA3xpai4W1Z54r=+jJ5tXi3inPAJRbLeWu7g@mail.gmail.com>
 <E9F599AB-43DD-453C-BC28-AAB820A7D84D@vallimamod.org>
Message-ID: <CAHubkyPh7wEu7MVRv4RCCfbpVPyKFLukdjfTVfwA7BC54izpBg@mail.gmail.com>

I ended up getting past it and calls working, but still not sure why it was
not working. The config did not have any typos, it was the default from RPM:

<action application="bridge" data="user/${dialed_extension}@
${domain_name}"/>

I changed it to:

<action application="bridge"
data="sofia/internal/${dialed_extension}%${domain_name}"/>

And now it works.


><>
nathan stratton


On Tue, Jan 8, 2019 at 6:02 AM Vallimamod Abdullah <vma at vallimamod.org>
wrote:

> Hi,
>
> It looks like a typo in your config. Check if you don't have a newline
> between "$" and "{sofia_contact(*" somewhere in your dialplan or directory
> config files.
>
> Hope this helps.
>
>
> Best Regards,
> --
> Vallimamod Abdullah
> SIP Solutions
> vma at sip.solutions
> linkedin.com/in/vallimamod
> .
>
>
> On 3 Jan 2019, at 20:49, Nathan Stratton <nathan at robotics.net> wrote:
>
>
> New install, I have two users registered and I am trying to call between
> them. The error I am getting is:
>
> 2019-01-03 19:32:16.148530 [DEBUG] switch_ivr_originate.c:2204 Parsing
> global variables
> 2019-01-03 19:32:16.148530 [ERR] switch_core_session.c:538 Could not
> locate channel type $
>                         {sofia_contact(*
> 2019-01-03 19:32:16.148530 [NOTICE] switch_ivr_originate.c:2944 Cannot
> create outgoing channel of type [$
>                         {sofia_contact(*] cause: [CHAN_NOT_IMPLEMENTED]
> 2019-01-03 19:32:16.148530 [DEBUG] switch_ivr_originate.c:3941 Originate
> Resulted in Error Cause: 66 [CHAN_NOT_IMPLEMENTED]
> 2019-01-03 19:32:16.148530 [NOTICE] switch_ivr_originate.c:2944 Cannot
> create outgoing channel of type [user] cause: [CHAN_NOT_IMPLEMENTED]
> 2019-01-03 19:32:16.148530 [DEBUG] switch_ivr_originate.c:3941 Originate
> Resulted in Error Cause: 66 [CHAN_NOT_IMPLEMENTED]
> 2019-01-03 19:32:16.148530 [INFO] mod_dptools.c:3518 Originate Failed.
> Cause: CHAN_NOT_IMPLEMENTED
> EXECUTE sofia/internal/1000 at illumy1.com answer()
>
> Any ideas? This has been driving me crazy.
>
> ><>
> nathan stratton
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From nathan at robotics.net  Tue Jan  8 14:11:19 2019
From: nathan at robotics.net (Nathan Stratton)
Date: Tue, 8 Jan 2019 09:11:19 -0500
Subject: [Freeswitch-users] WSS certs
In-Reply-To: <CAHubkyN1F1kP1f0amJBWq7JED2xNTF_YVwpZnoHEkGpHB4awgg@mail.gmail.com>
References: <CAHubkyN1F1kP1f0amJBWq7JED2xNTF_YVwpZnoHEkGpHB4awgg@mail.gmail.com>
Message-ID: <CAHubkyNH1zT4gGrvnVT7obY79k_aXwZYe9-j5WtvnEWO011YyA@mail.gmail.com>

In case anyone else runs into this, you need:

<param name="tls-cert-dir" value="/etc/freeswitch/certs"/>

I thought it was just for tls, but it's also used to point to wss.pem.

><>
nathan stratton


On Fri, Jan 4, 2019 at 9:46 AM Nathan Stratton <nathan at robotics.net> wrote:

> I created my wss.pem from my .crt .key and .ca-bundle from my wildcard
> cert from COMODO. I have my internal.xml file configured to use wss on 443,
> and in vars.xml I point ca_certs to /etc/freeswitch/certs where my
> wss.pem file lives owned by freeswitch:daemon.
>
> However, when I try to test it, I get back the default self signed cert,
> not my cert from wss.pem.
>
>
> nathan at marge cert $ openssl s_client -connect as1-east.illumy1.com:443
> CONNECTED(00000003)
> depth=0 C = US, CN = FreeSWITCH
> verify error:num=18:self signed certificate
> verify return:1
> depth=0 C = US, CN = FreeSWITCH
> verify return:1
> ---
> Certificate chain
>  0 s:/C=US/CN=FreeSWITCH
>    i:/C=US/CN=FreeSWITCH
> ---
> Server certificate
> -----BEGIN CERTIFICATE-----
> MIIEujCCAqICAQAwDQYJKoZIhvcNAQEFBQAwIjELMAkGA1UEBhMCVVMxEzARBgNV
> BAMMCkZyZWVTV0lUQ0gwIBcNMTgxMjI3MDEzOTIyWhgPMjExODEyMTAwMTM5MjJa
> MCIxCzAJBgNVBAYTAlVTMRMwEQYDVQQDDApGcmVlU1dJVENIMIICIjANBgkqhkiG
> 9w0BAQEFAAOCAg8AMIICCgKCAgEA/xUekzi8uf0ea+/GreneBm3sm9IsQ7L1Yfha
> 8hSvxaX8ElZIlUHUudkCHoreUgoN+AX/F/I4BY93zfAooZ0+q7CVLfJiLsnW5+Do
> 3o0eDXQWQ1qWm1a2tv4h7pFWTM9erGDIharhIUj45CJhtKM2Z5TxbRIp2HtAOMen
> N5M1v+zni7xKS0AOoY6H3i0qHnAeQt5QrpC11575/+5aEWW777W18v5iup9Cn7sR
> 4LxCdQrnJ9UzthNDvkLz5jYX10JZibVs/DehURv9jimVUYaan1fOzhDtVQh/av22
> m4KlTB8xzPSAm0TooRcB0zNbyXCAbnvl9E67orZrxvTzmaKxaPHkGPTqBN962Ti6
> TGSYlz31nKGNeABACSbDSRkRZcnv96+VMo6FKoppHpJISXTZwRQhOJ9Im7HVwISE
> zqhOgDSMo64DcCyif3LOL/gesRjPkc439HulLikDBBS9oAZq8vNg8x8FPA/urpka
> I+mLPTiE39o7vlb6CeBbGeQktUTB+egun8sBYi+DHXW4lX07HLFM6lqhqO8ZYNqY
> 1hEcPZY0GovVNlPVvebCIJhti/bBa/5EAwBGVJnEWjqTTYeIn1jF8eAxAMFHw94P
> RwWAOUgVmq9c5GuRTaw9QWkYg/4Hr4PojGMAIaD0R6m60fIEGOLkzBEUS1Wa6mqK
> EZKy6dkCAwEAATANBgkqhkiG9w0BAQUFAAOCAgEAoFXoWVf/in6dYKWgxSIOsUWA
> yyZiGOexO5P/WW5dVoQ0P67iE2wHkABMTkFe4ir3fHlyeKbbcCB3bU28rsPg/wwo
> P0TIbKNAucrwZ8JKhVQErri/bCYMuctdEN1YxqgQh4YVHYs2/tLr3koqD73crpUL
> aiq0DNWxx6nbTu7223b40zvKjzLNcjuD6DnKAeMaSdsYjfDtrLk5D1WzMXmG1jzu
> wwTAHVn0ru0aiQr3dSpUOD8/V+JqCLO7FbrJL6hpd6NemMasdUjgIr1FenuOmyXn
> A0PFIfQgW1LBlJP1UEGW+yWnVFBNn6pS5AwreVWpS1Tsewa8TTPB/A7ZUAlUb4Lg
> RsvaBc/56ACG9X2DqOBeYUaK/1Hio4/0n29EpB0zN2R0PPOV2QzBScMecIqbGyf3
> gstrMM8KG0GyZRTVOhElWkcgrxre6jM4bzTtOiaZD752pBrYP4EVGtf+oyC9UlKx
> 7ruCkYuNgyGzJgFfSC8s8zYOBAged3aggQYAL4k7rG3uUTnqOHmQg6XFghWCvc3j
> I+TEVKeaGhobcKmZp6CwTjmr63in2D1Kn902wVE6WVdHhJSxT1kuIGf3UYhUWZ6Y
> iBdtAKKhd8QmUYvhzpEHyTPV9bVrrIJJRQW3kzB8jVuyrSYYWnxibcpVxAE1CyHb
> IlJagm7ZFZDqZ4Gn4TM=
> -----END CERTIFICATE-----
> subject=/C=US/CN=FreeSWITCH
> issuer=/C=US/CN=FreeSWITCH
> ---
> No client certificate CA names sent
> ---
> SSL handshake has read 1527 bytes and written 863 bytes
> ---
> New, TLSv1/SSLv3, Cipher is AES256-GCM-SHA384
> Server public key is 4096 bit
> Secure Renegotiation IS supported
> Compression: NONE
> Expansion: NONE
> No ALPN negotiated
> SSL-Session:
>     Protocol  : TLSv1.2
>     Cipher    : AES256-GCM-SHA384
>     Session-ID:
> 99DD9750EDDD173E6E41606FB834F0F5AA4B27AA0CCF8284F8D87F47E607D9A9
>     Session-ID-ctx:
>     Master-Key:
> D1F6C0AD00EB7151098BF0DD68670DE9D4631ACED00CE97EAD684B2670BDE283D34FC85CF7D6CED82FB79C68A150988A
>     Key-Arg   : None
>     Krb5 Principal: None
>     PSK identity: None
>     PSK identity hint: None
>     TLS session ticket lifetime hint: 300 (seconds)
>     TLS session ticket:
>     0000 - 9a 90 ee 94 ba 4d da e1-d7 c9 6d f1 bb 86 0b 74
>  .....M....m....t
>     0010 - 53 d3 62 eb ca 6b 3e 2b-c4 36 f4 34 ff 73 e0 6a
>  S.b..k>+.6.4.s.j
>     0020 - 79 f7 72 d7 ca 24 fa 60-bb 37 c8 b9 cd df 71 74
>  y.r..$.`.7....qt
>     0030 - 00 d8 37 c6 a2 ef dc 49-08 15 36 04 45 58 f5 af
>  ..7....I..6.EX..
>     0040 - 0c 09 66 36 98 34 6f d0-6d cb 4a 6e 9e 2a 67 d1
>  ..f6.4o.m.Jn.*g.
>     0050 - b2 84 a1 f2 ff 6f 7a 89-6f 92 5f ca 8b 6a 96 d1
>  .....oz.o._..j..
>     0060 - 7a 18 f4 b8 50 8e 31 d1-d0 9f 52 d0 01 43 ba eb
>  z...P.1...R..C..
>     0070 - 6b 89 bb 9e 7c 60 dd 16-ce 2e 14 c4 44 ca 32 74
>  k...|`......D.2t
>     0080 - da 66 fc 17 ac a3 04 29-3d f6 b8 39 c4 c2 48 81
>  .f.....)=..9..H.
>     0090 - 75 a1 2e 93 bc 2d 23 c5-5d 35 1b 88 1e 75 97 ee
>  u....-#.]5...u..
>
>     Start Time: 1546611916
>     Timeout   : 300 (sec)
>     Verify return code: 18 (self signed certificate)
> ---
> read:errno=0
>
>
> ><>
> nathan stratton
>
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From tg-maillistings at level5.de  Tue Jan  8 17:30:16 2019
From: tg-maillistings at level5.de (=?UTF-8?Q?Thorsten_G=c3=b6llner?=)
Date: Tue, 8 Jan 2019 18:30:16 +0100
Subject: [Freeswitch-users] 3CX-Sip-Trunk / Freeswitch cannot reach 3CX
In-Reply-To: <34d3fea8-1a28-b495-386a-a787469e99e3@level5.de>
References: <34d3fea8-1a28-b495-386a-a787469e99e3@level5.de>
Message-ID: <18543a2f-5e01-1d55-440d-a52da2033b6a@level5.de>

Hi,

solved:

you have to fill out "Outbound Proxy" (same ip/domain as
Registrar/Server/ Hostname). This forces 3CX to answer on Port 5060 to
freeswitch and connection will be established. Else 3CX answers on the
wrong (not opened) port.

Best regard, Thorsten

Am 12.12.2018 um 16:52 schrieb Thorsten Göllner:
> Hi,
>
> has anybody experience with 3CX-SIP-Trunks? I have installed Freeswitch
> and added an extension (trunk-1). On 3CX-Side I added a SIP-Trunk
> (generic sip provider) and I can see a succesfull registration. Now I
> make a call to Freeswitch and want to pass it to 3CX-Sip-Trunk. But when
> bridging to user/trunk-1 I get a "Originate Resulted in Error Cause: 41
> [NORMAL_TEMPORARY_FAILURE]" error (with no delay).
>
> 3CX is installed as a VM in a Azure-Cloud behind NAT. I disabled local
> iptables and the Firewall-Test on 3CX is all fine.
>
> Freeswitch is on bare metal with a public ip.
>
> When making a call, wireshark on 3CX-machine shows the INVITE followed
> by "Destination unreachable (Port unreachable)". But I am not sure, if
> this is a firewall-issue or something else.
>
> Any idea?
>
> Thanks in advance,
> Thorsten
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com




From bipin at xbipin.com  Tue Jan  8 15:00:39 2019
From: bipin at xbipin.com (Bipin Patel)
Date: Tue, 8 Jan 2019 19:00:39 +0400
Subject: [Freeswitch-users] WSS certs
In-Reply-To: <CAHubkyN1F1kP1f0amJBWq7JED2xNTF_YVwpZnoHEkGpHB4awgg@mail.gmail.com>
References: <CAHubkyN1F1kP1f0amJBWq7JED2xNTF_YVwpZnoHEkGpHB4awgg@mail.gmail.com>
Message-ID: <d9dc8a81-239f-37f4-9f73-0868efa204ec@xbipin.com>

- create wss.pem file for freeswitch and format it likes this and save 
in freeswitch certs folder

-----BEGIN CERTIFICATE-----
<lots of gibberish from the actual certificate>
-----END CERTIFICATE-----
-----BEGIN RSA PRIVATE KEY-----
<lots of gibberish from the actual certificate>
-----END RSA PRIVATE KEY-----
-----BEGIN CERTIFICATE-----
<lots of gibberish from the intermediate certificate>
-----END CERTIFICATE-----
-----BEGIN CERTIFICATE-----
<even more gibberish from the root certificate>
-----END CERTIFICATE-----

- tls.pem and agent.pem will have this

-----BEGIN CERTIFICATE-----
<lots of gibberish from the actual certificate>
-----END CERTIFICATE-----
-----BEGIN RSA PRIVATE KEY-----
<lots of gibberish from the actual certificate>
-----END RSA PRIVATE KEY-----

- cafile.pem will have this

-----BEGIN CERTIFICATE-----
<lots of gibberish from the intermediate certificate>
-----END CERTIFICATE-----
-----BEGIN CERTIFICATE-----
<even more gibberish from the root certificate>
-----END CERTIFICATE-----


Regards,
Bipin


------------------------------------------------------------------------
On 4/1/2019 6:46 PM, Nathan Stratton wrote:
> I created my wss.pem from my .crt .key and .ca-bundle from my wildcard 
> cert from COMODO. I have my internal.xml file configured to use wss on 
> 443, and in vars.xml I point ca_certs to /etc/freeswitch/certs where 
> my wss.pem file lives owned by freeswitch:daemon.
>
> However, when I try to test it, I get back the default self 
> signed cert, not my cert from wss.pem.
>
>
> nathan at marge cert $ openssl s_client -connect as1-east.illumy1.com:443 
> <http://as1-east.illumy1.com:443>
> CONNECTED(00000003)
> depth=0 C = US, CN = FreeSWITCH
> verify error:num=18:self signed certificate
> verify return:1
> depth=0 C = US, CN = FreeSWITCH
> verify return:1
> ---
> Certificate chain
>  0 s:/C=US/CN=FreeSWITCH
>    i:/C=US/CN=FreeSWITCH
> ---
> Server certificate
> -----BEGIN CERTIFICATE-----
> MIIEujCCAqICAQAwDQYJKoZIhvcNAQEFBQAwIjELMAkGA1UEBhMCVVMxEzARBgNV
> BAMMCkZyZWVTV0lUQ0gwIBcNMTgxMjI3MDEzOTIyWhgPMjExODEyMTAwMTM5MjJa
> MCIxCzAJBgNVBAYTAlVTMRMwEQYDVQQDDApGcmVlU1dJVENIMIICIjANBgkqhkiG
> 9w0BAQEFAAOCAg8AMIICCgKCAgEA/xUekzi8uf0ea+/GreneBm3sm9IsQ7L1Yfha
> 8hSvxaX8ElZIlUHUudkCHoreUgoN+AX/F/I4BY93zfAooZ0+q7CVLfJiLsnW5+Do
> 3o0eDXQWQ1qWm1a2tv4h7pFWTM9erGDIharhIUj45CJhtKM2Z5TxbRIp2HtAOMen
> N5M1v+zni7xKS0AOoY6H3i0qHnAeQt5QrpC11575/+5aEWW777W18v5iup9Cn7sR
> 4LxCdQrnJ9UzthNDvkLz5jYX10JZibVs/DehURv9jimVUYaan1fOzhDtVQh/av22
> m4KlTB8xzPSAm0TooRcB0zNbyXCAbnvl9E67orZrxvTzmaKxaPHkGPTqBN962Ti6
> TGSYlz31nKGNeABACSbDSRkRZcnv96+VMo6FKoppHpJISXTZwRQhOJ9Im7HVwISE
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> 1hEcPZY0GovVNlPVvebCIJhti/bBa/5EAwBGVJnEWjqTTYeIn1jF8eAxAMFHw94P
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> EZKy6dkCAwEAATANBgkqhkiG9w0BAQUFAAOCAgEAoFXoWVf/in6dYKWgxSIOsUWA
> yyZiGOexO5P/WW5dVoQ0P67iE2wHkABMTkFe4ir3fHlyeKbbcCB3bU28rsPg/wwo
> P0TIbKNAucrwZ8JKhVQErri/bCYMuctdEN1YxqgQh4YVHYs2/tLr3koqD73crpUL
> aiq0DNWxx6nbTu7223b40zvKjzLNcjuD6DnKAeMaSdsYjfDtrLk5D1WzMXmG1jzu
> wwTAHVn0ru0aiQr3dSpUOD8/V+JqCLO7FbrJL6hpd6NemMasdUjgIr1FenuOmyXn
> A0PFIfQgW1LBlJP1UEGW+yWnVFBNn6pS5AwreVWpS1Tsewa8TTPB/A7ZUAlUb4Lg
> RsvaBc/56ACG9X2DqOBeYUaK/1Hio4/0n29EpB0zN2R0PPOV2QzBScMecIqbGyf3
> gstrMM8KG0GyZRTVOhElWkcgrxre6jM4bzTtOiaZD752pBrYP4EVGtf+oyC9UlKx
> 7ruCkYuNgyGzJgFfSC8s8zYOBAged3aggQYAL4k7rG3uUTnqOHmQg6XFghWCvc3j
> I+TEVKeaGhobcKmZp6CwTjmr63in2D1Kn902wVE6WVdHhJSxT1kuIGf3UYhUWZ6Y
> iBdtAKKhd8QmUYvhzpEHyTPV9bVrrIJJRQW3kzB8jVuyrSYYWnxibcpVxAE1CyHb
> IlJagm7ZFZDqZ4Gn4TM=
> -----END CERTIFICATE-----
> subject=/C=US/CN=FreeSWITCH
> issuer=/C=US/CN=FreeSWITCH
> ---
> No client certificate CA names sent
> ---
> SSL handshake has read 1527 bytes and written 863 bytes
> ---
> New, TLSv1/SSLv3, Cipher is AES256-GCM-SHA384
> Server public key is 4096 bit
> Secure Renegotiation IS supported
> Compression: NONE
> Expansion: NONE
> No ALPN negotiated
> SSL-Session:
>     Protocol  : TLSv1.2
>     Cipher    : AES256-GCM-SHA384
>     Session-ID: 
> 99DD9750EDDD173E6E41606FB834F0F5AA4B27AA0CCF8284F8D87F47E607D9A9
>     Session-ID-ctx:
>     Master-Key: 
> D1F6C0AD00EB7151098BF0DD68670DE9D4631ACED00CE97EAD684B2670BDE283D34FC85CF7D6CED82FB79C68A150988A
>     Key-Arg   : None
>     Krb5 Principal: None
>     PSK identity: None
>     PSK identity hint: None
>     TLS session ticket lifetime hint: 300 (seconds)
>     TLS session ticket:
>     0000 - 9a 90 ee 94 ba 4d da e1-d7 c9 6d f1 bb 86 0b 74  
>  .....M....m....t
>     0010 - 53 d3 62 eb ca 6b 3e 2b-c4 36 f4 34 ff 73 e0 6a  
>  S.b..k>+.6.4.s.j
>     0020 - 79 f7 72 d7 ca 24 fa 60-bb 37 c8 b9 cd df 71 74  
>  y.r..$.`.7....qt
>     0030 - 00 d8 37 c6 a2 ef dc 49-08 15 36 04 45 58 f5 af  
>  ..7....I..6.EX..
>     0040 - 0c 09 66 36 98 34 6f d0-6d cb 4a 6e 9e 2a 67 d1  
>  ..f6.4o.m.Jn.*g.
>     0050 - b2 84 a1 f2 ff 6f 7a 89-6f 92 5f ca 8b 6a 96 d1  
>  .....oz.o._..j..
>     0060 - 7a 18 f4 b8 50 8e 31 d1-d0 9f 52 d0 01 43 ba eb  
>  z...P.1...R..C..
>     0070 - 6b 89 bb 9e 7c 60 dd 16-ce 2e 14 c4 44 ca 32 74  
>  k...|`......D.2t
>     0080 - da 66 fc 17 ac a3 04 29-3d f6 b8 39 c4 c2 48 81  
>  .f.....)=..9..H.
>     0090 - 75 a1 2e 93 bc 2d 23 c5-5d 35 1b 88 1e 75 97 ee  
>  u....-#.]5...u..
>
>     Start Time: 1546611916
>     Timeout   : 300 (sec)
>     Verify return code: 18 (self signed certificate)
> ---
> read:errno=0
>
>
> ><>
> nathan stratton
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com

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From ahmedmunir007 at gmail.com  Tue Jan  8 18:07:23 2019
From: ahmedmunir007 at gmail.com (Ahmed Chohan)
Date: Tue, 8 Jan 2019 13:07:23 -0500
Subject: [Freeswitch-users] Max participants join Conference bridge
Message-ID: <CAGMN=JcO0-neNbqMDs1w1MYB5XYKHrqqHH8TmNV8_DSh8bL_=A@mail.gmail.com>

Hi,

I'm currently exploring varity of products in market that support
conference service, I would like to how many participants can join each
conference in freeswitch?

Please advise.

-- 
Regards,

Ahmed Munir Chohan
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From bilaln018 at gmail.com  Wed Jan  9 07:36:32 2019
From: bilaln018 at gmail.com (Bilal Abbasi)
Date: Wed, 9 Jan 2019 12:36:32 +0500
Subject: [Freeswitch-users] [mod avmd][1.6.X vs 1.8.X]
Message-ID: <CADSVerAAcMRxVmtA+DuwwC+YvOzdLT8wFubzkwrr_yKxaLv8_Q@mail.gmail.com>

Hi Users,
I was using mod_avmd in FreeSWITCH version 1.6.20, it was working
really well, but now as i shifted to new version i noticed that VM
detection number get dropped.
While investigating i noticed that the default parameters in 1.8.2
avmd.xml.conf are unable to detect the attached beep sound while 1.6.20 was
able to detect that, i tried to change certain parameters in the
avmd.xml.conf file but no luck.
I was wondering if somebody has used avmd on 1.8.X and have the correct
parameters/values OR can somebody point me the right direction to detect
the attached file.

Regards
Bilal Abbasi
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From gmaruzz at gmail.com  Wed Jan  9 13:54:48 2019
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Wed, 9 Jan 2019 14:54:48 +0100
Subject: [Freeswitch-users] Max participants join Conference bridge
In-Reply-To: <CAGMN=JcO0-neNbqMDs1w1MYB5XYKHrqqHH8TmNV8_DSh8bL_=A@mail.gmail.com>
References: <CAGMN=JcO0-neNbqMDs1w1MYB5XYKHrqqHH8TmNV8_DSh8bL_=A@mail.gmail.com>
Message-ID: <CALXCt0rNHBMp=0fmwDZkj1Zquzk2sworkC_ure1zZa937D=d9w@mail.gmail.com>

On Wed, Jan 9, 2019 at 1:46 PM Ahmed Chohan <ahmedmunir007 at gmail.com> wrote:

> Hi
> I'm currently exploring varity of products in market that support
> conference service, I would like to how many participants can join each
> conference in freeswitch?
>
>
FreeSWITCH is open source, so you can connect how many calls into a
conference, no need for licenses.

As per cpu load, it depends on what are you doing. Eg, will you transcode
or not? Audio or video too?

No transcoding is lightest.

If you need professional help/evaluation for your use case, look at the
bottom of the mail, there is the mail address.

-giovanni





> Please advise.
>
> --
> Regards,
>
> Ahmed Munir Chohan
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com



-- 
Sincerely,

Giovanni Maruzzelli
OpenTelecom.IT
cell: +39 347 266 56 18
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From mike at freeswitch.com  Wed Jan  9 16:54:52 2019
From: mike at freeswitch.com (Mike Jerris)
Date: Wed, 9 Jan 2019 11:54:52 -0500
Subject: [Freeswitch-users] [mod avmd][1.6.X vs 1.8.X]
In-Reply-To: <CADSVerAAcMRxVmtA+DuwwC+YvOzdLT8wFubzkwrr_yKxaLv8_Q@mail.gmail.com>
References: <CADSVerAAcMRxVmtA+DuwwC+YvOzdLT8wFubzkwrr_yKxaLv8_Q@mail.gmail.com>
Message-ID: <CADrq_yFuerPcf9CXiKL5txQ+5GOxzRB-ikN1g0YAiqExF0OGdQ@mail.gmail.com>

file a jira on this.  We have a test framework we can add this tone to and
see what needs to be tweaked

On Wed, Jan 9, 2019 at 7:46 AM Bilal Abbasi <bilaln018 at gmail.com> wrote:

> Hi Users,
> I was using mod_avmd in FreeSWITCH version 1.6.20, it was working
> really well, but now as i shifted to new version i noticed that VM
> detection number get dropped.
> While investigating i noticed that the default parameters in 1.8.2
> avmd.xml.conf are unable to detect the attached beep sound while 1.6.20 was
> able to detect that, i tried to change certain parameters in the
> avmd.xml.conf file but no luck.
> I was wondering if somebody has used avmd on 1.8.X and have the correct
> parameters/values OR can somebody point me the right direction to detect
> the attached file.
>
> Regards
> Bilal Abbasi
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From bilaln018 at gmail.com  Thu Jan 10 06:47:10 2019
From: bilaln018 at gmail.com (Bilal Abbasi)
Date: Thu, 10 Jan 2019 11:47:10 +0500
Subject: [Freeswitch-users] [mod avmd][1.6.X vs 1.8.X]
In-Reply-To: <CADrq_yFuerPcf9CXiKL5txQ+5GOxzRB-ikN1g0YAiqExF0OGdQ@mail.gmail.com>
References: <CADSVerAAcMRxVmtA+DuwwC+YvOzdLT8wFubzkwrr_yKxaLv8_Q@mail.gmail.com>
 <CADrq_yFuerPcf9CXiKL5txQ+5GOxzRB-ikN1g0YAiqExF0OGdQ@mail.gmail.com>
Message-ID: <CADSVerApoXGMKnXM3uhEg2fTRUCAxqmn3+npZvsuxMVgxpxt-A@mail.gmail.com>

Done

On Wed, Jan 9, 2019 at 11:09 PM Mike Jerris <mike at freeswitch.com> wrote:

> file a jira on this.  We have a test framework we can add this tone to and
> see what needs to be tweaked
>
> On Wed, Jan 9, 2019 at 7:46 AM Bilal Abbasi <bilaln018 at gmail.com> wrote:
>
>> Hi Users,
>> I was using mod_avmd in FreeSWITCH version 1.6.20, it was working
>> really well, but now as i shifted to new version i noticed that VM
>> detection number get dropped.
>> While investigating i noticed that the default parameters in 1.8.2
>> avmd.xml.conf are unable to detect the attached beep sound while 1.6.20 was
>> able to detect that, i tried to change certain parameters in the
>> avmd.xml.conf file but no luck.
>> I was wondering if somebody has used avmd on 1.8.X and have the correct
>> parameters/values OR can somebody point me the right direction to detect
>> the attached file.
>>
>> Regards
>> Bilal Abbasi
>> _________________________________________________________________________
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From steveayre at gmail.com  Thu Jan 10 16:40:24 2019
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 10 Jan 2019 16:40:24 +0000
Subject: [Freeswitch-users] Max participants join Conference bridge
In-Reply-To: <CALXCt0rNHBMp=0fmwDZkj1Zquzk2sworkC_ure1zZa937D=d9w@mail.gmail.com>
References: <CAGMN=JcO0-neNbqMDs1w1MYB5XYKHrqqHH8TmNV8_DSh8bL_=A@mail.gmail.com>
 <CALXCt0rNHBMp=0fmwDZkj1Zquzk2sworkC_ure1zZa937D=d9w@mail.gmail.com>
Message-ID: <CAFiqYumQ2pB_yrhCmdEWbxT__Mi9KV6-mHnuvfSGQ9n72tLYwQ@mail.gmail.com>

Conferences will always require transcoding.

On Wed, 9 Jan 2019 at 19:03, Giovanni Maruzzelli <gmaruzz at gmail.com> wrote:

> On Wed, Jan 9, 2019 at 1:46 PM Ahmed Chohan <ahmedmunir007 at gmail.com>
> wrote:
>
>> Hi
>> I'm currently exploring varity of products in market that support
>> conference service, I would like to how many participants can join each
>> conference in freeswitch?
>>
>>
> FreeSWITCH is open source, so you can connect how many calls into a
> conference, no need for licenses.
>
> As per cpu load, it depends on what are you doing. Eg, will you transcode
> or not? Audio or video too?
>
> No transcoding is lightest.
>
> If you need professional help/evaluation for your use case, look at the
> bottom of the mail, there is the mail address.
>
> -giovanni
>
>
>
>
>
>> Please advise.
>>
>> --
>> Regards,
>>
>> Ahmed Munir Chohan
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
>
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> OpenTelecom.IT
> cell: +39 347 266 56 18
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From ch.chhatra at gmail.com  Fri Jan 11 17:09:54 2019
From: ch.chhatra at gmail.com (Chhorm Chhatra)
Date: Sat, 12 Jan 2019 00:09:54 +0700
Subject: [Freeswitch-users] Transfer call to Call Group cause LOSE_RACE
Message-ID: <CAJssr-g7kVNna2YvDCXri--pc8zKNx-rt4asBd+E=q2=E3CYjw@mail.gmail.com>

Hello,
Currently, I faced a problem that when I transfer a call to call group, it
causes *Originate Failed.  Cause: LOSE_RACE.*
My setup is that I export bind digit action on the Leg B so that he can
transfer Leg A to another extension.
> Transfer Configuration:
<action application="bind_digit_action"
data="my_transfer,~069\d{3},exec:execute_extension,digit_transfer
XML default,peer,peer"/>
<action application="set" data=
"bridge_pre_execute_bleg_app=bind_digit_action"/>
<action application="set" data="bridge_pre_execute_bleg_data=my_transfer"/>

> Digit_transfer Dialplan Configuration:
<extension name="Display digits dialled AND TRANSFER">
<condition field="destination_number" expression="^digit_transfer$">
    <action application="log" data="INFO Called dialled
${last_matching_digits}"/>
      <action application="log" data="INFO en number iterated
${last_matching_digits}"/>
      <action application="transfer" data="-bleg ${last_matching_digits}
XML default"/>
   </condition>
</extension>

>From this part, the digit_transfer extension will transfer based on
last_matching_digits to the target extension bound by Leg B.
Note: normal call to group call can be bridged successfully.
Any help would be really appreaciated.

Best Regards,
Chhorm Chhatra
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From gmaruzz at gmail.com  Fri Jan 11 17:26:14 2019
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Fri, 11 Jan 2019 18:26:14 +0100
Subject: [Freeswitch-users] Max participants join Conference bridge
In-Reply-To: <CAFiqYumQ2pB_yrhCmdEWbxT__Mi9KV6-mHnuvfSGQ9n72tLYwQ@mail.gmail.com>
References: <CAGMN=JcO0-neNbqMDs1w1MYB5XYKHrqqHH8TmNV8_DSh8bL_=A@mail.gmail.com>
 <CALXCt0rNHBMp=0fmwDZkj1Zquzk2sworkC_ure1zZa937D=d9w@mail.gmail.com>
 <CAFiqYumQ2pB_yrhCmdEWbxT__Mi9KV6-mHnuvfSGQ9n72tLYwQ@mail.gmail.com>
Message-ID: <CALXCt0o1ugpavTMs2VsNZSuDsMCu_CH3_uMon0G++ByWWecxcw@mail.gmail.com>

On Fri, Jan 11, 2019, 17:14 Steven Ayre <steveayre at gmail.com wrote:

> Conferences will always require transcoding.
>

Yep, kind of.

IIRC.
Audio from the speaker(s) to a conference will always be transformed into
the common internal format (signed linear) to be mixed togheter (if more
than one speaker surpass the audio energy threshold), and then the result
of the speakers mix will be sent to each listening participant, transcoded
to each participant own codec.
So, if all participants have same codec, only one linear to codec is done
and the result is distributed.

Also, not sure if with one only speaker over the energy threshold, and
listeners with same codec as speaker (eg, no mix needed), maybe the speaker
is directly distributed without even transcoding to linear?
Can't remember...

Anyway, transcoding audio il not so heavy.

Transcoding video, or just mixing video, is very heavy. If you want video
conf without load, use pass through mode instead of mux. This way
(conference passthroufh mode, as opposed to.mux mode) it will just
distribute the "video floor" video stream (eg, the speaker stream) to all
participants, without noticeable load.

-giovanni


> On Wed, 9 Jan 2019 at 19:03, Giovanni Maruzzelli <gmaruzz at gmail.com>
> wrote:
>
>> On Wed, Jan 9, 2019 at 1:46 PM Ahmed Chohan <ahmedmunir007 at gmail.com>
>> wrote:
>>
>>> Hi
>>> I'm currently exploring varity of products in market that support
>>> conference service, I would like to how many participants can join each
>>> conference in freeswitch?
>>>
>>>
>> FreeSWITCH is open source, so you can connect how many calls into a
>> conference, no need for licenses.
>>
>> As per cpu load, it depends on what are you doing. Eg, will you transcode
>> or not? Audio or video too?
>>
>> No transcoding is lightest.
>>
>> If you need professional help/evaluation for your use case, look at the
>> bottom of the mail, there is the mail address.
>>
>> -giovanni
>>
>>
>>
>>
>>
>>> Please advise.
>>>
>>> --
>>> Regards,
>>>
>>> Ahmed Munir Chohan
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Services
>>> sales at freeswitch.com
>>> https://freeswitch.com
>>>
>>> Official FreeSWITCH Sites
>>> https://freeswitch.com/oss
>>> https://freeswitch.org/confluence
>>> https://cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> https://freeswitch.com
>>
>>
>>
>> --
>> Sincerely,
>>
>> Giovanni Maruzzelli
>> OpenTelecom.IT
>> cell: +39 347 266 56 18
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From ch.chhatra at gmail.com  Sun Jan 13 05:56:28 2019
From: ch.chhatra at gmail.com (Chhorm Chhatra)
Date: Sun, 13 Jan 2019 12:56:28 +0700
Subject: [Freeswitch-users] MOD_CALLCENTER become unload and unknown while
	freeswitch is still running
Message-ID: <CAJssr-iSR5v3_3ctqSTScnBOKe4GXyTxME8s2EDH80jtrvvSHg@mail.gmail.com>

Hi,
Currently, I faced a very strange issue that
1) After several days of FreeSWITCH running, I could not connect to its CLI
at all. But Freeswitch is still working as it is. I have to tail the log
file to debug this issue.
2) When the issue (1) happens, the application *callcenter *is unknown and
therefore the callcenter failed.
FS version: 1.6
Any help would be appreciated.
Thank you in advance.
Best regards,
Chhorm Chhatra
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From mickael at winlux.fr  Mon Jan 14 16:40:55 2019
From: mickael at winlux.fr (Mickael Hubert)
Date: Mon, 14 Jan 2019 17:40:55 +0100
Subject: [Freeswitch-users] Basic Nagios script for Freeswitch
Message-ID: <CA+fRVSWPrDg84-mr1FUZEqUVCTUxx5QV-XrHbtRU9mTvJDX6Ww@mail.gmail.com>

Hi all,
today I developed a basic nagios script to check freeswitch instance.
I publish it in github, maybe it can help you.

https://github.com/Mickaelh51/pynagfreeswitch

Ex:

pynagfreeswitch.py -t -w 10 -c 20 ==> OK - Total calls count is 0
pynagfreeswitch.py -g gateway1 -G failedcallsout -w 10 -c 20 ==> OK -
gateway1 failedcallsout is 2
pynagfreeswitch.py -g GWSCR1 -G pingtime -w 0.20 -c 0.56 ==> WARNING -
GWSCR1 pingtime is 0.55
pynagfreeswitch.py -g GWSCR1 -G status -S up ==> OK - GWSCR1 status is up
pynagfreeswitch.py -g GWSCR1 -G state -S noreg ==> OK - GWSCR1 state is noreg

./pynagfreeswitch.py -h
usage: pynagfreeswitch.py [-h] [-a AUTH] [-s SERVER] [-p PORT] [-w WARNING]
                          [-c CRITICAL] [-g GATEWAY] [-G GATEWAYCHECK] [-t]
                          [-S STRINGOK]

optional arguments:
  -h, --help            show this help message and exit
  -a AUTH, --auth AUTH  ESL password
  -s SERVER, --server SERVER
                        FreeSWITCH server IP address
  -p PORT, --port PORT  FreeSWITCH server event socket port
  -w WARNING, --warning WARNING
                        threshold that generates a Nagios warning
  -c CRITICAL, --critical CRITICAL
                        threshold that generates a Nagios critical warning
  -g GATEWAY, --gateway GATEWAY
                        select gateway
  -G GATEWAYCHECK, --gatewaycheck GATEWAYCHECK
                        type of checking (failedcallsout / status / state /
...)
  -t, --totalcallscount
                        total calls count
  -S STRINGOK, --stringok STRINGOK
                        if this string is found = OK

++
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From jprangi at gmail.com  Mon Jan 14 18:28:19 2019
From: jprangi at gmail.com (Jai Rangi)
Date: Mon, 14 Jan 2019 10:28:19 -0800
Subject: [Freeswitch-users] Call Quality with Recording
Message-ID: <CAPfjz3gUfEfNXXg07XJwVxmhOdobgh3dUwOAbQTBzDObyM78DQ@mail.gmail.com>

We experience poor call quality when recording is on. We are using
record_session to do call recording.

session:execute("record_session","$${recordings_dir}/${uuid}.mp3")

Any one else experiencing similar issue? Any comment on recording wav vs
mp3.

Thank you,
Jai
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From jungleboogie0 at gmail.com  Mon Jan 14 23:24:39 2019
From: jungleboogie0 at gmail.com (Jungle Boogie)
Date: Mon, 14 Jan 2019 15:24:39 -0800
Subject: [Freeswitch-users] ESL help on remote machines
Message-ID: <20190114232439.GA69791@puffer.in.lylie.net>

Hi All,

I know I'm a little late to the ESL party, but I have a use for it and I want to
try it out.

As the inbound diagram depicts, I would like a perl script on a remote machine
to trigger a simple lua script on the FreeSWITCH machine that generates an outbound call.

https://freeswitch.org/confluence/display/FREESWITCH/mod_event_socket

-----------
Machine A -
perl esl
-----------
    |
	|
    \/
----------
Freeswitch -
lua originate
----------
	|
	|
	\/
----------
phone call
---------

The perl esl page states the module must be compiled. Does this mean machine
A needs to have the entire FS source code to compile the module?
https://freeswitch.org/confluence/display/FREESWITCH/Perl+ESL

If the entire src isn't needed, what's it take to install the ESL module
successfully on machine A?

Thanks,
sean


From alex at freeswitch.com  Mon Jan 14 23:44:24 2019
From: alex at freeswitch.com (Alexey Sibyakin)
Date: Tue, 15 Jan 2019 08:44:24 +0900
Subject: [Freeswitch-users] MOD_CALLCENTER become unload and unknown
 while freeswitch is still running
In-Reply-To: <CAJssr-iSR5v3_3ctqSTScnBOKe4GXyTxME8s2EDH80jtrvvSHg@mail.gmail.com>
References: <CAJssr-iSR5v3_3ctqSTScnBOKe4GXyTxME8s2EDH80jtrvvSHg@mail.gmail.com>
Message-ID: <CAGJLwFi2oTV5XeF7HzhLPwq7nBdbfhDHbEYJJ4Mf+wxh8NTfbQ@mail.gmail.com>

Hi,

Try to update to 1.8 first. FreeSWITCH 1.6 is EOL.

Regards,

Alex

Alex Sibyakin | Support Engineer SignalWire | 228 Hamilton Ave 3rd Floor,
Palo Alto, CA 94303 Email: alex at freeswitch.com Website:
https://www.signalwire.com



On Tue, Jan 15, 2019 at 2:07 AM Chhorm Chhatra <ch.chhatra at gmail.com> wrote:

> Hi,
> Currently, I faced a very strange issue that
> 1) After several days of FreeSWITCH running, I could not connect to its
> CLI at all. But Freeswitch is still working as it is. I have to tail the
> log file to debug this issue.
> 2) When the issue (1) happens, the application *callcenter *is unknown
> and therefore the callcenter failed.
> FS version: 1.6
> Any help would be appreciated.
> Thank you in advance.
> Best regards,
> Chhorm Chhatra
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From gmaruzz at gmail.com  Tue Jan 15 09:11:36 2019
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Tue, 15 Jan 2019 10:11:36 +0100
Subject: [Freeswitch-users] ESL help on remote machines
In-Reply-To: <20190114232439.GA69791@puffer.in.lylie.net>
References: <20190114232439.GA69791@puffer.in.lylie.net>
Message-ID: <CALXCt0oXf_fk7GAxfyp8s=1BHEvYXMiT-+ONVQxw6vSic1fUXg@mail.gmail.com>

On Tue, Jan 15, 2019 at 1:54 AM Jungle Boogie <jungleboogie0 at gmail.com>
wrote:

> The perl esl page states the module must be compiled. Does this mean
> machine
> A needs to have the entire FS source code to compile the module?
> https://freeswitch.org/confluence/display/FREESWITCH/Perl+ESL
>
> If the entire src isn't needed, what's it take to install the ESL module
> successfully on machine A?
>
>
I suppose you installed FreeSWITCH from packages, correct?
A search in installable packages gives you the following:

# apt-cache search freeswitch | grep -i perl
freeswitch-mod-perl - mod_perl for FreeSWITCH
freeswitch-mod-perl-dbg - mod_perl for FreeSWITCH (debug)
libesl-perl - Cross-Platform Scalable Multi-Protocol Soft Switch

I would install libesl-perl :

# apt-get install libesl-perl

-giovanni








> Thanks,
> sean
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com



-- 
Sincerely,

Giovanni Maruzzelli
OpenTelecom.IT
cell: +39 347 266 56 18
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From david.villasmil.work at gmail.com  Tue Jan 15 12:44:31 2019
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Tue, 15 Jan 2019 12:44:31 +0000
Subject: [Freeswitch-users] ESL help on remote machines
In-Reply-To: <20190114232439.GA69791@puffer.in.lylie.net>
References: <20190114232439.GA69791@puffer.in.lylie.net>
Message-ID: <CAFGRPVq=SWe2f4COzwjwj73iZDVn3X8WN3qWas8-iwTwe01xDA@mail.gmail.com>

You can simply connect to the esl socket (port 8021) on fs from Box A. No
need for that lua script.
You simply need to configure the ESL socket to listen on (I.e.: 0.0.0.0)
and iptables it to only allow your box A.
As far as the library is concerned, in my experience, yes. The simplest is
to have the source, so You need the whole fs source and compile. You can
afterwards remove it.

Hope that helps.

If all you need is to originate (meaning you don’t care about events
afterwards) there is another way of originating, and that’s via the httapi:
https://freeswitch.org/confluence/plugins/servlet/mobile#content/view/3966423

With httapi you can all any api in fs.

Hope that helps

On Tue, 15 Jan 2019 at 00:50, Jungle Boogie <jungleboogie0 at gmail.com> wrote:

> Hi All,
>
> I know I'm a little late to the ESL party, but I have a use for it and I
> want to
> try it out.
>
> As the inbound diagram depicts, I would like a perl script on a remote
> machine
> to trigger a simple lua script on the FreeSWITCH machine that generates an
> outbound call.
>
> https://freeswitch.org/confluence/display/FREESWITCH/mod_event_socket
>
> -----------
> Machine A -
> perl esl
> -----------
>     |
>         |
>     \/
> ----------
> Freeswitch -
> lua originate
> ----------
>         |
>         |
>         \/
> ----------
> phone call
> ---------
>
> The perl esl page states the module must be compiled. Does this mean
> machine
> A needs to have the entire FS source code to compile the module?
> https://freeswitch.org/confluence/display/FREESWITCH/Perl+ESL
>
> If the entire src isn't needed, what's it take to install the ESL module
> successfully on machine A?
>
> Thanks,
> sean
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From surender.s at hcl.com  Tue Jan 15 12:51:10 2019
From: surender.s at hcl.com (Surender Singh)
Date: Tue, 15 Jan 2019 12:51:10 +0000
Subject: [Freeswitch-users] How to enable the tls avaya J series phones
Message-ID: <SG2PR0401MB1967F87F708CFB12EEA1FA3283810@SG2PR0401MB1967.apcprd04.prod.outlook.com>

Hi Team,

I am new for freeswitch and I installed the FS on my window7 machine . I want to register the avaya j100 series phone with FS with TLS .

I have one CA.pem certificate which I generated through SSL tool or I need to generate the CA certificate using FS only.

Can you please guide or share the steps to enable the TLS with FS for window based FS ?

Regads
Surender Singh
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From aqsyounas at gmail.com  Tue Jan 15 20:49:37 2019
From: aqsyounas at gmail.com (Aqs Younas)
Date: Wed, 16 Jan 2019 01:49:37 +0500
Subject: [Freeswitch-users] Play the sound file to caller from the same
 location where it had left in previous call
Message-ID: <CAGVOjVWAV7XKJx=kBwHW2EQ8ixfK33PUs_22xyH+tfjazpdTPw@mail.gmail.com>

Greetings,

I would like to play the sound file to the caller at the same location
where he had left on the previous call. How I see is that, save call
duration upon call end into database and check it again upon new call
against DID and sleek the file to that point.

I have seen a post from one guy on the same topic but could not able to
find this.

Any better thoughts?

Best Regards,

Aqs Younas
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From grcamauer at gmail.com  Tue Jan 15 23:23:04 2019
From: grcamauer at gmail.com (Guillermo Ruiz Camauer)
Date: Tue, 15 Jan 2019 20:23:04 -0300
Subject: [Freeswitch-users] Call Quality with Recording
In-Reply-To: <CAPfjz3gUfEfNXXg07XJwVxmhOdobgh3dUwOAbQTBzDObyM78DQ@mail.gmail.com>
References: <CAPfjz3gUfEfNXXg07XJwVxmhOdobgh3dUwOAbQTBzDObyM78DQ@mail.gmail.com>
Message-ID: <CAPNTnnrOgrponux73vAteLSyVHgMLJaW_rCBB_x6h-UqyN+P0A@mail.gmail.com>

How many sessions are you recording simultaneously?  Does it occuer when
the machine has no load?

Guillermo

On Mon, Jan 14, 2019 at 7:48 PM Jai Rangi <jprangi at gmail.com> wrote:

> We experience poor call quality when recording is on. We are using
> record_session to do call recording.
>
> session:execute("record_session","$${recordings_dir}/${uuid}.mp3")
>
> Any one else experiencing similar issue? Any comment on recording wav vs
> mp3.
>
> Thank you,
> Jai
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com



-- 
Guillermo Ruiz Camauer
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From brian at freeswitch.com  Wed Jan 16 00:44:30 2019
From: brian at freeswitch.com (Brian West)
Date: Tue, 15 Jan 2019 18:44:30 -0600
Subject: [Freeswitch-users] Call Quality with Recording
In-Reply-To: <CAPfjz3gUfEfNXXg07XJwVxmhOdobgh3dUwOAbQTBzDObyM78DQ@mail.gmail.com>
References: <CAPfjz3gUfEfNXXg07XJwVxmhOdobgh3dUwOAbQTBzDObyM78DQ@mail.gmail.com>
Message-ID: <CAEJMVkBP4YTOC5YmcNegGk3fSkZNhUwSe-yO5mNqyNsSg5GPBQ@mail.gmail.com>

What revision are you using?

On Mon, Jan 14, 2019 at 4:21 PM Jai Rangi <jprangi at gmail.com> wrote:

> We experience poor call quality when recording is on. We are using
> record_session to do call recording.
>
> session:execute("record_session","$${recordings_dir}/${uuid}.mp3")
>
> Any one else experiencing similar issue? Any comment on recording wav vs
> mp3.
>
> Thank you,
> Jai
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com



-- 

Brian West | Co-founder and Developer

Need Commercial support? email sales at freeswitch.com

FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045
<https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g>

Email: brian at freeswitch.com

Mobile: 918-424-9378

Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/>

[image: https://www.facebook.com/signalwireinc?src=email]
<https://www.facebook.com/freeswitch> [image:
https://twitter.com/freeswitch] <https://twitter.com/freeswitch>
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From jprangi at gmail.com  Wed Jan 16 00:49:24 2019
From: jprangi at gmail.com (Jai Rangi)
Date: Tue, 15 Jan 2019 16:49:24 -0800
Subject: [Freeswitch-users] Call Quality with Recording
In-Reply-To: <CAPNTnnrOgrponux73vAteLSyVHgMLJaW_rCBB_x6h-UqyN+P0A@mail.gmail.com>
References: <CAPfjz3gUfEfNXXg07XJwVxmhOdobgh3dUwOAbQTBzDObyM78DQ@mail.gmail.com>
 <CAPNTnnrOgrponux73vAteLSyVHgMLJaW_rCBB_x6h-UqyN+P0A@mail.gmail.com>
Message-ID: <CAPfjz3ijXB+upp+WanupzVq0cYMkzhwNE1GGQsfFLTVn++Ej4A@mail.gmail.com>

Not many, 60 to 70.

On Tue, Jan 15, 2019 at 4:25 PM Guillermo Ruiz Camauer <grcamauer at gmail.com>
wrote:

> How many sessions are you recording simultaneously?  Does it occuer when
> the machine has no load?
>
> Guillermo
>
> On Mon, Jan 14, 2019 at 7:48 PM Jai Rangi <jprangi at gmail.com> wrote:
>
>> We experience poor call quality when recording is on. We are using
>> record_session to do call recording.
>>
>> session:execute("record_session","$${recordings_dir}/${uuid}.mp3")
>>
>> Any one else experiencing similar issue? Any comment on recording wav vs
>> mp3.
>>
>> Thank you,
>> Jai
>> _________________________________________________________________________
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
>
>
> --
> Guillermo Ruiz Camauer
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From jprangi at didforsale.com  Wed Jan 16 00:50:28 2019
From: jprangi at didforsale.com (Jai Rangi)
Date: Tue, 15 Jan 2019 16:50:28 -0800
Subject: [Freeswitch-users] Call Quality with Recording
In-Reply-To: <CAEJMVkBP4YTOC5YmcNegGk3fSkZNhUwSe-yO5mNqyNsSg5GPBQ@mail.gmail.com>
References: <CAPfjz3gUfEfNXXg07XJwVxmhOdobgh3dUwOAbQTBzDObyM78DQ@mail.gmail.com>
 <CAEJMVkBP4YTOC5YmcNegGk3fSkZNhUwSe-yO5mNqyNsSg5GPBQ@mail.gmail.com>
Message-ID: <CAPfjz3jSABS+0-0GgEs7=LN5EaTXxBefJ7sSYJgU-P0sU=rPVQ@mail.gmail.com>

FreeSWITCH (Version 1.6.16 git 3da6bd0 2017-04-07 16:49:13Z 64bit)

*Jai Rangi*
Cebod Technologies LLC dba DIDforSale/Cebod Telecom
O 949-471-0102 | C 949-419-7634 <1-949-419-7634> | F 949-269-0449 /
949-232-1410 | jprangi at didforsale.com  www.cebod.com |
www.didforsale.com |2372
Morse Ave., Ste. 124, Irvine, CA 92614|







On Tue, Jan 15, 2019 at 4:46 PM Brian West <brian at freeswitch.com> wrote:

> What revision are you using?
>
> On Mon, Jan 14, 2019 at 4:21 PM Jai Rangi <jprangi at gmail.com> wrote:
>
>> We experience poor call quality when recording is on. We are using
>> record_session to do call recording.
>>
>> session:execute("record_session","$${recordings_dir}/${uuid}.mp3")
>>
>> Any one else experiencing similar issue? Any comment on recording wav vs
>> mp3.
>>
>> Thank you,
>> Jai
>> _________________________________________________________________________
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
>
>
> --
>
> Brian West | Co-founder and Developer
>
> Need Commercial support? email sales at freeswitch.com
>
> FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045
> <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g>
>
> Email: brian at freeswitch.com
>
> Mobile: 918-424-9378
>
> Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/>
>
> [image: https://www.facebook.com/signalwireinc?src=email]
> <https://www.facebook.com/freeswitch> [image:
> https://twitter.com/freeswitch] <https://twitter.com/freeswitch>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From moshe.rosenberg at gmail.com  Wed Jan 16 05:43:00 2019
From: moshe.rosenberg at gmail.com (Moshe Rosenberg)
Date: Wed, 16 Jan 2019 00:43:00 -0500
Subject: [Freeswitch-users] Screen pop application
Message-ID: <CAL1=tORT3_kQrgcO-=5ygAu+hQ0B77EoLRNUqVVCfj9i8OtK1g@mail.gmail.com>

Hi

I am looking for a windows based application that will pop up a web browser
with the url + caller I’d each time the phone rings
-- 
Moshe Rosenberg
Tel. 718 633 1444
Cel. 347 678 3993
www.data <http://www.dataday.consulting/>phone.cloud
Moshe at dataphone.cloud
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From infos at madovsky.org  Wed Jan 16 13:46:00 2019
From: infos at madovsky.org (Madovsky)
Date: Wed, 16 Jan 2019 05:46:00 -0800
Subject: [Freeswitch-users] FS and gcc 8.x.x
In-Reply-To: <c39aa5c6-1afb-898a-e86e-a552be1b3a1b@madovsky.org>
References: <aeed5aef-64b9-6cf2-f362-70d8409c2426@madovsky.org>
 <CAEJMVkCqVxSTDni83QjQcuVOwZY5ipFKaiQ7-BnFGkU32t_hcg@mail.gmail.com>
 <c39aa5c6-1afb-898a-e86e-a552be1b3a1b@madovsky.org>
Message-ID: <6b0f3278-c67f-040f-9561-b5ece7f2ead3@madovsky.org>

for those who will encounter the same compiling issue on Fedora 28 64bits

just use

./configure CFLAGS="-D__alloca=alloca" .....

and everything is ok

On 12/11/2018 5:35 AM, Madovsky wrote:
>
> https://freeswitch.org/jira/browse/FS-11170
>
> thanks
>
> On 12/10/2018 6:02 AM, Brian West wrote:
>> Can you please send a bug report to JIRA?
>> Thanks,
>> /b
>>
>>
>> On Mon, Dec 10, 2018 at 3:09 AM Madovsky <infos at madovsky.org 
>> <mailto:infos at madovsky.org>> wrote:
>>
>>     Just FYI FS does not compile anymore with gcc 8.x.x
>>
>>     major changes have been made with this new version of gcc,
>>     especially
>>     strncpy and strncat.
>>
>>
>>     _________________________________________________________________________
>>     Professional FreeSWITCH Services
>>     sales at freeswitch.com <mailto:sales at freeswitch.com>
>>     https://freeswitch.com
>>
>>     Official FreeSWITCH Sites
>>     https://freeswitch.com/oss
>>     https://freeswitch.org/confluence
>>     https://cluecon.com
>>
>>     FreeSWITCH-users mailing list
>>     FreeSWITCH-users at lists.freeswitch.org
>>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>     https://freeswitch.com
>>
>>
>>
>> -- 
>>
>> Brian West | Co-founder and Developer
>>
>> Need Commercial support? email sales at freeswitch.com 
>> <mailto:sales at freeswitch.com>
>>
>> FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 
>> <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g>
>>
>> Email: brian at freeswitch.com <mailto:brian at freeswitch.com>
>>
>> Mobile: 918-424-9378
>>
>> Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/>
>>
>> https://www.facebook.com/signalwireinc?src=email 
>> <https://www.facebook.com/freeswitch> https://twitter.com/freeswitch 
>> <https://twitter.com/freeswitch>
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>

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From grcamauer at gmail.com  Wed Jan 16 20:25:35 2019
From: grcamauer at gmail.com (Guillermo Ruiz Camauer)
Date: Wed, 16 Jan 2019 17:25:35 -0300
Subject: [Freeswitch-users] Call Quality with Recording
In-Reply-To: <CAPfjz3ijXB+upp+WanupzVq0cYMkzhwNE1GGQsfFLTVn++Ej4A@mail.gmail.com>
References: <CAPfjz3gUfEfNXXg07XJwVxmhOdobgh3dUwOAbQTBzDObyM78DQ@mail.gmail.com>
 <CAPNTnnrOgrponux73vAteLSyVHgMLJaW_rCBB_x6h-UqyN+P0A@mail.gmail.com>
 <CAPfjz3ijXB+upp+WanupzVq0cYMkzhwNE1GGQsfFLTVn++Ej4A@mail.gmail.com>
Message-ID: <CAPNTnnp9=WxDRzvjRHiv0uteG-6PWANMkWyerh8B=Uita-q3Kg@mail.gmail.com>

On what hardware?  Specially, on what type and number of disks?

Guillermo

On Wed, Jan 16, 2019 at 2:00 PM Jai Rangi <jprangi at gmail.com> wrote:

> Not many, 60 to 70.
>
> On Tue, Jan 15, 2019 at 4:25 PM Guillermo Ruiz Camauer <
> grcamauer at gmail.com> wrote:
>
>> How many sessions are you recording simultaneously?  Does it occuer when
>> the machine has no load?
>>
>> Guillermo
>>
>> On Mon, Jan 14, 2019 at 7:48 PM Jai Rangi <jprangi at gmail.com> wrote:
>>
>>> We experience poor call quality when recording is on. We are using
>>> record_session to do call recording.
>>>
>>> session:execute("record_session","$${recordings_dir}/${uuid}.mp3")
>>>
>>> Any one else experiencing similar issue? Any comment on recording wav vs
>>> mp3.
>>>
>>> Thank you,
>>> Jai
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Services
>>> sales at freeswitch.com
>>> https://freeswitch.com
>>>
>>> Official FreeSWITCH Sites
>>> https://freeswitch.com/oss
>>> https://freeswitch.org/confluence
>>> https://cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> https://freeswitch.com
>>
>>
>>
>> --
>> Guillermo Ruiz Camauer
>> _________________________________________________________________________
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com



-- 
Guillermo Ruiz Camauer
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From grcamauer at gmail.com  Wed Jan 16 20:27:07 2019
From: grcamauer at gmail.com (Guillermo Ruiz Camauer)
Date: Wed, 16 Jan 2019 17:27:07 -0300
Subject: [Freeswitch-users] Call Quality with Recording
In-Reply-To: <CAPfjz3ijXB+upp+WanupzVq0cYMkzhwNE1GGQsfFLTVn++Ej4A@mail.gmail.com>
References: <CAPfjz3gUfEfNXXg07XJwVxmhOdobgh3dUwOAbQTBzDObyM78DQ@mail.gmail.com>
 <CAPNTnnrOgrponux73vAteLSyVHgMLJaW_rCBB_x6h-UqyN+P0A@mail.gmail.com>
 <CAPfjz3ijXB+upp+WanupzVq0cYMkzhwNE1GGQsfFLTVn++Ej4A@mail.gmail.com>
Message-ID: <CAPNTnno2e=CoiZyvKo_yzB+vO2Q97erf8rWp6kmwsx4XHB6qeQ@mail.gmail.com>

Does the bad call quality occur when you have less load (10-20 simultaneous
calls)?

On Wed, Jan 16, 2019 at 2:00 PM Jai Rangi <jprangi at gmail.com> wrote:

> Not many, 60 to 70.
>
> On Tue, Jan 15, 2019 at 4:25 PM Guillermo Ruiz Camauer <
> grcamauer at gmail.com> wrote:
>
>> How many sessions are you recording simultaneously?  Does it occuer when
>> the machine has no load?
>>
>> Guillermo
>>
>> On Mon, Jan 14, 2019 at 7:48 PM Jai Rangi <jprangi at gmail.com> wrote:
>>
>>> We experience poor call quality when recording is on. We are using
>>> record_session to do call recording.
>>>
>>> session:execute("record_session","$${recordings_dir}/${uuid}.mp3")
>>>
>>> Any one else experiencing similar issue? Any comment on recording wav vs
>>> mp3.
>>>
>>> Thank you,
>>> Jai
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Services
>>> sales at freeswitch.com
>>> https://freeswitch.com
>>>
>>> Official FreeSWITCH Sites
>>> https://freeswitch.com/oss
>>> https://freeswitch.org/confluence
>>> https://cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> https://freeswitch.com
>>
>>
>>
>> --
>> Guillermo Ruiz Camauer
>> _________________________________________________________________________
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com



-- 
Guillermo Ruiz Camauer
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From jprangi at gmail.com  Wed Jan 16 21:32:36 2019
From: jprangi at gmail.com (Jai Rangi)
Date: Wed, 16 Jan 2019 13:32:36 -0800
Subject: [Freeswitch-users] Call Quality with Recording
In-Reply-To: <CAPNTnnp9=WxDRzvjRHiv0uteG-6PWANMkWyerh8B=Uita-q3Kg@mail.gmail.com>
References: <CAPfjz3gUfEfNXXg07XJwVxmhOdobgh3dUwOAbQTBzDObyM78DQ@mail.gmail.com>
 <CAPNTnnrOgrponux73vAteLSyVHgMLJaW_rCBB_x6h-UqyN+P0A@mail.gmail.com>
 <CAPfjz3ijXB+upp+WanupzVq0cYMkzhwNE1GGQsfFLTVn++Ej4A@mail.gmail.com>
 <CAPNTnnp9=WxDRzvjRHiv0uteG-6PWANMkWyerh8B=Uita-q3Kg@mail.gmail.com>
Message-ID: <CAPfjz3jA-6_MGu-mVBPnw3u_c2_8pMW=-79=t9wgAE651qiqFg@mail.gmail.com>

Dell C6100, CPU and RAM is not the issue. Tried recording on local as well
as NFS. Hard drives are SSDs.
Wanted to add an update here,
One thing we found is that its combination of G729 + Recording, Forcing the
codec to G711 seems to be no issue.

Thank you,

On Wed, Jan 16, 2019 at 12:53 PM Guillermo Ruiz Camauer <grcamauer at gmail.com>
wrote:

> On what hardware?  Specially, on what type and number of disks?
>
> Guillermo
>
> On Wed, Jan 16, 2019 at 2:00 PM Jai Rangi <jprangi at gmail.com> wrote:
>
>> Not many, 60 to 70.
>>
>> On Tue, Jan 15, 2019 at 4:25 PM Guillermo Ruiz Camauer <
>> grcamauer at gmail.com> wrote:
>>
>>> How many sessions are you recording simultaneously?  Does it occuer when
>>> the machine has no load?
>>>
>>> Guillermo
>>>
>>> On Mon, Jan 14, 2019 at 7:48 PM Jai Rangi <jprangi at gmail.com> wrote:
>>>
>>>> We experience poor call quality when recording is on. We are using
>>>> record_session to do call recording.
>>>>
>>>> session:execute("record_session","$${recordings_dir}/${uuid}.mp3")
>>>>
>>>> Any one else experiencing similar issue? Any comment on recording wav
>>>> vs mp3.
>>>>
>>>> Thank you,
>>>> Jai
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Services
>>>> sales at freeswitch.com
>>>> https://freeswitch.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> https://freeswitch.com/oss
>>>> https://freeswitch.org/confluence
>>>> https://cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> https://freeswitch.com
>>>
>>>
>>>
>>> --
>>> Guillermo Ruiz Camauer
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Services
>>> sales at freeswitch.com
>>> https://freeswitch.com
>>>
>>> Official FreeSWITCH Sites
>>> https://freeswitch.com/oss
>>> https://freeswitch.org/confluence
>>> https://cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> https://freeswitch.com
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
>
>
> --
> Guillermo Ruiz Camauer
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From sunil.more64s at gmail.com  Thu Jan 17 07:35:15 2019
From: sunil.more64s at gmail.com (Sunil More)
Date: Thu, 17 Jan 2019 13:05:15 +0530
Subject: [Freeswitch-users] Call Hangup on Freeswitch
Message-ID: <CANWJJq-FwmPnGF4JCGyYrw+F74axVajvsqu7uHFHo7Ey57KNsQ@mail.gmail.com>

Hello All,

I am using the below originate command and the call gets hung-up at-least 2
times in 10 attempts. This happens randomly and I can see on console log
that freeswitch UNBRIDGES the call. The call is originate and bridge , it
fails with Destination out of order. To add the call is successfully
bridged on the PSTN and agent side and then fails quickly. We use bgapi to
fire this command.

originate {ignore_early_media=false,absolute_codec_string=PCMU,media_webrtc=true,origination_uuid=f8702710-1992-11e9-898c-c371816bde1b,hangup_after_bridge=true,park_after_bridge=true}sofia/gateway/6008_groups/86067830
&bridge([absolute_codec_string=PCMU,hangup_after_bridge=true,park_after_bridge=true,origination_uuid=f8704e20-1992-11e9-898c-c371816bde1b,cc_caller_uuid=f8702710-1992-11e9-898c-c371816bde1b,cc_called_number=919503338275]sofia/gateway/6008_groups/16468107050)

Can you please point me in the right direction?

Thanking You,
Sunil More
Ph : 9503338275
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From nandy1925 at gmail.com  Thu Jan 17 09:54:28 2019
From: nandy1925 at gmail.com (Nandy Dagondon)
Date: Thu, 17 Jan 2019 09:54:28 +0000
Subject: [Freeswitch-users] Double NAT w/ same Private Network Address
Message-ID: <CAHKqMb7c9h6KJEPBsREDPY3yT_03UYHaDKE6LKvCLUa58rNBrA@mail.gmail.com>

Hi everybody,

I'm testing mod_verto in a Double NAT  setup.  The problem: no media.  This
is the setup:

FS (192.168.1.x/24) <> NAT <> Internet <> NAT <> WebRTC Client
(192.168.1.x/24)

Both FS and the Client routers have static IP WAN addresses.  I suspect
that since Private Network addresses are the same (192.168.1.x/24),  the
ICE candidate chosen is always 192.168.1.x, thinking that it's in the Local
Network.  Examining the log file indeed showed that the WAN IP's are
included in the candidate. But, the one chosen is always a private IP.

Am I correct?  What needs to be done to ensure that WAN candidates will be
selected?  Changing the Network Address of either side might solve the
problem but I need the solution via FS/Verto configuration.  (Btw, the
local TURN service is not running.)

Appreciate for your help.

/Nandy
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From julien.terrasson at gmail.com  Thu Jan 17 14:54:25 2019
From: julien.terrasson at gmail.com (Julien Terrasson)
Date: Thu, 17 Jan 2019 15:54:25 +0100
Subject: [Freeswitch-users] when using goup_confirm_key=exec,
	the ringback tone is not heard.
Message-ID: <CAFi=dz-6Um_Y+cYrZGBAjCraQ=7k9+jZngB3s=Kj2UuCHCYmKQ@mail.gmail.com>

Hi everyone,

I'm working with a dialplan where i need the called party confirmation
before establishing the call.

So i'm using group_confirm_key=exec (because, not only i need to play a
file to the caller, but also to the called party, so a little bit of lua
scripting is needed).
When doing this, the ringback tone is not heard anymore by the caller (when
180 Rining is received, nothing happen).

This doesn't happen with then standard usage of group_confirm_key and
group_confirm_file, for example group_confirm_key=#.
group_confirm_file=hello.wav

Any clue on how the RBT can be heard ?

Best regards,

J.Terrasson
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From brian at freeswitch.com  Thu Jan 17 20:08:59 2019
From: brian at freeswitch.com (Brian West)
Date: Thu, 17 Jan 2019 14:08:59 -0600
Subject: [Freeswitch-users] Call Hangup on Freeswitch
In-Reply-To: <CANWJJq-FwmPnGF4JCGyYrw+F74axVajvsqu7uHFHo7Ey57KNsQ@mail.gmail.com>
References: <CANWJJq-FwmPnGF4JCGyYrw+F74axVajvsqu7uHFHo7Ey57KNsQ@mail.gmail.com>
Message-ID: <CAEJMVkDrafo6Qr8qafguPuPc=zXdU56O0CgB28ZzGt4zm_c2aw@mail.gmail.com>

Where did you get the info/idea to do ignore_early_media=false? You don't
have to do that.

On Thu, Jan 17, 2019 at 9:45 AM Sunil More <sunil.more64s at gmail.com> wrote:

> Hello All,
>
> I am using the below originate command and the call gets hung-up at-least
> 2 times in 10 attempts. This happens randomly and I can see on console log
> that freeswitch UNBRIDGES the call. The call is originate and bridge , it
> fails with Destination out of order. To add the call is successfully
> bridged on the PSTN and agent side and then fails quickly. We use bgapi to
> fire this command.
>
> originate {ignore_early_media=false,absolute_codec_string=PCMU,media_webrtc=true,origination_uuid=f8702710-1992-11e9-898c-c371816bde1b,hangup_after_bridge=true,park_after_bridge=true}sofia/gateway/6008_groups/86067830 &bridge([absolute_codec_string=PCMU,hangup_after_bridge=true,park_after_bridge=true,origination_uuid=f8704e20-1992-11e9-898c-c371816bde1b,cc_caller_uuid=f8702710-1992-11e9-898c-c371816bde1b,cc_called_number=919503338275]sofia/gateway/6008_groups/16468107050)
>
> Can you please point me in the right direction?
>
> Thanking You,
> Sunil More
> Ph : 9503338275
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com



-- 

Brian West | Co-founder and Developer

Need Commercial support? email sales at freeswitch.com

FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045
<https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g>

Email: brian at freeswitch.com

Mobile: 918-424-9378

Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/>

[image: https://www.facebook.com/signalwireinc?src=email]
<https://www.facebook.com/freeswitch> [image:
https://twitter.com/freeswitch] <https://twitter.com/freeswitch>
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From brian at freeswitch.com  Thu Jan 17 20:10:28 2019
From: brian at freeswitch.com (Brian West)
Date: Thu, 17 Jan 2019 14:10:28 -0600
Subject: [Freeswitch-users] when using goup_confirm_key=exec,
 the ringback tone is not heard.
In-Reply-To: <CAFi=dz-6Um_Y+cYrZGBAjCraQ=7k9+jZngB3s=Kj2UuCHCYmKQ@mail.gmail.com>
References: <CAFi=dz-6Um_Y+cYrZGBAjCraQ=7k9+jZngB3s=Kj2UuCHCYmKQ@mail.gmail.com>
Message-ID: <CAEJMVkCySs-2McVH+a1cXj-2b80aGXCSt9yj2mnfR4xRxvzpfA@mail.gmail.com>

Work up a test case and file a JIRA.

/b


On Thu, Jan 17, 2019 at 9:45 AM Julien Terrasson <julien.terrasson at gmail.com>
wrote:

> Hi everyone,
>
> I'm working with a dialplan where i need the called party confirmation
> before establishing the call.
>
> So i'm using group_confirm_key=exec (because, not only i need to play a
> file to the caller, but also to the called party, so a little bit of lua
> scripting is needed).
> When doing this, the ringback tone is not heard anymore by the caller
> (when 180 Rining is received, nothing happen).
>
> This doesn't happen with then standard usage of group_confirm_key and
> group_confirm_file, for example group_confirm_key=#.
> group_confirm_file=hello.wav
>
> Any clue on how the RBT can be heard ?
>
> Best regards,
>
> J.Terrasson
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com



-- 

Brian West | Co-founder and Developer

Need Commercial support? email sales at freeswitch.com

FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045
<https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g>

Email: brian at freeswitch.com

Mobile: 918-424-9378

Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/>

[image: https://www.facebook.com/signalwireinc?src=email]
<https://www.facebook.com/freeswitch> [image:
https://twitter.com/freeswitch] <https://twitter.com/freeswitch>
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From KLH at fullrate.dk  Fri Jan 18 06:51:08 2019
From: KLH at fullrate.dk (Kenn Leth Hansen)
Date: Fri, 18 Jan 2019 06:51:08 +0000
Subject: [Freeswitch-users] Screen pop application
In-Reply-To: <CAL1=tORT3_kQrgcO-=5ygAu+hQ0B77EoLRNUqVVCfj9i8OtK1g@mail.gmail.com>
References: <CAL1=tORT3_kQrgcO-=5ygAu+hQ0B77EoLRNUqVVCfj9i8OtK1g@mail.gmail.com>
Message-ID: <DB7PR07MB5419C220330E4DF237C81CAEA59C0@DB7PR07MB5419.eurprd07.prod.outlook.com>

Zoiper Premium should be able to do just that


Regards,

Kenn



________________________________
From: FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> on behalf of Moshe Rosenberg <moshe.rosenberg at gmail.com>
Sent: Wednesday, January 16, 2019 06:43
To: FreeSWITCH Users Help
Subject: [EXT] [Freeswitch-users] Screen pop application

Hi

I am looking for a windows based application that will pop up a web browser with the url + caller I’d each time the phone rings
--
Moshe Rosenberg
Tel. 718 633 1444
Cel. 347 678 3993
www.data<http://www.dataday.consulting/>phone.cloud
Moshe at dataphone.cloud


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From sunil.more64s at gmail.com  Fri Jan 18 07:43:11 2019
From: sunil.more64s at gmail.com (Sunil More)
Date: Fri, 18 Jan 2019 13:13:11 +0530
Subject: [Freeswitch-users] Call Hangup on Freeswitch
In-Reply-To: <CAEJMVkDrafo6Qr8qafguPuPc=zXdU56O0CgB28ZzGt4zm_c2aw@mail.gmail.com>
References: <CANWJJq-FwmPnGF4JCGyYrw+F74axVajvsqu7uHFHo7Ey57KNsQ@mail.gmail.com>
 <CAEJMVkDrafo6Qr8qafguPuPc=zXdU56O0CgB28ZzGt4zm_c2aw@mail.gmail.com>
Message-ID: <CANWJJq-pT=1+_Pk3VTLBWbv-NoeEpEiVJB_R8J-vX_GpAFcwGQ@mail.gmail.com>

Hello Brain,

I have removed it from the dialstring. I have retried after that but still
the call hangs up in the same pattern.

Regards,
Sunil

On Fri, Jan 18, 2019, 2:33 AM Brian West <brian at freeswitch.com wrote:

> Where did you get the info/idea to do ignore_early_media=false? You don't
> have to do that.
>
> On Thu, Jan 17, 2019 at 9:45 AM Sunil More <sunil.more64s at gmail.com>
> wrote:
>
>> Hello All,
>>
>> I am using the below originate command and the call gets hung-up at-least
>> 2 times in 10 attempts. This happens randomly and I can see on console log
>> that freeswitch UNBRIDGES the call. The call is originate and bridge , it
>> fails with Destination out of order. To add the call is successfully
>> bridged on the PSTN and agent side and then fails quickly. We use bgapi to
>> fire this command.
>>
>> originate {ignore_early_media=false,absolute_codec_string=PCMU,media_webrtc=true,origination_uuid=f8702710-1992-11e9-898c-c371816bde1b,hangup_after_bridge=true,park_after_bridge=true}sofia/gateway/6008_groups/86067830 &bridge([absolute_codec_string=PCMU,hangup_after_bridge=true,park_after_bridge=true,origination_uuid=f8704e20-1992-11e9-898c-c371816bde1b,cc_caller_uuid=f8702710-1992-11e9-898c-c371816bde1b,cc_called_number=919503338275]sofia/gateway/6008_groups/16468107050)
>>
>> Can you please point me in the right direction?
>>
>> Thanking You,
>> Sunil More
>> Ph : 9503338275
>> _________________________________________________________________________
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
>
>
> --
>
> Brian West | Co-founder and Developer
>
> Need Commercial support? email sales at freeswitch.com
>
> FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045
> <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g>
>
> Email: brian at freeswitch.com
>
> Mobile: 918-424-9378
>
> Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/>
>
> [image: https://www.facebook.com/signalwireinc?src=email]
> <https://www.facebook.com/freeswitch> [image:
> https://twitter.com/freeswitch] <https://twitter.com/freeswitch>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From sagarmalam at gmail.com  Fri Jan 18 09:28:13 2019
From: sagarmalam at gmail.com (sagar malam)
Date: Fri, 18 Jan 2019 14:58:13 +0530
Subject: [Freeswitch-users] Shared presence not working with callcenter
 application of Freeswitch
In-Reply-To: <CAEZuF3vsdivA5KbMs-JspTYDrOgEUTo5AZiac4LMc8-yU+5c4Q@mail.gmail.com>
References: <CAEZuF3vgN4OdaRha=edmCkhd_Ac-ZLThs6_uXx2LT3FGdVgfdQ@mail.gmail.com>
 <LNXP265MB1180E1DFD30C173B4DFC6759EBE70@LNXP265MB1180.GBRP265.PROD.OUTLOOK.COM>
 <CAEZuF3tbp1tPH9Bt7jDqiVeUAzBToBsD3Ah9mNS2RF+o7B+C9A@mail.gmail.com>
 <6ABFDA54-2016-495F-BB0D-DCFB55874429@jerris.com>
 <CAEZuF3tZz101nOiVLugcszQUa+TxrixnL79375jdW=VOPWf8Rg@mail.gmail.com>
 <CAEZuF3vsdivA5KbMs-JspTYDrOgEUTo5AZiac4LMc8-yU+5c4Q@mail.gmail.com>
Message-ID: <CAEZuF3s4TM1nwdLG2RDf9G-Fz7NaoSDYcqPTp7W6BgPRBOFtWA@mail.gmail.com>

This is not working with latest FS version 1.8 as well.

I will report it to Jira.

On Tue, Oct 30, 2018 at 6:04 PM sagar malam <sagarmalam at gmail.com> wrote:

> Any update on this ?
> Has anyone got this working? I have tested this with FS 1.6.20
> without success.
>
> On Thu, Oct 18, 2018 at 3:54 PM sagar malam <sagarmalam at gmail.com> wrote:
>
>> Yes.
>> Below parameters are enabled :
>> manage-presence
>> manage-shared-appearance
>> dbname = share_presence
>>
>> On Sat, Oct 13, 2018 at 4:07 AM Michael Jerris <mike at jerris.com> wrote:
>>
>>> Is manage presense enabled?
>>>
>>> On Oct 10, 2018, at 5:29 AM, sagar malam <sagarmalam at gmail.com> wrote:
>>>
>>> I have already tried that setting along with sip_invite_domain.But
>>> without any success.Further when i enable SLA and Presence debug , i dont
>>> see any SLA / presence logs which appears in case of one to one calls.
>>>
>>> On Wed, Oct 10, 2018 at 4:54 AM Shaun Stokes <
>>> shaun.stokes at itec-support.co.uk> wrote:
>>>
>>>> You need to set the presence_id variable of your agent when you bridge
>>>> to the agent in the agent contact field.
>>>>
>>>> Shaun
>>>>
>>>> Get Outlook for iOS <https://aka.ms/o0ukef>
>>>> ------------------------------
>>>> *From:* 20110170300n behalf of
>>>> *Sent:* Tuesday, October 9, 2018 19:48
>>>> *To:* FreeSWITCH Users Help
>>>> *Subject:* [Freeswitch-users] Shared presence not working with
>>>> callcenter application of Freeswitch
>>>>
>>>> I am facing issue related to presence/ shared presence for agents in
>>>> callcenter(mod_callcenter).
>>>> Suppose we are dialling a user 1001 at example.com and it is a shared
>>>> user in two phones. Shared presence works perfectly if we directly dial
>>>> extension 1001.
>>>>
>>>> But if same extension is configured as agent of callcenter and call is
>>>> bridged to agent(extension) through callcenter, shared presence does not
>>>> work. Freeswitch does not generate notify for agent state change.
>>>>
>>>> Is this expected behaviour ? If not then what can fix this ?
>>>>
>>>> Thanks in advance
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Services
>>> sales at freeswitch.com
>>> https://freeswitch.com
>>>
>>> Official FreeSWITCH Sites
>>> https://freeswitch.com/oss
>>> https://freeswitch.org/confluence
>>> https://cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> https://freeswitch.com
>>
>>
>
> --
> Thanks,
>
> Sagar
>


-- 
Thanks,

Sagar
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From paul.muaddib83 at gmail.com  Fri Jan 18 10:16:04 2019
From: paul.muaddib83 at gmail.com (Paul Muaddib)
Date: Fri, 18 Jan 2019 11:16:04 +0100
Subject: [Freeswitch-users] sip trunk caller extension is not displayed,
	only main number
Message-ID: <CAEjNe_HwOHJv3TsxmJi9ohi_WzOuMRbTsTq9u4ehpqQNZH1ZnA@mail.gmail.com>

Hi,

I have a SIP Trunk account. From some companys that call me, which also
have a SIP Trunk account, I can only see the main number in the phone
display but not the extension. The correct number with the right extension
is only inside the sip_P-Asserted-Identity variable and not the
caller_id_number.

So I thought I could assign the sip_P-Asserted-Identity to
effective_caller_id_number and all is fine but that is not working.

conf/dialplan/public/voip_inbound.xml

<extension name="telekom-voip-inbound-extension">
   <condition field="${sip_h_P-Called-Party-ID}"
expression="(XXXXXXX(\d{2}))">
      <action application="set" data="domain_name=$${domain}"/>
      <action application="set"
data="effective_caller_id_number=${sip_P-Asserted-Identity}"/>
      <action application="set"
data="effective_caller_id_name=${sip_P-Asserted-Identity}"/>
       <action application="transfer" data="$2 XML default"/>
   </condition>
</extension>
What am I doing wrong?

Best regards
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From jungleboogie0 at gmail.com  Fri Jan 18 22:39:33 2019
From: jungleboogie0 at gmail.com (Jungle Boogie)
Date: Fri, 18 Jan 2019 14:39:33 -0800
Subject: [Freeswitch-users] ESL help on remote machines
In-Reply-To: <CALXCt0oXf_fk7GAxfyp8s=1BHEvYXMiT-+ONVQxw6vSic1fUXg@mail.gmail.com>
References: <20190114232439.GA69791@puffer.in.lylie.net>
 <CALXCt0oXf_fk7GAxfyp8s=1BHEvYXMiT-+ONVQxw6vSic1fUXg@mail.gmail.com>
Message-ID: <20190118223933.GA88312@puffer.in.lylie.net>

On Tue 15 Jan 2019 10:11 AM, Giovanni Maruzzelli wrote:
> On Tue, Jan 15, 2019 at 1:54 AM Jungle Boogie <jungleboogie0 at gmail.com>
> wrote:
> 
> > The perl esl page states the module must be compiled. Does this mean
> > machine
> > A needs to have the entire FS source code to compile the module?
> > https://freeswitch.org/confluence/display/FREESWITCH/Perl+ESL
> >
> > If the entire src isn't needed, what's it take to install the ESL module
> > successfully on machine A?
> >
> >
> I suppose you installed FreeSWITCH from packages, correct?
> A search in installable packages gives you the following:
> 

I installed from master via git. Additionally, machine A isn't running debian. I
think I'll just use a different machine and install debian onto it and try
again.

Thanks for the pkg suggestions, though!

> # apt-cache search freeswitch | grep -i perl
> freeswitch-mod-perl - mod_perl for FreeSWITCH
> freeswitch-mod-perl-dbg - mod_perl for FreeSWITCH (debug)
> libesl-perl - Cross-Platform Scalable Multi-Protocol Soft Switch
> 
> I would install libesl-perl :
> 
> # apt-get install libesl-perl
> 
> -giovanni


From jungleboogie0 at gmail.com  Fri Jan 18 22:42:49 2019
From: jungleboogie0 at gmail.com (Jungle Boogie)
Date: Fri, 18 Jan 2019 14:42:49 -0800
Subject: [Freeswitch-users] ESL help on remote machines
In-Reply-To: <CAFGRPVq=SWe2f4COzwjwj73iZDVn3X8WN3qWas8-iwTwe01xDA@mail.gmail.com>
References: <20190114232439.GA69791@puffer.in.lylie.net>
 <CAFGRPVq=SWe2f4COzwjwj73iZDVn3X8WN3qWas8-iwTwe01xDA@mail.gmail.com>
Message-ID: <20190118224249.GB88312@puffer.in.lylie.net>

On Tue 15 Jan 2019 12:44 PM, David Villasmil wrote:
> You can simply connect to the esl socket (port 8021) on fs from Box A. No
> need for that lua script.
> You simply need to configure the ESL socket to listen on (I.e.: 0.0.0.0)
> and iptables it to only allow your box A.
> As far as the library is concerned, in my experience, yes. The simplest is
> to have the source, so You need the whole fs source and compile. You can
> afterwards remove it.
> 

Box A wasn't Debian, but I tried anyway to install freeswitch onto it for the
perl module. It didn't work out, but I'm not disappointed. The machine was a
pine64 arm64 board running OpenBSD -current. I may file a jira on the build
issues soon.

> Hope that helps.
> 
> If all you need is to originate (meaning you don’t care about events
> afterwards) there is another way of originating, and that’s via the httapi:
> https://freeswitch.org/confluence/plugins/servlet/mobile#content/view/3966423
> 
> With httapi you can all any api in fs.

Ah, that's a nice tip. I'll take a little time to study this method.

> 
> Hope that helps
> 

yes, thanks for your reply!


From akaccr at gmail.com  Sat Jan 19 00:12:18 2019
From: akaccr at gmail.com (Shishir Pokharel)
Date: Fri, 18 Jan 2019 16:12:18 -0800
Subject: [Freeswitch-users] freeswitch core function switch_time_now() vs
 switch_time_now() returns wrong time i.e (now + 3 sec.) vs
 switch_micro_time_now()
Message-ID: <CAESETPk6zYE0MDshH9ia7HtnGCFQ4UiA_DNOs1HaPxhgiTjP8g@mail.gmail.com>

Hi,

I have tried asking the same question on IRC, but I am not sure I missed
the response or I never got any response. So, I am asking the same question
again on the mailing list.

Freeswitch core function switch_time_now() returns wrong timestamp i.e (now
+ 3 sec.) vs switch_micro_time_now() returns correctly. This behavior is
observed only when we do a system reboot and FreeSWITCH is at autostart on
system boot.

If we restart the FreeSWITCH service after a system reboot, both
function returns the timestamp correctly, and no issue is observed. Looking
at the source code of both of the functions I don't see much difference as
switch_micro_time_now() might use switch_time_now()  in some condition.

Any thoughts? on why FreeSWITCH on startup service will have this problem?

Thanks
/Shishir
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From richard at treeboxsolutions.com  Sun Jan 20 16:20:32 2019
From: richard at treeboxsolutions.com (Richard Chan)
Date: Mon, 21 Jan 2019 00:20:32 +0800
Subject: [Freeswitch-users] Are source debs/tar.gz available for libks
	signalwire-client-c?
Message-ID: <CAMXXUi=AUD6RBxJ=LBJn0n2xgnX8kzVOx5KUN=Hi_0sQEXaPLg@mail.gmail.com>

When building 1.8.4 RPMs on CentOS 7 it fails when looking for libks,
signalwire-client-c.

I was looking for the source to build RPMs but couldn't find the source
debs either.

Are these binary only packages?

root at 6f121bb43cbd:/# apt-cache showsrc libks
W: Unable to locate package libks
N: No packages found

root at 6f121bb43cbd:/# apt-cache showsrc signalwire-client-c
W: Unable to locate package signalwire-client-c
N: No packages found


-- 
Richard Chan
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From alex at freeswitch.com  Mon Jan 21 00:01:52 2019
From: alex at freeswitch.com (Alexey Sibyakin)
Date: Mon, 21 Jan 2019 09:01:52 +0900
Subject: [Freeswitch-users] Are source debs/tar.gz available for libks
	signalwire-client-c?
In-Reply-To: <CAMXXUi=AUD6RBxJ=LBJn0n2xgnX8kzVOx5KUN=Hi_0sQEXaPLg@mail.gmail.com>
References: <CAMXXUi=AUD6RBxJ=LBJn0n2xgnX8kzVOx5KUN=Hi_0sQEXaPLg@mail.gmail.com>
Message-ID: <CAGJLwFiatuL+pKhg09NDSzA8keE_FuoK=RivdLGNcX7qdZD8Cw@mail.gmail.com>

https://github.com/signalwire

Regards,

Alex

Alex Sibyakin | Support Engineer SignalWire | 228 Hamilton Ave 3rd Floor,
Palo Alto, CA 94303 Email: alex at freeswitch.com Website:
https://www.signalwire.com



On Mon, Jan 21, 2019 at 4:53 AM Richard Chan <richard at treeboxsolutions.com>
wrote:

> When building 1.8.4 RPMs on CentOS 7 it fails when looking for libks,
> signalwire-client-c.
>
> I was looking for the source to build RPMs but couldn't find the source
> debs either.
>
> Are these binary only packages?
>
> root at 6f121bb43cbd:/# apt-cache showsrc libks
> W: Unable to locate package libks
> N: No packages found
>
> root at 6f121bb43cbd:/# apt-cache showsrc signalwire-client-c
> W: Unable to locate package signalwire-client-c
> N: No packages found
>
>
> --
> Richard Chan
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From cmrienzo at gmail.com  Mon Jan 21 00:56:21 2019
From: cmrienzo at gmail.com (cmrienzo at gmail.com)
Date: Sun, 20 Jan 2019 19:56:21 -0500
Subject: [Freeswitch-users] Are source debs/tar.gz available for libks
	signalwire-client-c?
In-Reply-To: <CAMXXUi=AUD6RBxJ=LBJn0n2xgnX8kzVOx5KUN=Hi_0sQEXaPLg@mail.gmail.com>
References: <CAMXXUi=AUD6RBxJ=LBJn0n2xgnX8kzVOx5KUN=Hi_0sQEXaPLg@mail.gmail.com>
Message-ID: <A66AB509-E0F0-472A-A071-CAC2196D040C@gmail.com>

Check https://github.com/signalwire for signalwire-c and libks. 

Chris


> On Jan 20, 2019, at 11:20, Richard Chan <richard at treeboxsolutions.com> wrote:
> 
> When building 1.8.4 RPMs on CentOS 7 it fails when looking for libks, signalwire-client-c.
> 
> I was looking for the source to build RPMs but couldn't find the source debs either.
> 
> Are these binary only packages?
> 
> root at 6f121bb43cbd:/# apt-cache showsrc libks
> W: Unable to locate package libks
> N: No packages found
> 
> root at 6f121bb43cbd:/# apt-cache showsrc signalwire-client-c
> W: Unable to locate package signalwire-client-c
> N: No packages found
> 
> 
> -- 
> Richard Chan
> 
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
> 
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From sunil.more64s at gmail.com  Mon Jan 21 05:44:55 2019
From: sunil.more64s at gmail.com (Sunil More)
Date: Mon, 21 Jan 2019 11:14:55 +0530
Subject: [Freeswitch-users] Call Hangup on Freeswitch
In-Reply-To: <CANWJJq-pT=1+_Pk3VTLBWbv-NoeEpEiVJB_R8J-vX_GpAFcwGQ@mail.gmail.com>
References: <CANWJJq-FwmPnGF4JCGyYrw+F74axVajvsqu7uHFHo7Ey57KNsQ@mail.gmail.com>
 <CAEJMVkDrafo6Qr8qafguPuPc=zXdU56O0CgB28ZzGt4zm_c2aw@mail.gmail.com>
 <CANWJJq-pT=1+_Pk3VTLBWbv-NoeEpEiVJB_R8J-vX_GpAFcwGQ@mail.gmail.com>
Message-ID: <CANWJJq-eQV2SGomaSiTW0XL1BnODA9238AvrV_zTijmSHFoefA@mail.gmail.com>

Hello All,

Please guide me what to look for. In this case, I can see UNBRIDGE message
in the console logs.

Thanking You,
Sunil More
Ph : 9503338275


On Fri, Jan 18, 2019 at 1:13 PM Sunil More <sunil.more64s at gmail.com> wrote:

> Hello Brain,
>
> I have removed it from the dialstring. I have retried after that but still
> the call hangs up in the same pattern.
>
> Regards,
> Sunil
>
> On Fri, Jan 18, 2019, 2:33 AM Brian West <brian at freeswitch.com wrote:
>
>> Where did you get the info/idea to do ignore_early_media=false? You
>> don't have to do that.
>>
>> On Thu, Jan 17, 2019 at 9:45 AM Sunil More <sunil.more64s at gmail.com>
>> wrote:
>>
>>> Hello All,
>>>
>>> I am using the below originate command and the call gets hung-up
>>> at-least 2 times in 10 attempts. This happens randomly and I can see on
>>> console log that freeswitch UNBRIDGES the call. The call is originate and
>>> bridge , it fails with Destination out of order. To add the call is
>>> successfully bridged on the PSTN and agent side and then fails quickly. We
>>> use bgapi to fire this command.
>>>
>>> originate {ignore_early_media=false,absolute_codec_string=PCMU,media_webrtc=true,origination_uuid=f8702710-1992-11e9-898c-c371816bde1b,hangup_after_bridge=true,park_after_bridge=true}sofia/gateway/6008_groups/86067830 &bridge([absolute_codec_string=PCMU,hangup_after_bridge=true,park_after_bridge=true,origination_uuid=f8704e20-1992-11e9-898c-c371816bde1b,cc_caller_uuid=f8702710-1992-11e9-898c-c371816bde1b,cc_called_number=919503338275]sofia/gateway/6008_groups/16468107050)
>>>
>>> Can you please point me in the right direction?
>>>
>>> Thanking You,
>>> Sunil More
>>> Ph : 9503338275
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Services
>>> sales at freeswitch.com
>>> https://freeswitch.com
>>>
>>> Official FreeSWITCH Sites
>>> https://freeswitch.com/oss
>>> https://freeswitch.org/confluence
>>> https://cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> https://freeswitch.com
>>
>>
>>
>> --
>>
>> Brian West | Co-founder and Developer
>>
>> Need Commercial support? email sales at freeswitch.com
>>
>> FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045
>> <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g>
>>
>> Email: brian at freeswitch.com
>>
>> Mobile: 918-424-9378
>>
>> Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/>
>>
>> [image: https://www.facebook.com/signalwireinc?src=email]
>> <https://www.facebook.com/freeswitch> [image:
>> https://twitter.com/freeswitch] <https://twitter.com/freeswitch>
>> _________________________________________________________________________
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
>
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From tg-maillistings at level5.de  Mon Jan 21 09:58:49 2019
From: tg-maillistings at level5.de (=?UTF-8?Q?Thorsten_G=c3=b6llner?=)
Date: Mon, 21 Jan 2019 10:58:49 +0100
Subject: [Freeswitch-users] sip trunk caller extension is not displayed,
 only main number
In-Reply-To: <CAEjNe_HwOHJv3TsxmJi9ohi_WzOuMRbTsTq9u4ehpqQNZH1ZnA@mail.gmail.com>
References: <CAEjNe_HwOHJv3TsxmJi9ohi_WzOuMRbTsTq9u4ehpqQNZH1ZnA@mail.gmail.com>
Message-ID: <abded6c2-3eb2-8508-9da3-6c0fa398728e@level5.de>

I am not sure. Did you try "export" instead of "set"?

Am 18.01.2019 um 11:16 schrieb Paul Muaddib:
>
> Hi,
>
> I have a SIP Trunk account. From some companys that call me, which
> also have a SIP Trunk account, I can only see the main number in the
> phone display but not the extension. The correct number with the right
> extension is only inside the sip_P-Asserted-Identity variable and not
> the caller_id_number.
>
> So I thought I could assign the sip_P-Asserted-Identity to
> effective_caller_id_number and all is fine but that is not working.
>
> conf/dialplan/public/voip_inbound.xml
>
> <extension name="telekom-voip-inbound-extension">
>    <condition field="${sip_h_P-Called-Party-ID}"
> expression="(XXXXXXX(\d{2}))">
>       <action application="set" data="domain_name=$${domain}"/>
>       <action application="set"
> data="effective_caller_id_number=${sip_P-Asserted-Identity}"/>
>       <action application="set"
> data="effective_caller_id_name=${sip_P-Asserted-Identity}"/>
>        <action application="transfer" data="$2 XML default"/>
>    </condition>
> </extension>
>
> What am I doing wrong?
>
> Best regards
>
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com

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From mailings at itutu.de  Mon Jan 21 12:12:01 2019
From: mailings at itutu.de (=?utf-8?Q?ITutu_-_Mailings?=)
Date: Mon, 21 Jan 2019 13:12:01 +0100
Subject: [Freeswitch-users] sip trunk caller extension is not displayed,
 only main number
In-Reply-To: <abded6c2-3eb2-8508-9da3-6c0fa398728e@level5.de>
References: <CAEjNe_HwOHJv3TsxmJi9ohi_WzOuMRbTsTq9u4ehpqQNZH1ZnA@mail.gmail.com> 
 <abded6c2-3eb2-8508-9da3-6c0fa398728e@level5.de>
Message-ID: <zarafa.5c45b711.7540.4fe7bbc7773180e9@mailit01.itutu.local>

Try <param name="extension" value="auto_to_user"/> in the gateway configuration.

 
Von: FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> Im Auftrag von Thorsten Göllner
Gesendet: Montag, 21. Januar 2019 10:59
An: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>; Paul Muaddib <paul.muaddib83 at gmail.com>
Betreff: Re: [Freeswitch-users] sip trunk caller extension is not displayed, only main number

 
I am not sure. Did you try "export" instead of "set"?

Am 18.01.2019 um 11:16 schrieb Paul Muaddib:

Hi,

I have a SIP Trunk account. From some companys that call me, which also have a SIP Trunk account, I can only see the main number in the phone display but not the extension. The correct number with the right extension is only inside the sip_P-Asserted-Identity variable and not the caller_id_number.

So I thought I could assign the sip_P-Asserted-Identity to effective_caller_id_number and all is fine but that is not working. 

conf/dialplan/public/voip_inbound.xml

<extension name="telekom-voip-inbound-extension">
   <condition field="${sip_h_P-Called-Party-ID}" expression="(XXXXXXX(\d{2}))">
      <action application="set" data="domain_name=$${domain}"/> 
      <action application="set" data="effective_caller_id_number=${sip_P-Asserted-Identity}"/>
      <action application="set" data="effective_caller_id_name=${sip_P-Asserted-Identity}"/> 
       <action application="transfer" data="$2 XML default"/>
   </condition>
</extension>

What am I doing wrong?

 
Best regards

 
 
 




_________________________________________________________________________


Professional FreeSWITCH Services


sales at freeswitch.com <mailto:sales at freeswitch.com> 


https://freeswitch.com


 

Official FreeSWITCH Sites


https://freeswitch.com/oss


https://freeswitch.org/confluence


https://cluecon.com


 

FreeSWITCH-users mailing list


FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org> 


http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> 


UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users> 


https://freeswitch.com

 

_________________________________________________________________________



Professional FreeSWITCH Services



sales at freeswitch.com <mailto:sales at freeswitch.com> 



https://freeswitch.com







Official FreeSWITCH Sites



https://freeswitch.com/oss



https://freeswitch.org/confluence



https://cluecon.com







FreeSWITCH-users mailing list



FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org> 



http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> 



UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users



https://freeswitch.com

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From joe at expert.net  Mon Jan 21 15:31:48 2019
From: joe at expert.net (Joseph Barrero)
Date: Mon, 21 Jan 2019 09:31:48 -0600
Subject: [Freeswitch-users] DOS against FreeSwitch
In-Reply-To: <CAPNTnnq-Wd8LywtvJ9q07LAU2sD3PwfcYZhNB-T559WEaa573w@mail.gmail.com>
References: <CAPNTnnq-Wd8LywtvJ9q07LAU2sD3PwfcYZhNB-T559WEaa573w@mail.gmail.com>
Message-ID: <CAGAa33t0hV=2RoKqVWqho5Ttb_-iCjQOJ+_K2fc+=pPVvovSNA@mail.gmail.com>

This isn't a solution but a possible way to mitigate the effect could be to
enable Sofia Watchdog.

On Tue, Jan 8, 2019 at 3:09 AM Guillermo Ruiz Camauer <grcamauer at gmail.com>
wrote:

> Is the core team aware of this?  Any solutions?
>
>
> http://startrinity.com/VS2/FreeswitchPenetrationTests.aspx#freeswitch_181118
>
>
>
> --
> Guillermo Ruiz Camauer
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From infos at madovsky.org  Mon Jan 21 21:09:08 2019
From: infos at madovsky.org (Madovsky)
Date: Mon, 21 Jan 2019 13:09:08 -0800
Subject: [Freeswitch-users] uuid_record and video settings
Message-ID: <19f759a4-eed7-bcd6-7fdd-758e8ddab680@madovsky.org>

Hi folks,

I wonder where are the settings to control the video (if any) dimension

of a call recording with mod_av? actually it shows a concat legA legB 
video of 1280x720

and would like to reduce it at 480x360. I didn't find it in FS sources



From william at williamcollsassoc.ca  Tue Jan 22 01:50:25 2019
From: william at williamcollsassoc.ca (William Colls)
Date: Mon, 21 Jan 2019 20:50:25 -0500
Subject: [Freeswitch-users] Upgrade from 1.6.? to 1.8.?
Message-ID: <800b6fdb-23ef-69db-3556-fb328fbf8bab@williamcollsassoc.ca>

For reasons beyond my control, I have been away from FreeSWITCH for a 
while.  I now need to get back to it, and get current. Is there a wiki 
entry or other guidance to help me move from 1.6 to 1.8. I see there is 
good guidance for actually building 1.8, so that should not be a 
problem. I am looking for guidance as to changes that I may need to make 
to the various .xml scripts that configure my telephones in house, 
connect to a provider, etc.

Thanks for your time.

William.



From mario_fs at mgtech.com  Tue Jan 22 19:12:48 2019
From: mario_fs at mgtech.com (mario_fs)
Date: Tue, 22 Jan 2019 11:12:48 -0800
Subject: [Freeswitch-users] Upgrade from 1.6.? to 1.8.?
In-Reply-To: <800b6fdb-23ef-69db-3556-fb328fbf8bab@williamcollsassoc.ca>
References: <800b6fdb-23ef-69db-3556-fb328fbf8bab@williamcollsassoc.ca>
Message-ID: <55161664-DA60-4F45-84CA-DD76BA526C1C@mgtech.com>

FYI, while testing before bringing up 1.8 full-time, I just used my 1.6 config and for me it worked, at least for testing. Just do what I did and probably others:

• View the 1.6 and 1.8 conf folders side by side.
• Duplicate the suppllied config files you are going to change. I keep the old one something like .old so I can easily see what I need to change next release. Or use the file change date if you don’t have the old ones.
• Edit the new conf files to add your changes, and review all the lines to see if anything new might be required.
• View new conf files that did not previously exists to see what’s new and if you need to change something.

I updated like this since 2010 and never had an issue. Even if there was a wiki page it may not contain all changes so it’s a good idea to review everything.
MarioG

> On Jan 21, 2019, at 5:50 PM, William Colls <william at williamcollsassoc.ca> wrote:
> 
> For reasons beyond my control, I have been away from FreeSWITCH for a while.  I now need to get back to it, and get current. Is there a wiki entry or other guidance to help me move from 1.6 to 1.8. I see there is good guidance for actually building 1.8, so that should not be a problem. I am looking for guidance as to changes that I may need to make to the various .xml scripts that configure my telephones in house, connect to a provider, etc.
> 
> Thanks for your time.
> 
> William.
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
> 
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com



From mike at freeswitch.com  Tue Jan 22 20:31:25 2019
From: mike at freeswitch.com (Mike Jerris)
Date: Tue, 22 Jan 2019 15:31:25 -0500
Subject: [Freeswitch-users] Upgrade from 1.6.? to 1.8.?
In-Reply-To: <800b6fdb-23ef-69db-3556-fb328fbf8bab@williamcollsassoc.ca>
References: <800b6fdb-23ef-69db-3556-fb328fbf8bab@williamcollsassoc.ca>
Message-ID: <CADrq_yEM+MrctZoodcrxe_LLFwBkUr0HsYMfs3PyxeXFKyqexA@mail.gmail.com>

for upgrade purposes, treat 1.8 like more 1.6 versions, shouldn’t be
anything major

On Tue, Jan 22, 2019 at 10:29 AM William Colls <william at williamcollsassoc.ca>
wrote:

> For reasons beyond my control, I have been away from FreeSWITCH for a
> while.  I now need to get back to it, and get current. Is there a wiki
> entry or other guidance to help me move from 1.6 to 1.8. I see there is
> good guidance for actually building 1.8, so that should not be a
> problem. I am looking for guidance as to changes that I may need to make
> to the various .xml scripts that configure my telephones in house,
> connect to a provider, etc.
>
> Thanks for your time.
>
> William.
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From nandy1925 at gmail.com  Tue Jan 22 23:09:55 2019
From: nandy1925 at gmail.com (Nandy Dagondon)
Date: Tue, 22 Jan 2019 23:09:55 +0000
Subject: [Freeswitch-users] Selecting mod_verto/webrtc ICE candidate
Message-ID: <CAHKqMb7UAhrW6iBP9MgXnKG9fL4Fo=x1M852G-33Lco-aqTv8w@mail.gmail.com>

Is it possible to choose which ICE candidate to use?  I found there are
private candidates (in the internal LAN network) and public candidates.  I
want to choose, for example, public candidate.  Where and how is it done?

Tks. /Nandy
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From darshanmody at avaya.com  Wed Jan 23 10:50:49 2019
From: darshanmody at avaya.com (Mody, Darshan Arvindkumar (Darshan))
Date: Wed, 23 Jan 2019 10:50:49 +0000
Subject: [Freeswitch-users] Configure Freeswitch as B2BUA
Message-ID: <MN2PR15MB2671D6F591F4CCDDCEC087ACA7990@MN2PR15MB2671.namprd15.prod.outlook.com>

Hi

I have gone through the documentation of Freeswitch. I did not find any documentation on how to configure Freeswitch as B2BUA?

Do we have any configuration that requires for Freeswitch to specifically work as B2BUA?

Thanks
Darshan
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From ko at sv01.de  Wed Jan 23 11:57:25 2019
From: ko at sv01.de (Kevin Olbrich)
Date: Wed, 23 Jan 2019 12:57:25 +0100
Subject: [Freeswitch-users] Upgrade from 1.6.? to 1.8.?
In-Reply-To: <CADrq_yEM+MrctZoodcrxe_LLFwBkUr0HsYMfs3PyxeXFKyqexA@mail.gmail.com>
References: <800b6fdb-23ef-69db-3556-fb328fbf8bab@williamcollsassoc.ca>
 <CADrq_yEM+MrctZoodcrxe_LLFwBkUr0HsYMfs3PyxeXFKyqexA@mail.gmail.com>
Message-ID: <CA+gLzy_zGvJJm6FUTYiDbE3PvkMniK+wrMAwGgfyPYV++eZ2mQ@mail.gmail.com>

Am Di., 22. Jan. 2019 um 22:15 Uhr schrieb Mike Jerris <mike at freeswitch.com>:
>
> for upgrade purposes, treat 1.8 like more 1.6 versions, shouldn’t be anything major

I second that. Upgraded a large (by concurrent calls) installation
from Deb8 + FS 1.6 to Deb9 + FS1.8 over night.
Works perfectly fine with the same configuration.

> On Tue, Jan 22, 2019 at 10:29 AM William Colls <william at williamcollsassoc.ca> wrote:
>>
>> For reasons beyond my control, I have been away from FreeSWITCH for a
>> while.  I now need to get back to it, and get current. Is there a wiki
>> entry or other guidance to help me move from 1.6 to 1.8. I see there is
>> good guidance for actually building 1.8, so that should not be a
>> problem. I am looking for guidance as to changes that I may need to make
>> to the various .xml scripts that configure my telephones in house,
>> connect to a provider, etc.
>>
>> Thanks for your time.
>>
>> William.
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com


From mike at freeswitch.com  Wed Jan 23 14:27:38 2019
From: mike at freeswitch.com (Mike Jerris)
Date: Wed, 23 Jan 2019 09:27:38 -0500
Subject: [Freeswitch-users] FS and gcc 8.x.x
In-Reply-To: <6b0f3278-c67f-040f-9561-b5ece7f2ead3@madovsky.org>
References: <aeed5aef-64b9-6cf2-f362-70d8409c2426@madovsky.org>
 <CAEJMVkCqVxSTDni83QjQcuVOwZY5ipFKaiQ7-BnFGkU32t_hcg@mail.gmail.com>
 <c39aa5c6-1afb-898a-e86e-a552be1b3a1b@madovsky.org>
 <6b0f3278-c67f-040f-9561-b5ece7f2ead3@madovsky.org>
Message-ID: <CADrq_yFo7fYkpLnVxRcu0eM4OtT0Dtb9fe76oS=PpsTQ7=WHYA@mail.gmail.com>

please submit patches that makes this work when needed without special args
like this

On Wed, Jan 16, 2019 at 11:16 AM Madovsky <infos at madovsky.org> wrote:

> for those who will encounter the same compiling issue on Fedora 28 64bits
>
> just use
>
> ./configure CFLAGS="-D__alloca=alloca" .....
>
> and everything is ok
> On 12/11/2018 5:35 AM, Madovsky wrote:
>
> https://freeswitch.org/jira/browse/FS-11170
>
> thanks
>
> On 12/10/2018 6:02 AM, Brian West wrote:
>
> Can you please send a bug report to JIRA?
> Thanks,
> /b
>
>
> On Mon, Dec 10, 2018 at 3:09 AM Madovsky <infos at madovsky.org> wrote:
>
>> Just FYI FS does not compile anymore with gcc 8.x.x
>>
>> major changes have been made with this new version of gcc, especially
>> strncpy and strncat.
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
>
>
> --
>
> Brian West | Co-founder and Developer
>
> Need Commercial support? email sales at freeswitch.com
>
> FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045
> <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g>
>
> Email: brian at freeswitch.com
>
> Mobile: 918-424-9378
>
> Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/>
>
> [image: https://www.facebook.com/signalwireinc?src=email]
> <https://www.facebook.com/freeswitch> [image:
> https://twitter.com/freeswitch] <https://twitter.com/freeswitch>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Servicessales at freeswitch.comhttps://freeswitch.com
>
> Official FreeSWITCH Siteshttps://freeswitch.com/osshttps://freeswitch.org/confluencehttps://cluecon.com
>
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttps://freeswitch.com
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From gmaruzz at gmail.com  Wed Jan 23 17:16:15 2019
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Wed, 23 Jan 2019 18:16:15 +0100
Subject: [Freeswitch-users] Configure Freeswitch as B2BUA
In-Reply-To: <MN2PR15MB2671D6F591F4CCDDCEC087ACA7990@MN2PR15MB2671.namprd15.prod.outlook.com>
References: <MN2PR15MB2671D6F591F4CCDDCEC087ACA7990@MN2PR15MB2671.namprd15.prod.outlook.com>
Message-ID: <CALXCt0qD+E0OM_0QYzmjBTnBkaTj1DA3R9_+EtH9wXEGGuCBWw@mail.gmail.com>

On Wed, Jan 23, 2019 at 6:07 PM Mody, Darshan Arvindkumar (Darshan) <
darshanmody at avaya.com> wrote:

>
>
> I have gone through the documentation of Freeswitch. I did not find any
> documentation on how to configure Freeswitch as B2BUA?
>
>
>
> Do we have any configuration that requires for Freeswitch to specifically
> work as B2BUA?
>


FreeSWITCH is ONLY a B2BUA, cannot work in any other way, and no specific
configuration is needed for making it work as B2BUA.

All documentation is all about the B2BUA configuration, eg, FreeSWITCH
configuration.

-giovanni




>
>
> Thanks
>
> Darshan
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com



-- 
Sincerely,

Giovanni Maruzzelli
OpenTelecom.IT
cell: +39 347 266 56 18
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From keith at rhizomatica.org  Wed Jan 23 22:45:00 2019
From: keith at rhizomatica.org (Keith)
Date: Wed, 23 Jan 2019 23:45:00 +0100
Subject: [Freeswitch-users] change ring- back after 180 from one of multiple
	destinations?
Message-ID: <c63859ab-d27c-ab51-30f7-259160b62bfa@rhizomatica.org>

Hi all.

I'm trying to design a dialplan (in python) with some audible feedback
for users.

It occurred to me that it would be nice if I use instant_ringback and
then could change the ring-back once one of multiple destinations sends
a 180.

I know that I can do it where if the remote sends 183 and ring-back, it
will "take-over", but in this case, my callee won't generate ringback.

An alternative would be to do it the other way around, I bridge to two
destinations and one destination 1 hangs up with SUBSCRIBER_ABSENT for
example, I could change the ring-back.

I cannot however find anyway to do this in documentation or hints on
forums etc.

I supposed the basic question here is, Is there anyway to know what is
happening, or indeed prepare for such events before calling bridge?


Many thanks!!

Keith.




From julien.terrasson at gmail.com  Fri Jan 25 15:34:19 2019
From: julien.terrasson at gmail.com (Julien Terrasson)
Date: Fri, 25 Jan 2019 16:34:19 +0100
Subject: [Freeswitch-users] when using goup_confirm_key=exec,
 the ringback tone is not heard. (Brian West)
In-Reply-To: <mailman.0.1547812803.65194.freeswitch-users@lists.freeswitch.org>
References: <mailman.0.1547812803.65194.freeswitch-users@lists.freeswitch.org>
Message-ID: <CAFi=dz-mp=-8Hg75YaHz0jzuGfQfVyxMyi3D9xKo189FaJveVg@mail.gmail.com>

Thank for the reply Brian but i found a workaround :

I put A_party is in hold just before B_party is bridged.
=> "A_party" will listen music_on_hold when "B_party" is ringing.
That way A_party knows that his call is still being processed (eventhough
no RBT can be heard)

Julien


On Fri, Jan 18, 2019 at 1:00 PM <
freeswitch-users-request at lists.freeswitch.org> wrote:

> Send FreeSWITCH-users mailing list submissions to
>         freeswitch-users at lists.freeswitch.org
>
> To subscribe or unsubscribe via the World Wide Web, visit
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> or, via email, send a message with subject or body 'help' to
>         freeswitch-users-request at lists.freeswitch.org
>
> You can reach the person managing the list at
>         freeswitch-users-owner at lists.freeswitch.org
>
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of FreeSWITCH-users digest..."
> Today's Topics:
>
>    1. Re: when using goup_confirm_key=exec, the ringback tone is
>       not heard. (Brian West)
>
>
>
> ---------- Forwarded message ----------
> From: Brian West <brian at freeswitch.com>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Cc:
> Bcc:
> Date: Thu, 17 Jan 2019 14:10:28 -0600
> Subject: Re: [Freeswitch-users] when using goup_confirm_key=exec, the
> ringback tone is not heard.
> Work up a test case and file a JIRA.
>
> /b
>
>
> On Thu, Jan 17, 2019 at 9:45 AM Julien Terrasson <
> julien.terrasson at gmail.com> wrote:
>
>> Hi everyone,
>>
>> I'm working with a dialplan where i need the called party confirmation
>> before establishing the call.
>>
>> So i'm using group_confirm_key=exec (because, not only i need to play a
>> file to the caller, but also to the called party, so a little bit of lua
>> scripting is needed).
>> When doing this, the ringback tone is not heard anymore by the caller
>> (when 180 Rining is received, nothing happen).
>>
>> This doesn't happen with then standard usage of group_confirm_key and
>> group_confirm_file, for example group_confirm_key=#.
>> group_confirm_file=hello.wav
>>
>> Any clue on how the RBT can be heard ?
>>
>> Best regards,
>>
>> J.Terrasson
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
>
>
> --
>
> Brian West | Co-founder and Developer
>
> Need Commercial support? email sales at freeswitch.com
>
> FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045
> <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g>
>
> Email: brian at freeswitch.com
>
> Mobile: 918-424-9378
>
> Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/>
>
> [image: https://www.facebook.com/signalwireinc?src=email]
> <https://www.facebook.com/freeswitch> [image:
> https://twitter.com/freeswitch] <https://twitter.com/freeswitch>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From julien.terrasson at gmail.com  Fri Jan 25 16:00:28 2019
From: julien.terrasson at gmail.com (Julien Terrasson)
Date: Fri, 25 Jan 2019 17:00:28 +0100
Subject: [Freeswitch-users] how to keep called leg alive whan calling leg
 hangup while group_confirm is used ?
In-Reply-To: <CAFi=dz-6Um_Y+cYrZGBAjCraQ=7k9+jZngB3s=Kj2UuCHCYmKQ@mail.gmail.com>
References: <CAFi=dz-6Um_Y+cYrZGBAjCraQ=7k9+jZngB3s=Kj2UuCHCYmKQ@mail.gmail.com>
Message-ID: <CAFi=dz9YQ61zQM+LhV76yhuePomDtiyWp2NoUFMYv8_x_Bm2tg@mail.gmail.com>

Hi everyone,

I'm using the group_confirm to request a confirmation from the called
party, before bridging the call.
However, if the caller hangup when the the confirmation message is played,
the called party get disconnected without understanding why.
Is there a way to keep called party alive for a while, time to playback a
file and let him know why the call is hanged up ?

PS : Note that hangup_after_bridge doesn't help here (since the call is not
yet bridged..)

Julien




On Thu, Jan 17, 2019 at 3:54 PM Julien Terrasson <julien.terrasson at gmail.com>
wrote:

> Hi everyone,
>
> I'm working with a dialplan where i need the called party confirmation
> before establishing the call.
>
> So i'm using group_confirm_key=exec (because, not only i need to play a
> file to the caller, but also to the called party, so a little bit of lua
> scripting is needed).
> When doing this, the ringback tone is not heard anymore by the caller
> (when 180 Rining is received, nothing happen).
>
> This doesn't happen with then standard usage of group_confirm_key and
> group_confirm_file, for example group_confirm_key=#.
> group_confirm_file=hello.wav
>
> Any clue on how the RBT can be heard ?
>
> Best regards,
>
> J.Terrasson
>
>
>
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From vma at vallimamod.org  Mon Jan 28 21:39:38 2019
From: vma at vallimamod.org (Vallimamod Abdullah)
Date: Mon, 28 Jan 2019 22:39:38 +0100
Subject: [Freeswitch-users] how to keep called leg alive whan calling
 leg hangup while group_confirm is used ?
In-Reply-To: <CAFi=dz9YQ61zQM+LhV76yhuePomDtiyWp2NoUFMYv8_x_Bm2tg@mail.gmail.com>
References: <CAFi=dz-6Um_Y+cYrZGBAjCraQ=7k9+jZngB3s=Kj2UuCHCYmKQ@mail.gmail.com>
 <CAFi=dz9YQ61zQM+LhV76yhuePomDtiyWp2NoUFMYv8_x_Bm2tg@mail.gmail.com>
Message-ID: <43D0120B-F570-4947-A342-74339D465D5F@vallimamod.org>

Hi,

Have you set continue_on_fail=true along with hangup_after_bridge=true ?
That way, you can add a playback after the bridge and it will only be played when the bridge fails.


Best Regards,
-- 
Vallimamod Abdullah
SIP Solutions
vma at sip.solutions
linkedin.com/in/vallimamod
.



> On 25 Jan 2019, at 17:00, Julien Terrasson <julien.terrasson at gmail.com> wrote:
> 
> Hi everyone,
> 
> I'm using the group_confirm to request a confirmation from the called party, before bridging the call.
> However, if the caller hangup when the the confirmation message is played, the called party get disconnected without understanding why.
> Is there a way to keep called party alive for a while, time to playback a file and let him know why the call is hanged up ?
> 
> PS : Note that hangup_after_bridge doesn't help here (since the call is not yet bridged..)
> 
> Julien
> 
> 
> 
> 
> On Thu, Jan 17, 2019 at 3:54 PM Julien Terrasson <julien.terrasson at gmail.com> wrote:
> Hi everyone,
> 
> I'm working with a dialplan where i need the called party confirmation before establishing the call.
> 
> So i'm using group_confirm_key=exec (because, not only i need to play a file to the caller, but also to the called party, so a little bit of lua scripting is needed).
> When doing this, the ringback tone is not heard anymore by the caller (when 180 Rining is received, nothing happen).
> 
> This doesn't happen with then standard usage of group_confirm_key and group_confirm_file, for example group_confirm_key=#. group_confirm_file=hello.wav
> 
> Any clue on how the RBT can be heard ?
> 
> Best regards,
> 
> J.Terrasson



From mansig130 at gmail.com  Wed Jan 30 08:46:42 2019
From: mansig130 at gmail.com (Mansi Gupta)
Date: Wed, 30 Jan 2019 14:16:42 +0530
Subject: [Freeswitch-users] Unimrcp recognizer channel error
Message-ID: <CAGN7-NC0mH2+TPYbn-NEebdiqt8D84Wgv0FMY-kWeHH-i535fQ@mail.gmail.com>

Hello Everyone,
I'm having an issue working with mod_unimrcp .
I keep getting the error of RECOGNIZER CHANNEL ERROR!.
Anyone having idea how to solve the issue...
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From mansig130 at gmail.com  Wed Jan 30 09:38:29 2019
From: mansig130 at gmail.com (Mansi Gupta)
Date: Wed, 30 Jan 2019 02:38:29 -0700 (MST)
Subject: [Freeswitch-users] ASR Recognizer channel error
Message-ID: <1548841109174-0.post@n2.nabble.com>

Hello Everyone,
I'm having an issue working with mod_unimrcp.I am in the process of
configuring a speech recognition IVR with the help of mod_unimrcp in
freeswitch 1.6.This seems to work fine however, i keep getting the error of
RECOGNIZER CHANNEL:
 
[ERR] mod_unimrcp.c:1913 (ASR-12) RECOGNIZER channel error!

Here are few relevant lines of the log:

2019-01-30 04:28:20.072090 [DEBUG] apt_task.c:265 () Signal Message to [MRCP
Client] [0x7fed34016200;1;0]
2019-01-30 04:28:20.072090 [DEBUG] apt_poller_task.c:251 () Wait for
Messages [voxeo-prophecy8.0-mrcp1]
2019-01-30 04:28:20.072090 [DEBUG] apt_task.c:337 () Process Message [MRCP
Client] [0x7fed34016200;1;0]
2019-01-30 04:28:20.072090 [ERR] mod_unimrcp.c:1913 (ASR-13) RECOGNIZER
channel error!
2019-01-30 04:28:20.072090 [DEBUG] mod_unimrcp.c:1577 (ASR-13) CLOSED ==>
ERROR
2019-01-30 04:28:20.091506 [DEBUG] apt_consumer_task.c:141 () Wait for
Messages [MRCP Client]
2019-01-30 04:28:20.091506 [DEBUG] mod_unimrcp.c:1061 (ASR-13) Terminating
MRCP session
2019-01-30 04:28:20.091506 [DEBUG] apt_task.c:265 () Signal Message to [MRCP
Client] [0x7fed300afe10;4;0]
2019-01-30 04:28:20.091506 [DEBUG] apt_task.c:337 () Process Message [MRCP
Client] [0x7fed300afe10;4;0]
2019-01-30 04:28:20.091506 [INFO] mrcp_client_session.c:387 (ASR-13) Receive
App Request ASR-13 <new> [1]
2019-01-30 04:28:20.091506 [DEBUG] mrcp_client_session.c:1283 (ASR-13)
Dispatch App Request ASR-13 <new> [1]

Any ideas will be appreciated. Thanks



--
Sent from: http://freeswitch-users.2379917.n2.nabble.com/


From mansig130 at gmail.com  Wed Jan 30 10:15:17 2019
From: mansig130 at gmail.com (Mansi Gupta)
Date: Wed, 30 Jan 2019 15:45:17 +0530
Subject: [Freeswitch-users] ASR recognizer channel error.
Message-ID: <CAGN7-NDC+GMdSoKbeUmb7OecHfXf1MOgBQ9jNORe_tUqdqquoA@mail.gmail.com>

Hello everyone,
I'm in the process of configuring a speech recognition IVR using freeswitch
1.6 with the help of mod_unimrcp.This seems to be work fine,however i keep
getting this error:
[ERR] mod_unimrcp.c:1913 (ASR-14) RECOGNIZER channel error!

Here are few relevant lines of log:

2019-01-30 04:30:27.312074 [DEBUG] apt_task.c:265 () Signal Message to
[MRCP Client] [0x7fed3404b980;1;0]
2019-01-30 04:30:27.312074 [DEBUG] apt_poller_task.c:251 () Wait for
Messages [voxeo-prophecy8.0-mrcp1]
2019-01-30 04:30:27.312074 [DEBUG] apt_task.c:337 () Process Message [MRCP
Client] [0x7fed3404b980;1;0]
2019-01-30 04:30:27.312074 [INFO] mrcp_client_session.c:151 (ASR-14)
Receive Answer ASR-14 <new> [c:0 a:0 v:0] Status 0
2019-01-30 04:30:27.312074 [INFO] mrcp_client_session.c:455 (ASR-14) Raise
App Response ASR-14 <new> [2] FAILURE [2]
*2019-01-30 04:30:27.312074 [ERR] mod_unimrcp.c:1913 (ASR-14) RECOGNIZER
channel error!*
2019-01-30 04:30:27.312074 [DEBUG] mod_unimrcp.c:1577 (ASR-14) CLOSED ==>
ERROR
2019-01-30 04:30:27.312074 [DEBUG] apt_consumer_task.c:141 () Wait for
Messages [MRCP Client]
2019-01-30 04:30:27.312074 [DEBUG] mod_unimrcp.c:1061 (ASR-14) Terminating
MRCP session
2019-01-30 04:30:27.312074 [DEBUG] apt_task.c:265 () Signal Message to
[MRCP Client] [0x7fed30023ae0;4;0]
2019-01-30 04:30:27.312074 [DEBUG] apt_task.c:337 () Process Message [MRCP
Client] [0x7fed30023ae0;4;0]
2019-01-30 04:30:27.312074 [INFO] mrcp_client_session.c:387 (ASR-14)
Receive App Request ASR-14 <new> [1]

Any opinions will be appreciated. Thanks.
-- 
Mansi Gupta
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From rayk at pontimax.com  Wed Jan 30 19:02:40 2019
From: rayk at pontimax.com (Ray Keating)
Date: Wed, 30 Jan 2019 14:02:40 -0500
Subject: [Freeswitch-users] Unimrcp recognizer channel error
In-Reply-To: <CAGN7-NC0mH2+TPYbn-NEebdiqt8D84Wgv0FMY-kWeHH-i535fQ@mail.gmail.com>
References: <CAGN7-NC0mH2+TPYbn-NEebdiqt8D84Wgv0FMY-kWeHH-i535fQ@mail.gmail.com>
Message-ID: <03df01d4b8ce$623dfb70$26b9f250$@pontimax.com>

 

A FS log entry of RECOGNIZER CHANNEL ERROR means one (or more, possibly) of the following is true:

 

the IP connection to the machine hosting the MRCP capable Speech Recognition Server is broken, or,

the Speech Recognition Server itself is not accepting the connection request, or,

your MRCP profile configuration entry doesn’t have the correct IP address/port number

for connecting to the MRCP Server.

 

Ray Keating/www.pontimax.com

 

“Pontimax’s mrcpSP11-STT- the lowest cost, by far, highest recognition accuracy MRCP Speech Recognition Server available”

 

From: Mansi Gupta <mansig130 at gmail.com> 
Sent: Wednesday, January 30, 2019 3:47 AM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Unimrcp recognizer channel error

 

Hello Everyone,
I'm having an issue working with mod_unimrcp .
I keep getting the error of RECOGNIZER CHANNEL ERROR!.
Anyone having idea how to solve the issue...

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From admin at smallunix.net  Wed Jan 30 19:19:58 2019
From: admin at smallunix.net (Andrea Mazzeo)
Date: Wed, 30 Jan 2019 20:19:58 +0100
Subject: [Freeswitch-users] zRTP + OPUS one-way audio issue
In-Reply-To: <CABms_cVpkrN9OORfqKRrLBKftzji_gpd0p4pm=0JQV=ZRcskyA@mail.gmail.com>
References: <CABms_cU_wasa_mZf-pvVykLFkNc+4+PSVcMrGZ9+JmOXfGd9Ng@mail.gmail.com>
 <CABms_cVpkrN9OORfqKRrLBKftzji_gpd0p4pm=0JQV=ZRcskyA@mail.gmail.com>
Message-ID: <CABms_cX=--Ci+_G+zke=tO8t8XMcrJK775=odwp0g6a5_8=J1A@mail.gmail.com>

Hi,

Is there any hack to suggest me to change payload 102 to 103 inside SDP
before to send to LEG-B?

I would like to send
m=audio 21820 RTP/SAVP 103 101 13
a=rtpmap:103 opus/48000/2
a=fmtp:103 useinbandfec=1; maxaveragebitrate=30000; maxplaybackrate=48000;
ptime=20; minptime=10; maxptime=40; stereo=1

instead of
m=audio 21820 RTP/SAVP 102 101 13
a=rtpmap:102 opus/48000/2
a=fmtp:102 useinbandfec=1; maxaveragebitrate=30000; maxplaybackrate=48000;
ptime=20; minptime=10; maxptime=40; stereo=1

I'm trying to use regex with switch_r_sdp before bridge, but I'm not
lucky...

Thank you,
Andrea

Il giorno mar 2 ott 2018 alle ore 11:09 Andrea Mazzeo <admin at smallunix.net>
ha scritto:

> Dear All,
>
> I opened a bug for this:
> https://freeswitch.org/jira/browse/FS-11402
>
> I would like to offer a bounty, but it's not easy for me to evaluate the
> amount of needed work.
>
> Please if someone is interested let me know or better share a bid.
>
> Thank you,
> Andrea
>
> Il giorno ven 21 set 2018 alle ore 12:47 Andrea Mazzeo <
> admin at smallunix.net> ha scritto:
>
>> Hi,
>>
>> I'm facing an one-way audio issue with zRTP while using OPUS. zRTP is set
>> in passthru mode.
>> Called party can hear Calling party but not vice versa.
>>
>> Full log:
>> https://pastebin.freeswitch.org/view/639adfe9
>>
>> This issue is happening *only* with dynamic payloads codecs.
>> Everything is working fine if I use G722 or G729.
>>
>> Scenario Leg-A (207) -> FS -> Leg-B (213)
>>
>> Checking FS logs, I see
>>
>> Leg-A's SDP:
>> a=rtpmap:103 opus/48000/2
>>
>> Leg-B's SDP:
>> a=rtpmap:102 opus/48000/2
>>
>> Actually having different payloads should not be an issue.
>> I opened a case to Acrobits, they said the issue is in the actual RTP
>> traffic sent to Leg-A from FS:
>>
>> From the client log Leg-A
>> Sending RTP packet #200 192.168.128.158:59176 > 31.102.111.134:28560,
>> len=116, really=116,
>> data=80679E55DF67E34A291167D899830CC03C124F353DA3C069574410EC0333125F
>> This is using 103, correct as agreed on this side of the call
>>
>> Received RTP packet #200 31.102.111.134:28560 > 192.168.128.158:59176,
>> len=90,
>> data=80663D3B22CC8CB414490864A230F49B6EFF42E7BCA9EB64B864374B1377965C
>> This is using 102, which is wrong for this side of the call.
>>
>> Seems that after negotiated payload 102 with Leg-B, FS is trying to use
>> it on Leg-A, where it should be used 102 instead.
>>
>> Any idea?
>>
>> FreeSWITCH Version 1.8.1-2-4f54cff~64bit (-2-4f54cff 64bit)
>> OS.: Debian 8.11
>> Linux pbx-186 3.16.0-4-amd64 #1 SMP Debian 3.16.51-3 (2017-12-13) x86_64
>> GNU/Linux
>>
>> Thank you,
>> Andrea Mazzeo
>>
>
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From mike at freeswitch.org  Wed Jan 30 20:29:07 2019
From: mike at freeswitch.org (Mike Jerris)
Date: Wed, 30 Jan 2019 15:29:07 -0500
Subject: [Freeswitch-users] ASR Recognizer channel error
In-Reply-To: <1548841109174-0.post@n2.nabble.com>
References: <1548841109174-0.post@n2.nabble.com>
Message-ID: <A3C73FA6-EE84-4758-B2F1-608F9D420457@freeswitch.org>

Try out latest freeswitch release first before you try anything else.

> On Jan 30, 2019, at 4:38 AM, Mansi Gupta <mansig130 at gmail.com> wrote:
> 
> Hello Everyone,
> I'm having an issue working with mod_unimrcp.I am in the process of
> configuring a speech recognition IVR with the help of mod_unimrcp in
> freeswitch 1.6.This seems to work fine however, i keep getting the error of
> RECOGNIZER CHANNEL:
> 
> [ERR] mod_unimrcp.c:1913 (ASR-12) RECOGNIZER channel error!
> 
> Here are few relevant lines of the log:
> 
> 2019-01-30 04:28:20.072090 [DEBUG] apt_task.c:265 () Signal Message to [MRCP
> Client] [0x7fed34016200;1;0]
> 2019-01-30 04:28:20.072090 [DEBUG] apt_poller_task.c:251 () Wait for
> Messages [voxeo-prophecy8.0-mrcp1]
> 2019-01-30 04:28:20.072090 [DEBUG] apt_task.c:337 () Process Message [MRCP
> Client] [0x7fed34016200;1;0]
> 2019-01-30 04:28:20.072090 [ERR] mod_unimrcp.c:1913 (ASR-13) RECOGNIZER
> channel error!
> 2019-01-30 04:28:20.072090 [DEBUG] mod_unimrcp.c:1577 (ASR-13) CLOSED ==>
> ERROR
> 2019-01-30 04:28:20.091506 [DEBUG] apt_consumer_task.c:141 () Wait for
> Messages [MRCP Client]
> 2019-01-30 04:28:20.091506 [DEBUG] mod_unimrcp.c:1061 (ASR-13) Terminating
> MRCP session
> 2019-01-30 04:28:20.091506 [DEBUG] apt_task.c:265 () Signal Message to [MRCP
> Client] [0x7fed300afe10;4;0]
> 2019-01-30 04:28:20.091506 [DEBUG] apt_task.c:337 () Process Message [MRCP
> Client] [0x7fed300afe10;4;0]
> 2019-01-30 04:28:20.091506 [INFO] mrcp_client_session.c:387 (ASR-13) Receive
> App Request ASR-13 <new> [1]
> 2019-01-30 04:28:20.091506 [DEBUG] mrcp_client_session.c:1283 (ASR-13)
> Dispatch App Request ASR-13 <new> [1]
> 
> Any ideas will be appreciated. Thanks
> 



From sdisanti.eteleco at gmail.com  Wed Jan 30 21:10:05 2019
From: sdisanti.eteleco at gmail.com (Sean DiSanti)
Date: Wed, 30 Jan 2019 13:10:05 -0800
Subject: [Freeswitch-users] ASR Recognizer channel error
In-Reply-To: <1548841109174-0.post@n2.nabble.com>
References: <1548841109174-0.post@n2.nabble.com>
Message-ID: <CAANGHxvJG9sH+-YhaVJw_OgYHwTbEwbEPvRkdph_V1VacMWeYA@mail.gmail.com>

If you're sure that your ip addresses are correct in the mrcp profile and
configurations, and you're specifying unimrcp:profilename in your
detections, restart freeswitch and unimrcp services and try again. That's
one thing I've run into where if I only restart one of the services, I get
that error until I restart the other.

On Wed, Jan 30, 2019 at 7:37 AM Mansi Gupta <mansig130 at gmail.com> wrote:

> Hello Everyone,
> I'm having an issue working with mod_unimrcp.I am in the process of
> configuring a speech recognition IVR with the help of mod_unimrcp in
> freeswitch 1.6.This seems to work fine however, i keep getting the error of
> RECOGNIZER CHANNEL:
>
> [ERR] mod_unimrcp.c:1913 (ASR-12) RECOGNIZER channel error!
>
> Here are few relevant lines of the log:
>
> 2019-01-30 04:28:20.072090 [DEBUG] apt_task.c:265 () Signal Message to
> [MRCP
> Client] [0x7fed34016200;1;0]
> 2019-01-30 04:28:20.072090 [DEBUG] apt_poller_task.c:251 () Wait for
> Messages [voxeo-prophecy8.0-mrcp1]
> 2019-01-30 04:28:20.072090 [DEBUG] apt_task.c:337 () Process Message [MRCP
> Client] [0x7fed34016200;1;0]
> 2019-01-30 04:28:20.072090 [ERR] mod_unimrcp.c:1913 (ASR-13) RECOGNIZER
> channel error!
> 2019-01-30 04:28:20.072090 [DEBUG] mod_unimrcp.c:1577 (ASR-13) CLOSED ==>
> ERROR
> 2019-01-30 04:28:20.091506 [DEBUG] apt_consumer_task.c:141 () Wait for
> Messages [MRCP Client]
> 2019-01-30 04:28:20.091506 [DEBUG] mod_unimrcp.c:1061 (ASR-13) Terminating
> MRCP session
> 2019-01-30 04:28:20.091506 [DEBUG] apt_task.c:265 () Signal Message to
> [MRCP
> Client] [0x7fed300afe10;4;0]
> 2019-01-30 04:28:20.091506 [DEBUG] apt_task.c:337 () Process Message [MRCP
> Client] [0x7fed300afe10;4;0]
> 2019-01-30 04:28:20.091506 [INFO] mrcp_client_session.c:387 (ASR-13)
> Receive
> App Request ASR-13 <new> [1]
> 2019-01-30 04:28:20.091506 [DEBUG] mrcp_client_session.c:1283 (ASR-13)
> Dispatch App Request ASR-13 <new> [1]
>
> Any ideas will be appreciated. Thanks
>
>
>
> --
> Sent from: http://freeswitch-users.2379917.n2.nabble.com/
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From alaadin.abd at gmail.com  Wed Jan 30 23:20:01 2019
From: alaadin.abd at gmail.com (Alaadin Abdurrahman)
Date: Thu, 31 Jan 2019 01:20:01 +0200
Subject: [Freeswitch-users] Debian 9 FreeSWITCH 1.8 Sangoma Card A102
Message-ID: <CAJVaC-tLVsdoOfqKWx4S47_Lt+37US2fbvWMuJX2+VLD-Mt1sw@mail.gmail.com>

Dears,
I have an issue when i try to compile and make freeswitch 1.8 with
mod_freetdm to use sangoma card.
please note that i followed every step in the sangoma wiki

 https://wiki.freepbx.org/display/PC/Telephony+Cards+for+FreeSWITCH

and i still get this error message

cc1: all warnings being treated as errors
Makefile:1209: recipe for target
'ftmod_sangoma_isdn_la-ftmod_sangoma_isdn.lo' failed
make[2]: *** [ftmod_sangoma_isdn_la-ftmod_sangoma_isdn.lo] Error 1
make[2]: Leaving directory '/usr/src/freeswitch/libs/freetdm'
Makefile:15: recipe for target '../libfreetdm.la' failed
make[1]: *** [../libfreetdm.la] Error 2
make[1]: Leaving directory '/usr/src/freeswitch/libs/freetdm/mod_freetdm'
../../../build/modmake.rules:89: recipe for target 'all' failed
make: *** [all] Error 1
root at debian:/usr/src/freeswitch/libs/freetdm/mod_freetdm#

best regards
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From chad at apartmentlines.com  Thu Jan 31 00:12:37 2019
From: chad at apartmentlines.com (Chad Phillips)
Date: Wed, 30 Jan 2019 16:12:37 -0800
Subject: [Freeswitch-users] Anybody want to go in on some Spanish GM voices
	recordings?
Message-ID: <CAJ0sghRn7RswtrMJ0hcS8P7Z14khRbvHscu-hwPh9NU7GbBG8w@mail.gmail.com>

My company is just getting ready to do a round of Spanish voice prompts
through GM voices. They have excellent talent, and charge reasonable rates
per word/phrase, but their setup fees are a bit on the steep side.

Anyone else with this same need want to join in our order to defer some of
the setup costs? I imagine we can pay the per word/phrase costs
individually, and split the setup fee.

Also, FreeSWITCH core team: are you interested in having some Spanish
prompts done? My company would cover the cost of some, if there weren't too
many, as a service to the community...

Chad
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