From loi.dangthanh at gmail.com  Thu Jun  1 02:27:26 2017
From: loi.dangthanh at gmail.com (=?UTF-8?B?TOG7o2kgxJDhurduZw==?=)
Date: Thu, 01 Jun 2017 02:27:26 +0000
Subject: [Freeswitch-users] testing 1234
In-Reply-To: <1b0c01d2da5d$011a7310$034f5930$@freeswitch.org>
References: <1aa001d2da4b$ab19c1c0$014d4540$@freeswitch.org>
 <CAEJMVkAc0v8JLUnoN0+H2vQwqgZrDBK=Zg7AzHjGKGxMNc-RUQ@mail.gmail.com>
 <1abd01d2da54$c34b84a0$49e28de0$@freeswitch.org>
 <592F04DE020000310000A092@mail.tedssupply.com>
 <CADzhjpSBFS44VB1vVHuyT2K9v-aiRUCrXYTRuU8VKa8HQWPs4Q@mail.gmail.com>
 <CAEeV61hJxg0N8LdYei_LncMNRRuqKumzQ=2FEtSy1az++S0hsg@mail.gmail.com>
 <1b0c01d2da5d$011a7310$034f5930$@freeswitch.org>
Message-ID: <CAGsrq_teQLvTsfWhLX=-vjnGK9+phXBc70GEpFedBTYUNAJVMA@mail.gmail.com>

what's up guys? FS mailing is dead for several days?
I didn't receive any mail until this ping-pong today.
rgds,
Loi Dang

On Thu, Jun 1, 2017 at 5:27 AM Ken Rice <krice at freeswitch.org> wrote:

> Thanks for replying everyone.
>
>
>
> And just a reminder for those that are still on the list and want to
> unsubscribe, or just want to manage their list membership, there is a link
> at the bottom of every email that comes across the list that you can click.
> It will allow you to unsubscribe, or even change you from individual emails
> on the list to the digest where you get 1 email a day with all the other
> emails aggregated up.
>
>
>
> Thanks Guys!
>
> Ken
>
>
>
>
>
>
>
> *From:* FreeSWITCH-users [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Matt Broad
> *Sent:* Wednesday, May 31, 2017 5:20 PM
>
>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Subject:* Re: [Freeswitch-users] testing 1234
>
>
>
> roger roger
>
>
> Matt Broad
>
> Tel: +44 (0)203 011 1313 <+44%2020%203011%201313>
>
> Web: www.supportedbusiness.com
>
>
>
> On 31 May 2017 at 23:15, John Dalrymple <john at mobex.biz> wrote:
>
> unsubscribe (please)
>
>
> John Dalrymple
>
> President
>
> 813-990-0996 <(813)%20990-0996> call or text
>
>
>
>
>
>
>
>
>
> On Wed, May 31, 2017 at 6:01 PM, admin <admin at tedssupply.com> wrote:
>
> Loud and clear!
>
>
>
> >>> "Ken Rice" <krice at freeswitch.org> 05/31/17 5:33 PM >>>
>
> You got this?
>
>
>
> *From:* FreeSWITCH-users [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Brian West
> *Sent:* Wednesday, May 31, 2017 3:26 PM
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Subject:* Re: [Freeswitch-users] testing 1234
>
>
>
> I got this.
>
>
>
> /b
>
>
>
>
>
> On Wed, May 31, 2017 at 3:22 PM, Ken Rice <krice at freeswitch.org> wrote:
>
> This is just a test
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org
>
>
>
>
>
> --
>
> *Brian West*
> brian at freeswitch.org
>
> *Twitter: @FreeSWITCH , @briankwest*
>
> http://www.freeswitchbook.com
> http://www.freeswitchcookbook.com
>
> Book a phone call (CST) <https://freeswitch.com/appointment>
>
> Allison prompts for FreeSWITCH:
>
> *https://www.gofundme.com/allison-prompts-for-freeswitch*
> <https://www.gofundme.com/allison-prompts-for-freeswitch>
>
> Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
> /r/freeswitch <https://www.reddit.com/r/freeswitch>
>
> *T:*+19184209001 <(918)%20420-9001> | *F:*+19184209002 <(918)%20420-9002>
> | *M:*+1918424WEST (9378)
> *Skype:*briankwest
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170601/2d0c5f31/attachment.html>

From brian at freeswitch.org  Thu Jun  1 03:19:36 2017
From: brian at freeswitch.org (Brian West)
Date: Thu, 01 Jun 2017 03:19:36 +0000
Subject: [Freeswitch-users] testing 1234
In-Reply-To: <CAGsrq_teQLvTsfWhLX=-vjnGK9+phXBc70GEpFedBTYUNAJVMA@mail.gmail.com>
References: <1aa001d2da4b$ab19c1c0$014d4540$@freeswitch.org>
 <CAEJMVkAc0v8JLUnoN0+H2vQwqgZrDBK=Zg7AzHjGKGxMNc-RUQ@mail.gmail.com>
 <1abd01d2da54$c34b84a0$49e28de0$@freeswitch.org>
 <592F04DE020000310000A092@mail.tedssupply.com>
 <CADzhjpSBFS44VB1vVHuyT2K9v-aiRUCrXYTRuU8VKa8HQWPs4Q@mail.gmail.com>
 <CAEeV61hJxg0N8LdYei_LncMNRRuqKumzQ=2FEtSy1az++S0hsg@mail.gmail.com>
 <1b0c01d2da5d$011a7310$034f5930$@freeswitch.org>
 <CAGsrq_teQLvTsfWhLX=-vjnGK9+phXBc70GEpFedBTYUNAJVMA@mail.gmail.com>
Message-ID: <CAEJMVkDx5yOtjg62M2j_m+5n3efN0zDxeG3-np74e64tPXpkqw@mail.gmail.com>

We are doing upgrades, a small snafu caused this!

Hopefully we can prevent these types of human errors in the future.

/b


On Wed, May 31, 2017 at 9:30 PM Lợi Đặng <loi.dangthanh at gmail.com> wrote:

> what's up guys? FS mailing is dead for several days?
> I didn't receive any mail until this ping-pong today.
> rgds,
> Loi Dang
>
> On Thu, Jun 1, 2017 at 5:27 AM Ken Rice <krice at freeswitch.org> wrote:
>
>> Thanks for replying everyone.
>>
>>
>>
>> And just a reminder for those that are still on the list and want to
>> unsubscribe, or just want to manage their list membership, there is a link
>> at the bottom of every email that comes across the list that you can click.
>> It will allow you to unsubscribe, or even change you from individual emails
>> on the list to the digest where you get 1 email a day with all the other
>> emails aggregated up.
>>
>>
>>
>> Thanks Guys!
>>
>> Ken
>>
>>
>>
>>
>>
>>
>>
>> *From:* FreeSWITCH-users [mailto:
>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Matt Broad
>> *Sent:* Wednesday, May 31, 2017 5:20 PM
>>
>>
>> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> *Subject:* Re: [Freeswitch-users] testing 1234
>>
>>
>>
>> roger roger
>>
>>
>> Matt Broad
>>
>> Tel: +44 (0)203 011 1313 <+44%2020%203011%201313>
>>
>> Web: www.supportedbusiness.com
>>
>>
>>
>> On 31 May 2017 at 23:15, John Dalrymple <john at mobex.biz> wrote:
>>
>> unsubscribe (please)
>>
>>
>> John Dalrymple
>>
>> President
>>
>> 813-990-0996 <(813)%20990-0996> call or text
>>
>>
>>
>>
>>
>>
>>
>>
>>
>> On Wed, May 31, 2017 at 6:01 PM, admin <admin at tedssupply.com> wrote:
>>
>> Loud and clear!
>>
>>
>>
>> >>> "Ken Rice" <krice at freeswitch.org> 05/31/17 5:33 PM >>>
>>
>> You got this?
>>
>>
>>
>> *From:* FreeSWITCH-users [mailto:
>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Brian West
>> *Sent:* Wednesday, May 31, 2017 3:26 PM
>> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> *Subject:* Re: [Freeswitch-users] testing 1234
>>
>>
>>
>> I got this.
>>
>>
>>
>> /b
>>
>>
>>
>>
>>
>> On Wed, May 31, 2017 at 3:22 PM, Ken Rice <krice at freeswitch.org> wrote:
>>
>> This is just a test
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.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
>> http://www.freeswitch.org
>>
>>
>>
>>
>>
>> --
>>
>> *Brian West*
>> brian at freeswitch.org
>>
>> *Twitter: @FreeSWITCH , @briankwest*
>>
>> http://www.freeswitchbook.com
>> http://www.freeswitchcookbook.com
>>
>> Book a phone call (CST) <https://freeswitch.com/appointment>
>>
>> Allison prompts for FreeSWITCH:
>>
>> *https://www.gofundme.com/allison-prompts-for-freeswitch*
>> <https://www.gofundme.com/allison-prompts-for-freeswitch>
>>
>> Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
>> /r/freeswitch <https://www.reddit.com/r/freeswitch>
>>
>> *T:*+19184209001 <(918)%20420-9001> | *F:*+19184209002 <(918)%20420-9002>
>> | *M:*+1918424WEST (9378)
>> *Skype:*briankwest
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.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
>> http://www.freeswitch.org
>>
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.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
>> http://www.freeswitch.org
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.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
>> http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org

-- 

*Brian West*
brian at freeswitch.org

*Twitter: @FreeSWITCH , @briankwest*

http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

Book a phone call (CST) <https://freeswitch.com/appointment>

Allison prompts for FreeSWITCH:

*https://www.gofundme.com/allison-prompts-for-freeswitch*
<https://www.gofundme.com/allison-prompts-for-freeswitch>

Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
/r/freeswitch <https://www.reddit.com/r/freeswitch>

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*Skype:*briankwest
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170601/45d256eb/attachment-0001.html>

From fvillarroel at yahoo.com  Thu Jun  1 03:51:31 2017
From: fvillarroel at yahoo.com (FERNANDO VILLARROEL)
Date: Thu, 1 Jun 2017 03:51:31 +0000 (UTC)
Subject: [Freeswitch-users] FS 1.6 Video
References: <990207077.4430.1496289091672.ref@mail.yahoo.com>
Message-ID: <990207077.4430.1496289091672@mail.yahoo.com>

Dear All.
I am follow https://freeswitch.org/confluence/display/FREESWITCH/Debian+8+Jessie
But not works for me i received:
root at fswebrtc:/etc/apt# apt-get install -y --force-yes freeswitch-video-deps-mostLeyendo lista de paquetes... HechoCreando árbol de dependenciasLeyendo la información de estado... HechoNo se pudieron instalar algunos paquetes. Esto puede significar queusted pidió una situación imposible o, si está usando la distribucióninestable, que algunos paquetes necesarios aún no se han creado o sehan sacado de «Incoming».La siguiente información puede ayudar a resolver la situación:
Los siguientes paquetes tienen dependencias incumplidas: freeswitch-video-deps-most : Depende: libavcodec-extra pero no va a instalarseE: No se pudieron corregir los problemas, usted ha retenido paquetes rotos.

My /etc/apt/sources.list.d/freeswitch.list
deb http://files.freeswitch.org/repo/deb/debian/ jessie maindeb http://files.freeswitch.org/repo/deb/debian-unstable/ jessie main
uname -aLinux fswebrtc 3.16.0-4-amd64 #1 SMP Debian 3.16.7-ckt20-1+deb8u3 (2016-01-17) x86_64 GNU/Linux
root at fswebrtc:/etc/apt# grep -r deb /etc/apt/
/etc/apt/apt.conf.d/70debconf:// Pre-configure all packages with debconf before they are installed./etc/apt/apt.conf.d/50unattended-upgrades://     site          (eg, "http.debian.net")/etc/apt/apt.conf.d/50unattended-upgrades:// "apt-cache policy", and can be debugged by running/etc/apt/apt.conf.d/50unattended-upgrades:// derived from /etc/debian_version:/etc/apt/sources.list:# deb cdrom:[Debian GNU/Linux 8.3.0 _Jessie_ - Official amd64 NETINST Binary-1 20160123-18:59]/ jessie main/etc/apt/sources.list:#deb cdrom:[Debian GNU/Linux 8.3.0 _Jessie_ - Official amd64 NETINST Binary-1 20160123-18:59]/ jessie main/etc/apt/sources.list:deb http://http.debian.net/debian jessie main/etc/apt/sources.list:deb-src http://http.debian.net/debian jessie main/etc/apt/sources.list:deb http://security.debian.org/ jessie/updates main/etc/apt/sources.list:deb-src http://security.debian.org/ jessie/updates main/etc/apt/sources.list:deb http://http.debian.net/debian jessie-updates main/etc/apt/sources.list:deb-src http://http.debian.net/debian jessie-updates main/etc/apt/sources.list:deb http://www.deb-multimedia.org jessie main non-freeCoincidencia en el fichero binario /etc/apt/trusted.gpg.d/debian-archive-jessie-stable.gpgCoincidencia en el fichero binario /etc/apt/trusted.gpg.d/debian-archive-squeeze-stable.gpgCoincidencia en el fichero binario /etc/apt/trusted.gpg.d/debian-archive-jessie-security-automatic.gpgCoincidencia en el fichero binario /etc/apt/trusted.gpg.d/debian-archive-squeeze-automatic.gpgCoincidencia en el fichero binario /etc/apt/trusted.gpg.d/deb-multimedia-keyring.gpgCoincidencia en el fichero binario /etc/apt/trusted.gpg.d/debian-archive-wheezy-stable.gpgCoincidencia en el fichero binario /etc/apt/trusted.gpg.d/debian-archive-jessie-automatic.gpgCoincidencia en el fichero binario /etc/apt/trusted.gpg.d/debian-archive-wheezy-automatic.gpg/etc/apt/sources.list.d/freeswitch.list:deb http://files.freeswitch.org/repo/deb/debian/ jessie main/etc/apt/sources.list.d/freeswitch.list:deb http://files.freeswitch.org/repo/deb/debian-unstable/ jessie main
I appreciate some help or tips please.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170601/350f30c0/attachment.html>

From tculjaga at gmail.com  Thu Jun  1 06:46:33 2017
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Thu, 1 Jun 2017 08:46:33 +0200
Subject: [Freeswitch-users] originated_legs
In-Reply-To: <CAJoKKdWkSO0z1AKs_pze44hrocE+OWQrfrOOu3doU0Jf19KROg@mail.gmail.com>
References: <CAJoKKdWkSO0z1AKs_pze44hrocE+OWQrfrOOu3doU0Jf19KROg@mail.gmail.com>
Message-ID: <CAJoKKdUWc6f9eAhTe8iNmYED2HoLopdku2EV6JPKDXoHkVQWhw@mail.gmail.com>

anyone ? :=)

On 30 May 2017 at 14:40, Tihomir Culjaga <tculjaga at gmail.com> wrote:

> Hello,
>
> There is something that is bugging me pretty hard and i need to understand
> how variable_originated_legs in CHANNEL_HANGUP gets populated ?
>
>
> example:
>
> variable_originated_legs: ARRAY::faf71922-4526-11e7-b0f4-9fb65fdc7341;Outbound
> Call;888|:faf71922-4526-11e7-b0f4-9fb65fdc7341;Outbound
> Call;888|:faf76c56-4526-11e7-b0fd-9fb65fdc7341;Outbound Call;0916331550",
>
>
>
>
> the reason :=)
>
> i have a scenario like this:
>
>
> 1. Incoming calls (IN_CALL) are sent to PARK application
> 2. On CHANNEL_PARK i originate a call to an extension say EXT1
>
> e.g. originate {ogirination_uuid=<ext1_uuid>}user/1002 & park() ... and
> eventually i do uuid_bridge <in_call_uuid> <ext1_uuid>
>
>
> 3. EXT1 makes a call transfer to EXT2 (using att_xfer) and hangs up the
> call
> 4. on CHANNEL_HANGUP from EXT1 i learn EXT2 uuid and.,,, IN_CALL is
> talking to EXT2 and we are happy :=)
>
>
> ... this is all good if the transfer destination is a single extension (i
> get enough info from signal_bond etc...). But if i have multiple
> destination behind EXT2 e.g. user/1002,user/1003,sofia/gateway/gw1/012345
> the thing gets complicated.
>
> I was hoping i could exploit variable_originated_leg in CHANNEL_HANGUP
> from EXT1 to learn all origination legs uuid  after an att_xfer to a
> multiple destination.
>
> is that feasible ? I mean, will this variable contain all UUIDs att_xfer
> generated on a multi-destination transfer ?
>
>
>
> please i would appreciate any heads up here :=)
>
> thanks,
> Tihomir.
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170601/40c0bca1/attachment.html>

From alexdruzhilov at gmail.com  Thu Jun  1 09:08:27 2017
From: alexdruzhilov at gmail.com (=?UTF-8?B?0JDQu9C10LrRgdCw0L3QtNGAINCU0YDRg9C20LjQu9C+0LI=?=)
Date: Thu, 1 Jun 2017 12:08:27 +0300
Subject: [Freeswitch-users] Video conference with transcode mode is not
	working more than for 5-6 users
Message-ID: <CAMfvzkgEjyb7mD26VtuMX+Uwqzqddb32ANYT7Xj3dY8g3jZHiQ@mail.gmail.com>

OS: CentOS 7.2.1511
Freeswitch: 1.6.17

Steps:
1) create video conference (mod_sofia and conference-mode = transcode) with
one video stream from conference owner
2) add 4-6 members in this conference who will receive owner's video stream
3) everyone in this conference will see dramatic degradation of video
stream (incoming video stream bitrate falls from 1 Mbps to 0.5 Mbps, frame
rate falls down to 10 fps from 30 fps, lag between video and audio stream
appears)

But I don't see any problems with CPU, memory or network. So does anybody
knows whether it is an issue or it's how freeswitch 'transcode' mode should
works? And what to do to tune performance of this mode?
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170601/1bf3e74a/attachment.html>

From dist.lists at gmail.com  Thu Jun  1 10:55:55 2017
From: dist.lists at gmail.com (Dist Lists)
Date: Thu, 1 Jun 2017 12:55:55 +0200
Subject: [Freeswitch-users] replace b-leg in a bridge on 302 moved
	temporarily
Message-ID: <CAF5NwdNb8+RJMmReFoxo7eoijR16uAFpaaYYTHW-8sRD-qFyaA@mail.gmail.com>

Hi,

I am using ESL for my dial plan and I have the following scenario.

Leg-a is being bridged with the bridge() app to multiple endpoints
(b-legs). One of the b-legs sends 302 Moved Temporarily back. What I'd like
to do is to process the new destination number from the 302, because it may
be mapped to multiple endpoints and after figuring out the additional b-leg
endpoints to initiate calls to them, while at same time also keeping the
original (already ringing) b-legs from the initial call. Obviously on
answer the wining b-leg needs to be bridged to the a-leg, regardless if it
is an "original" b-leg or a "new" one.

To put it another way, I'd like to somehow "replace" the b-leg that sent
the 302 Moved Temporarily with multiple new endpoints (b-legs), while also
keeping the rest of the originally called b-legs. Sort of adding additional
endpoints via ESL to the bridge() application after the fact.

And the whole setup needs to work with bypass_media.

There is more to the story like the need to check if the endpoint is
authorized to redirect the call to the new destination and so on, but I'm
skipping this for the sake of simplicity. I have been trying to find a
solution or a workaround for a few days now, but nothing worked. What I
tested so far:

- letting Freeswith handle the 302 within the stack doesn't work, because I
cannot handle the mapping of the destination number to multiple endpoints.

- manual-redirect, which transfers the a-leg in the redirect context, but
also cancels all the other b-legs, didn't help either.

- outbound_redirect_fatal=true leaves the other b-legs ringing, but I
cannot "add" new b-legs to the bridge() app.

- originate-ing the b-legs via API and then trying to somehow link them
within the dial plan to the a-leg. This kind of works if I use api_on_ring,
api_on_media, etc to pass progress to the a-leg, but it feels too fragile.
Most importantly I couldn't figure out how to make bypass_media work, since
the a-leg and the b-legs are only loosely linked to one another via the
dial plan, but Freeswitch itself doesn't know anything about this relation.
I even tied setting some variables like originate_signal_bond and
originator, but it didn't help either.

- loopback endpoints, which I tried hoping that the loopback a-leg, which
can be transferred in a new context after a 302, will allow me to call
multiple new endpoint. Unfortunately this doesn't seem to work with
bypass_media.

I am out of ideas what to try next and was hoping for some feedback from
the list. It looks like what I'm trying to achieve is not possible with the
current set of features. If that is indeed the case, what would be the best
approach to add this functionality? Maybe by extending an existing
application/api?
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170601/17a3e027/attachment-0001.html>

From Paul.Mateer at outlook.com  Thu Jun  1 12:20:34 2017
From: Paul.Mateer at outlook.com (Paul Mateer)
Date: Thu, 1 Jun 2017 12:20:34 +0000
Subject: [Freeswitch-users] Embedding FreeSWITCH
Message-ID: <AM5PR10MB0708F41310A475F50E05C2EFF0F60@AM5PR10MB0708.EURPRD10.PROD.OUTLOOK.COM>

Guys, I'm looking to embed FreeSWITCH into a piece of Windows software which is extendable through the use of add-ins (written in C#).

When installed the software has an add-ins folder containing subfolders for each add-in to be loaded.


By default FreeSWITCH would expect the base directory to be that of the executable, but in this case I want it to be down in the add-ins subfolder since it makes more sense to have it with the add-in rather than the application. I figured that this could be solved easily enough by adding the following line of code before the call to switch_core_init():

freeswitch.switch_core_init((uint)flags, switch_bool_t.SWITCH_FALSE, out err);

However doing this seems to cause a problem when trying to load certain modules (in particular mod_sofia.dll). This seems to be because the new base directory that I have specified is not included in the alternate search path (set using SetDefaultDllDirectories and AddDllDirectory) so when a call is made to switch_dso_open to load the module it doesn't search that directory for modules such as pthread.dll. Now I could call AddDllDirectory as part of my freeSWITCH initialization code and then call RemoveDllDirectory when shutting down, but I was wondering if there was a possible justification for this being done in the CSharp_switch_directories_base_dir_set method of freeswitch_wrap.cxx?


Interestingly there is also a bug in the switch_dso_open method as the second call to LoadLibraryEx doesn't capture the return value in the lib variable, so even if the second call to LoadLibraryEx succeeds switch_dso_open will return NULL.



Paul

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170601/4088f303/attachment.html>

From rick at magicmail.mooo.com  Thu Jun  1 12:20:41 2017
From: rick at magicmail.mooo.com (Rick Jarvis)
Date: Thu, 1 Jun 2017 13:20:41 +0100
Subject: [Freeswitch-users] FS to FS
In-Reply-To: <5BD3EEFB-4F3C-4E14-B80C-1CED60244C5E@magicmail.mooo.com>
References: <5BD3EEFB-4F3C-4E14-B80C-1CED60244C5E@magicmail.mooo.com>
Message-ID: <C8FEF9C2-9138-40C5-9822-66CFC0756DDA@magicmail.mooo.com>

Apologies if you’ve seen this before, but not sure if anyone got it due to the mail issues and I’m still scratching my hairless head ;) ...

> On 31 May 2017, at 14:57, Rick Jarvis <rick at magicmail.mooo.com> wrote:
> 
> Sorry if this is a dumb question, but it’s something I’ve always struggled with that I’m convinced should be quite straightforward.
> 
> Getting one FS to register with another as an extension (I know there are other ways of getting FS to talk together, but this is the situation I’m faced with).
> 
> Box one (the host) is not behind NAT.
> Box two (the client) IS behind NAT.
> 
> In my mind it should be good enough to have a directory entry on box one, and a SIP profile on box two, then something like a catchall dial plan entry on box 2 to accept the calls made to the ‘extension’ it’s registered as and handle the call accordingly. But I don’t think it’s that simple?
> 
> I’ve looked through the soft phone configuration, which suggests using a directory entry with the register flag set to true, but this doesn’t seem to help me.
> 
> Thanks
> R



From Alexander.Haugg at c4b.de  Thu Jun  1 13:21:57 2017
From: Alexander.Haugg at c4b.de (Alexander Haugg)
Date: Thu, 1 Jun 2017 13:21:57 +0000
Subject: [Freeswitch-users] I get no Emails from the user list!
Message-ID: <42ddf3d866e348108134d659dfb04e8b@c4b.de>

Hi,

for serveral days i get no emails from the user list.
My account settings are ok.

My local system is checked.

What could be the problem?

Thanks!
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170601/5b7248ce/attachment.html>

From Alexander.Haugg at c4b.de  Thu Jun  1 13:24:20 2017
From: Alexander.Haugg at c4b.de (Alexander Haugg)
Date: Thu, 1 Jun 2017 13:24:20 +0000
Subject: [Freeswitch-users] WG: Turn configuration mod_sofia
Message-ID: <e4ed29d5806f49f7a953956e980c5782@c4b.de>

Hi Brian,

we use the Freeswitch as "Man in the Middle" for WebRTC.

The WebRTC Clients (ICE Link Stack from Frozen Mountain) registered via SIP on the Freeswitch.
The SIP signalling is a special szenario and works successfully in several LAN WAN setups.
But now we need srflx (that's fine with teh STUN configuration) an relay candidates in the SDP that's ganerated by the Freeswitch.

My think is, for this case i need the possibility to configure the turn server with credentials in the sip profile like the stun configuration too.

-          STUN configuration "<param name="ext-rtp-ip" value=" stun:stun.freeswitch.org"/>"

-          ICE configuration could be? "<param name="ext-rtp-ip" value=" { 'stun:stun.freeswitch.org', 'turn:turn.myserver.com;username=xxx;password=123'}"/>"

Thanks a lot


>From brian at freeswitch.org  Tue May 30 13:50:32 2017
From: brian at freeswitch.org (Brian West)
Date: Tue, 30 May 2017 08:50:32 -0500
Subject: [Freeswitch-users] Turn configuration mod_sofia
In-Reply-To: <a9e067a18e104afbb53e207712467baf at c4b.de<mailto:a9e067a18e104afbb53e207712467baf at c4b.de>>
References: <a9e067a18e104afbb53e207712467baf at c4b.de<mailto:a9e067a18e104afbb53e207712467baf at c4b.de>>
Message-ID: <CAEJMVkDy5e2COC8cH=xV=ozM0gepBS3cQ_YxUPQk4twMoaSVmg at mail.gmail.com<mailto:CAEJMVkDy5e2COC8cH=xV=ozM0gepBS3cQ_YxUPQk4twMoaSVmg at mail.gmail.com>>

There is little reason to use TURN when speaking to FreeSWITCH, What issue
are you trying to solve?


Von: Alexander Haugg
Gesendet: Montag, 29. Mai 2017 18:25
An: 'freeswitch-users at lists.freeswitch.org' <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
Betreff: Turn configuration mod_sofia

Hi all,

the stun configuration for the mod_sofia profile is very easy "<param name="ext-rtp-ip" value=" stun:stun.freeswitch.org"/>"
but what is to do, if i need a relay candidate in the sdp? How can i set the turn address and the login credentials?

Thanks a lot
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170601/49e4a9e0/attachment-0001.html>

From brian at freeswitch.org  Thu Jun  1 14:02:28 2017
From: brian at freeswitch.org (Brian West)
Date: Thu, 1 Jun 2017 09:02:28 -0500
Subject: [Freeswitch-users] FS to FS
In-Reply-To: <C8FEF9C2-9138-40C5-9822-66CFC0756DDA@magicmail.mooo.com>
References: <5BD3EEFB-4F3C-4E14-B80C-1CED60244C5E@magicmail.mooo.com>
 <C8FEF9C2-9138-40C5-9822-66CFC0756DDA@magicmail.mooo.com>
Message-ID: <CAEJMVkC2dWiN73Uqz-B11okrOigXtYNWZveAsHAUsm3+==3JQA@mail.gmail.com>

Its no different than setting up a gateway to an ITSP, you'll setup a user
in the directory on FS-A, then have FS-B register to it.  You could also
skip that completely and do an ACL list to authenticate the two systems
against each other.

/b


On Thu, Jun 1, 2017 at 7:20 AM, Rick Jarvis <rick at magicmail.mooo.com> wrote:

> Apologies if you’ve seen this before, but not sure if anyone got it due to
> the mail issues and I’m still scratching my hairless head ;) ...
>
> > On 31 May 2017, at 14:57, Rick Jarvis <rick at magicmail.mooo.com> wrote:
> >
> > Sorry if this is a dumb question, but it’s something I’ve always
> struggled with that I’m convinced should be quite straightforward.
> >
> > Getting one FS to register with another as an extension (I know there
> are other ways of getting FS to talk together, but this is the situation
> I’m faced with).
> >
> > Box one (the host) is not behind NAT.
> > Box two (the client) IS behind NAT.
> >
> > In my mind it should be good enough to have a directory entry on box
> one, and a SIP profile on box two, then something like a catchall dial plan
> entry on box 2 to accept the calls made to the ‘extension’ it’s registered
> as and handle the call accordingly. But I don’t think it’s that simple?
> >
> > I’ve looked through the soft phone configuration, which suggests using a
> directory entry with the register flag set to true, but this doesn’t seem
> to help me.
> >
> > Thanks
> > R
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org
>



-- 

*Brian West*
brian at freeswitch.org

*Twitter: @FreeSWITCH , @briankwest*

http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

Book a phone call (CST) <https://freeswitch.com/appointment>

Allison prompts for FreeSWITCH:

*https://www.gofundme.com/allison-prompts-for-freeswitch*
<https://www.gofundme.com/allison-prompts-for-freeswitch>

Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
/r/freeswitch <https://www.reddit.com/r/freeswitch>

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*Skype:*briankwest
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170601/baa68eff/attachment.html>

From brian at freeswitch.org  Thu Jun  1 17:04:22 2017
From: brian at freeswitch.org (Brian West)
Date: Thu, 1 Jun 2017 12:04:22 -0500
Subject: [Freeswitch-users] I get no Emails from the user list!
In-Reply-To: <42ddf3d866e348108134d659dfb04e8b@c4b.de>
References: <42ddf3d866e348108134d659dfb04e8b@c4b.de>
Message-ID: <CAEJMVkBcDTfH_T=SfNr17MgDTSFigZOm4zDokBP5tkr3zBhQ+g@mail.gmail.com>

Root cause, Migration of service, localhost was ::1 in /etc/hosts, ::1
wasn't on the relay list, but mail to anyone @freeswitch.org still received
email posts, so I failed to notice a problem.

/b


On Thu, Jun 1, 2017 at 8:21 AM, Alexander Haugg <Alexander.Haugg at c4b.de>
wrote:

> Hi,
>
>
>
> for serveral days i get no emails from the user list.
>
> My account settings are ok.
>
>
>
> My local system is checked.
>
>
>
> What could be the problem?
>
>
>
> Thanks!
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org
>



-- 

*Brian West*
brian at freeswitch.org

*Twitter: @FreeSWITCH , @briankwest*

http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

Book a phone call (CST) <https://freeswitch.com/appointment>

Allison prompts for FreeSWITCH:

*https://www.gofundme.com/allison-prompts-for-freeswitch*
<https://www.gofundme.com/allison-prompts-for-freeswitch>

Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
/r/freeswitch <https://www.reddit.com/r/freeswitch>

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*Skype:*briankwest
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170601/2f36f978/attachment.html>

From gmaruzz at gmail.com  Thu Jun  1 17:14:05 2017
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Thu, 1 Jun 2017 19:14:05 +0200
Subject: [Freeswitch-users] FS 1.6 Video
In-Reply-To: <990207077.4430.1496289091672@mail.yahoo.com>
References: <990207077.4430.1496289091672.ref@mail.yahoo.com>
 <990207077.4430.1496289091672@mail.yahoo.com>
Message-ID: <CALXCt0qEQbOyptRzWRjae-th8m8t5=nu3jHkG3K-Rp6x6DdwwA@mail.gmail.com>

Fernando,

Be sure to follow the steps exactly.

Its important you "apt-get update" after adding the repository, and before
i stalling the packages.

Hope this helps
-giovanni

sent from mobile
cell: +39 347 266 56 18
Giovanni Maruzzelli
OpenTelecom.IT

On Jun 1, 2017 5:56 AM, "FERNANDO VILLARROEL" <fvillarroel at yahoo.com> wrote:

Dear All.

I am follow https://freeswitch.org/confluence/display/FREESWITCH/
Debian+8+Jessie

But not works for me i received:

root at fswebrtc:/etc/apt# apt-get install -y --force-yes
freeswitch-video-deps-most
Leyendo lista de paquetes... Hecho
Creando árbol de dependencias
Leyendo la información de estado... Hecho
No se pudieron instalar algunos paquetes. Esto puede significar que
usted pidió una situación imposible o, si está usando la distribución
inestable, que algunos paquetes necesarios aún no se han creado o se
han sacado de «Incoming».
La siguiente información puede ayudar a resolver la situación:

Los siguientes paquetes tienen dependencias incumplidas:
 freeswitch-video-deps-most : Depende: libavcodec-extra pero no va a
instalarse
E: No se pudieron corregir los problemas, usted ha retenido paquetes rotos.


My /etc/apt/sources.list.d/freeswitch.list

deb http://files.freeswitch.org/repo/deb/debian/ jessie main
deb http://files.freeswitch.org/repo/deb/debian-unstable/ jessie main

uname -a
Linux fswebrtc 3.16.0-4-amd64 #1 SMP Debian 3.16.7-ckt20-1+deb8u3
(2016-01-17) x86_64 GNU/Linux

root at fswebrtc:/etc/apt# grep -r deb /etc/apt/

/etc/apt/apt.conf.d/70debconf:// Pre-configure all packages with debconf
before they are installed.
/etc/apt/apt.conf.d/50unattended-upgrades://     site          (eg, "
http.debian.net")
/etc/apt/apt.conf.d/50unattended-upgrades:// "apt-cache policy", and can be
debugged by running
/etc/apt/apt.conf.d/50unattended-upgrades:// derived from
/etc/debian_version:
/etc/apt/sources.list:# deb cdrom:[Debian GNU/Linux 8.3.0 _Jessie_ -
Official amd64 NETINST Binary-1 20160123-18:59]/ jessie main
/etc/apt/sources.list:#deb cdrom:[Debian GNU/Linux 8.3.0 _Jessie_ -
Official amd64 NETINST Binary-1 20160123-18:59]/ jessie main
/etc/apt/sources.list:deb http://http.debian.net/debian jessie main
/etc/apt/sources.list:deb-src http://http.debian.net/debian jessie main
/etc/apt/sources.list:deb http://security.debian.org/ jessie/updates main
/etc/apt/sources.list:deb-src http://security.debian.org/ jessie/updates
main
/etc/apt/sources.list:deb http://http.debian.net/debian jessie-updates main
/etc/apt/sources.list:deb-src http://http.debian.net/debian jessie-updates
main
/etc/apt/sources.list:deb http://www.deb-multimedia.org jessie main non-free
Coincidencia en el fichero binario /etc/apt/trusted.gpg.d/debian-
archive-jessie-stable.gpg
Coincidencia en el fichero binario /etc/apt/trusted.gpg.d/debian-
archive-squeeze-stable.gpg
Coincidencia en el fichero binario /etc/apt/trusted.gpg.d/debian-
archive-jessie-security-automatic.gpg
Coincidencia en el fichero binario /etc/apt/trusted.gpg.d/debian-
archive-squeeze-automatic.gpg
Coincidencia en el fichero binario /etc/apt/trusted.gpg.d/deb-
multimedia-keyring.gpg
Coincidencia en el fichero binario /etc/apt/trusted.gpg.d/debian-
archive-wheezy-stable.gpg
Coincidencia en el fichero binario /etc/apt/trusted.gpg.d/debian-
archive-jessie-automatic.gpg
Coincidencia en el fichero binario /etc/apt/trusted.gpg.d/debian-
archive-wheezy-automatic.gpg
/etc/apt/sources.list.d/freeswitch.list:deb http://files.freeswitch.org/
repo/deb/debian/ jessie main
/etc/apt/sources.list.d/freeswitch.list:deb http://files.freeswitch.org/
repo/deb/debian-unstable/ jessie main

I appreciate some help or tips please.

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.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
http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170601/85adeace/attachment-0001.html>

From tculjaga at gmail.com  Fri Jun  2 05:23:04 2017
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Fri, 2 Jun 2017 07:23:04 +0200
Subject: [Freeswitch-users] testing 1234
In-Reply-To: <CAEJMVkDx5yOtjg62M2j_m+5n3efN0zDxeG3-np74e64tPXpkqw@mail.gmail.com>
References: <1aa001d2da4b$ab19c1c0$014d4540$@freeswitch.org>
 <CAEJMVkAc0v8JLUnoN0+H2vQwqgZrDBK=Zg7AzHjGKGxMNc-RUQ@mail.gmail.com>
 <1abd01d2da54$c34b84a0$49e28de0$@freeswitch.org>
 <592F04DE020000310000A092@mail.tedssupply.com>
 <CADzhjpSBFS44VB1vVHuyT2K9v-aiRUCrXYTRuU8VKa8HQWPs4Q@mail.gmail.com>
 <CAEeV61hJxg0N8LdYei_LncMNRRuqKumzQ=2FEtSy1az++S0hsg@mail.gmail.com>
 <1b0c01d2da5d$011a7310$034f5930$@freeswitch.org>
 <CAGsrq_teQLvTsfWhLX=-vjnGK9+phXBc70GEpFedBTYUNAJVMA@mail.gmail.com>
 <CAEJMVkDx5yOtjg62M2j_m+5n3efN0zDxeG3-np74e64tPXpkqw@mail.gmail.com>
Message-ID: <00C4FE9F-691C-4A29-9375-9BCE4CB172FB@gmail.com>

Still, not a single new email in the list... quite strange ;)

Sent from my iPhone

> On 1 Jun 2017, at 05:19, Brian West <brian at freeswitch.org> wrote:
> 
> We are doing upgrades, a small snafu caused this!
> 
> Hopefully we can prevent these types of human errors in the future.
> 
> /b
> 
> 
>> On Wed, May 31, 2017 at 9:30 PM Lợi Đặng <loi.dangthanh at gmail.com> wrote:
>> what's up guys? FS mailing is dead for several days?
>> I didn't receive any mail until this ping-pong today.
>> rgds,
>> Loi Dang
>> 
>>> On Thu, Jun 1, 2017 at 5:27 AM Ken Rice <krice at freeswitch.org> wrote:
>>> Thanks for replying everyone.
>>> 
>>>  
>>> 
>>> And just a reminder for those that are still on the list and want to unsubscribe, or just want to manage their list membership, there is a link at the bottom of every email that comes across the list that you can click. It will allow you to unsubscribe, or even change you from individual emails on the list to the digest where you get 1 email a day with all the other emails aggregated up.
>>> 
>>>  
>>> 
>>> Thanks Guys!
>>> 
>>> Ken
>>> 
>>>  
>>> 
>>>  
>>> 
>>>  
>>> 
>>> From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Matt Broad
>>> Sent: Wednesday, May 31, 2017 5:20 PM
>>> 
>>> 
>>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>> Subject: Re: [Freeswitch-users] testing 1234
>>> 
>>>  
>>> 
>>> roger roger
>>> 
>>> 
>>> 
>>> Matt Broad
>>> 
>>> Tel: +44 (0)203 011 1313
>>> 
>>> Web: www.supportedbusiness.com
>>> 
>>> 
>>> 
>>>  
>>> 
>>> On 31 May 2017 at 23:15, John Dalrymple <john at mobex.biz> wrote:
>>> 
>>> unsubscribe (please)
>>> 
>>> 
>>> 
>>> John Dalrymple
>>> 
>>> President
>>> 
>>> 813-990-0996 call or text
>>> 
>>>  
>>> 
>>> 
>>> 
>>>  
>>> 
>>>  
>>> 
>>>  
>>> 
>>> On Wed, May 31, 2017 at 6:01 PM, admin <admin at tedssupply.com> wrote:
>>> 
>>> Loud and clear!
>>> 
>>>  
>>> 
>>> >>> "Ken Rice" <krice at freeswitch.org> 05/31/17 5:33 PM >>>
>>> 
>>> You got this?
>>> 
>>>  
>>> 
>>> From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West
>>> Sent: Wednesday, May 31, 2017 3:26 PM
>>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>> Subject: Re: [Freeswitch-users] testing 1234
>>> 
>>>  
>>> 
>>> I got this.
>>> 
>>>  
>>> 
>>> /b
>>> 
>>>  
>>> 
>>>  
>>> 
>>> On Wed, May 31, 2017 at 3:22 PM, Ken Rice <krice at freeswitch.org> wrote:
>>> 
>>> This is just a test
>>> 
>>> 
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.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
>>> http://www.freeswitch.org
>>> 
>>>  
>>> 
>>>  
>>> 
>>> --
>>> 
>>> Brian West
>>> brian at freeswitch.org
>>> 
>>> Twitter: @FreeSWITCH , @briankwest
>>> 
>>> http://www.freeswitchbook.com 
>>> http://www.freeswitchcookbook.com
>>> 
>>> Book a phone call (CST)
>>> 
>>> Allison prompts for FreeSWITCH:
>>> 
>>> https://www.gofundme.com/allison-prompts-for-freeswitch
>>> 
>>> Got Bugs? Report them here! | Reddit: /r/freeswitch
>>> 
>>> T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378)
>>> Skype:briankwest
>>> 
>>> 
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.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
>>> http://www.freeswitch.org
>>> 
>>>  
>>> 
>>> 
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.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
>>> http://www.freeswitch.org
>>> 
>>>  
>>> 
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.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
>>> http://www.freeswitch.org
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.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
>> http://www.freeswitch.org
> 
> -- 
> Brian West
> brian at freeswitch.org
> 
> Twitter: @FreeSWITCH , @briankwest
> 
> http://www.freeswitchbook.com 
> http://www.freeswitchcookbook.com
> 
> Book a phone call (CST)
> 
> Allison prompts for FreeSWITCH:
> 
> https://www.gofundme.com/allison-prompts-for-freeswitch
> 
> Got Bugs? Report them here! | Reddit: /r/freeswitch
> 
> T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378)
> Skype:briankwest
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170602/c8eae479/attachment-0001.html>

From gb at cm.nl  Fri Jun  2 08:54:12 2017
From: gb at cm.nl (Grant Bagdasarian)
Date: Fri, 2 Jun 2017 08:54:12 +0000
Subject: [Freeswitch-users] Multiple c params in SDP
Message-ID: <7d8d16ed5acb4ada8bc1753fcdae9384@cm.nl>

Hi,

We're currently facing an issue with one of our clients where they send two c parameters in the SDP.
Our Kamailio which is in front of the freeswitch communicates with RTPENGINE which in turn changes only the c parameters in the Media Description, not at Session Level.
Freeswitch however, uses the IP in the c param in the Session Description which causes the RTP stream to go directly to the client, instead of being bridged by the RTPENGINE.

Is there any way to force freeswitch to use the c param in the Media Description?

Regards,

Grant
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170602/e7d383b6/attachment.html>

From martin at maxnet.ao  Fri Jun  2 09:54:57 2017
From: martin at maxnet.ao (Martin Boese)
Date: Fri, 2 Jun 2017 10:54:57 +0100
Subject: [Freeswitch-users] Enabling new languages for Say
Message-ID: <20170602105457.68aa24f1@bones-tp>

Hi!

FreeSWITCH Version 1.6.15-32-bec4538~64bit (-32-bec4538 64bit)
Debian Jessie. Vanilla config.

I am trying to enable portuguese to say numbers using dptools "Say".

This is what I did:
 - Module mod_say_pt is loaded
 - in freeswitch.xml
   include lang/pt/pt_PT.xml
 - Downloaded sounds from
   https://github.com/jpawlowski/freeswitch-sounds-ttsand placed them
   into /usr/share/freeswitch/sounds
   Directory structure is now like: /pt/tts/google/ascii/16000/...

CLI> say_string t.wav pt NUMBER pronounced 123
[ERR] switch_xml.c:3180 Can't find phrases tag

..I found out that lang/pt_PT.xml seems to be missing the
<phrases><macros> tags within the <language> tag (vanilla config). I
fixed that.

Now:
CLI> say_string t.wav pt NUMBER pronounced 123
[ERR] switch_ivr.c:3726 Invalid SAY Interface [pt]!

BTW: English works fine:
CLI> say_string t.wav en NUMBER pronounced 123
file_string://digits/1.wav!digits/hundred.wav!digits/20.wav!digits/3.wav

I also tried other languages but have same error "Invalid SAY
Interface".

What am I missing. Please help.

Thanks,
Martin


From xxxman2008 at 126.com  Fri Jun  2 14:26:08 2017
From: xxxman2008 at 126.com (Raymond)
Date: Fri, 2 Jun 2017 22:26:08 +0800 (CST)
Subject: [Freeswitch-users] question about HA solution
In-Reply-To: <CADcrYG+RHGQ2fC7dSKK9x9WfVvYcTczOtbppCLgttRCfMGqArA@mail.gmail.com>
References: <CADcrYG+RHGQ2fC7dSKK9x9WfVvYcTczOtbppCLgttRCfMGqArA@mail.gmail.com>
Message-ID: <2628fa1d.b9a5.15c6932ffc0.Coremail.xxxman2008@126.com>

Hi, Denys


     Talking about HA ,it's complex , and have lots of detail.
     The "FS HA Solution" is very simple  . if you think deeper ,you will find more question,such as:
       A. if we have 5 servers in the group , what is the rule for transfer the calls ?
       B. Once the broken server have 1000 concurrent call ,  if  there's  new "performance issue"  when we transfer so many calls to a server.


So,  forgot your question , the "HA Solution" is just a demo . It need more work to use for business .


Raymond






在 2017-06-01 06:28:42，"Denys Pozniak" <denys.pozniak at crazycall.com> 写道：

Hello!


I built FS HA solution based on keepalived and mysql master-master.
It works ok generally, but as I understand FS after restarting cleaning own database.


So when node1 fails calls jump to node2, after script restarts node1 it is not possible to move calls back.


Tried options in switch.conf.xml, but no luck:


    <param name="auto-create-schemas" value="false"/>
    <param name="auto-clear-sql" value="false"/>


Is there is a way to solve this?


BR,
Denys



-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170602/38fbbc92/attachment.html>

From brian at freeswitch.org  Fri Jun  2 14:49:31 2017
From: brian at freeswitch.org (Brian West)
Date: Fri, 2 Jun 2017 09:49:31 -0500
Subject: [Freeswitch-users] Multiple c params in SDP
In-Reply-To: <7d8d16ed5acb4ada8bc1753fcdae9384@cm.nl>
References: <7d8d16ed5acb4ada8bc1753fcdae9384@cm.nl>
Message-ID: <CAEJMVkCCZv+=2NA97vc7+bb5oB3x=wkDTkSMHDvh7wV28becKA@mail.gmail.com>

I seen your post on reddit about this issue.  Can you show me traces of
what exactly its doing?

/b


On Fri, Jun 2, 2017 at 3:54 AM, Grant Bagdasarian <gb at cm.nl> wrote:

> Hi,
>
>
>
> We’re currently facing an issue with one of our clients where they send
> two c parameters in the SDP.
>
> Our Kamailio which is in front of the freeswitch communicates with
> RTPENGINE which in turn changes only the c parameters in the Media
> Description, not at Session Level.
>
> Freeswitch however, uses the IP in the c param in the Session Description
> which causes the RTP stream to go directly to the client, instead of being
> bridged by the RTPENGINE.
>
>
>
> Is there any way to force freeswitch to use the c param in the Media
> Description?
>
>
> Regards,
>
>
>
> Grant
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org
>



-- 

*Brian West*
brian at freeswitch.org

*Twitter: @FreeSWITCH , @briankwest*

http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

Book a phone call (CST) <https://freeswitch.com/appointment>

Allison prompts for FreeSWITCH:

*https://www.gofundme.com/allison-prompts-for-freeswitch*
<https://www.gofundme.com/allison-prompts-for-freeswitch>

Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
/r/freeswitch <https://www.reddit.com/r/freeswitch>

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*Skype:*briankwest
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170602/d60d6401/attachment.html>

From brian at freeswitch.org  Fri Jun  2 14:51:25 2017
From: brian at freeswitch.org (Brian West)
Date: Fri, 2 Jun 2017 09:51:25 -0500
Subject: [Freeswitch-users] Enabling new languages for Say
In-Reply-To: <20170602105457.68aa24f1@bones-tp>
References: <20170602105457.68aa24f1@bones-tp>
Message-ID: <CAEJMVkC2kGAZFeft1dX0GNyD5GZpfdn_To4=UGSVi_gS8iUbJg@mail.gmail.com>

I would guess mod_say_pt is not loaded.

On Fri, Jun 2, 2017 at 4:54 AM, Martin Boese <martin at maxnet.ao> wrote:

> Hi!
>
> FreeSWITCH Version 1.6.15-32-bec4538~64bit (-32-bec4538 64bit)
> Debian Jessie. Vanilla config.
>
> I am trying to enable portuguese to say numbers using dptools "Say".
>
> This is what I did:
>  - Module mod_say_pt is loaded
>  - in freeswitch.xml
>    include lang/pt/pt_PT.xml
>  - Downloaded sounds from
>    https://github.com/jpawlowski/freeswitch-sounds-ttsand placed them
>    into /usr/share/freeswitch/sounds
>    Directory structure is now like: /pt/tts/google/ascii/16000/...
>
> CLI> say_string t.wav pt NUMBER pronounced 123
> [ERR] switch_xml.c:3180 Can't find phrases tag
>
> ..I found out that lang/pt_PT.xml seems to be missing the
> <phrases><macros> tags within the <language> tag (vanilla config). I
> fixed that.
>
> Now:
> CLI> say_string t.wav pt NUMBER pronounced 123
> [ERR] switch_ivr.c:3726 Invalid SAY Interface [pt]!
>
> BTW: English works fine:
> CLI> say_string t.wav en NUMBER pronounced 123
> file_string://digits/1.wav!digits/hundred.wav!digits/20.wav!digits/3.wav
>
> I also tried other languages but have same error "Invalid SAY
> Interface".
>
> What am I missing. Please help.
>
> Thanks,
> Martin
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org




-- 

*Brian West*
brian at freeswitch.org

*Twitter: @FreeSWITCH , @briankwest*

http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

Book a phone call (CST) <https://freeswitch.com/appointment>

Allison prompts for FreeSWITCH:

*https://www.gofundme.com/allison-prompts-for-freeswitch*
<https://www.gofundme.com/allison-prompts-for-freeswitch>

Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
/r/freeswitch <https://www.reddit.com/r/freeswitch>

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*Skype:*briankwest
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170602/d72352fe/attachment.html>

From mjlopez at smartic.es  Thu Jun  1 16:00:38 2017
From: mjlopez at smartic.es (=?iso-8859-1?Q?Miguel_Jes=FAs_L=F3pez_Valverde?=)
Date: Thu, 1 Jun 2017 18:00:38 +0200
Subject: [Freeswitch-users] trouble with instalation of mod_fail2ban module
	in FreeSwitch.
Message-ID: <010a01d2daf0$36c6cc50$a45464f0$@smartic.es>

Hello:

 

I am sending you this email because I am having problems installing the
module mod_fail2ban following the recipe offered at
https://freeswitch.org/confluence/display/FREESWITCH/mod_fail2ban

because when I run make, I get this result:

 

/usr/local/freeswitch/mod/applications/mod_fail2ban$ sudo make

Makefile:2: ../../../../build/modmake.rules: No such file or directory

make: *** No rule to make target '../../../../build/modmake.rules'.  Stop.

 

Do you know in which folder the sentence “git clone” may to be executed?. Do
you know if there is anything else to keep in mind that can cause this
problem?

 

Thank you very much.

 

Miguel J. Lopez.



---
El software de antivirus Avast ha analizado este correo electrónico en busca de virus.
https://www.avast.com/antivirus
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170601/bc3ed69b/attachment-0001.html>

From luis.daniel.lucio at gmail.com  Fri Jun  2 15:30:03 2017
From: luis.daniel.lucio at gmail.com (Luis Daniel Lucio Quiroz)
Date: Fri, 2 Jun 2017 11:30:03 -0400
Subject: [Freeswitch-users] Not honoring ODBC in the core
Message-ID: <CAFLo2Qz569hQM35R_r+kbKOHrc3vcWmyQH_PbBxFbPCMnRrC-w@mail.gmail.com>

my FS 1.6.17 is not honoring my ODBC settings, it keeps creating core.db
and other sqlite db's. I can confirm ODBC works, isql command connects
without issues.

   <param name="core-db-dsn" value="freeswitch:USER:XXXXXXXXXXXX" />
   <param name="core-recovery-db-dsn" value="freeswitch:USER:XXXXXXXX" />

any ideas?

--
Luis Daniel Lucio Quiroz
CISSP, CISM, CISA
Linux, VoIP and much more fun
www.okay.com.mx

Need LCR? Check out LCR for FusionPBX with FreeSWITCH
Need Billing? Check out Billing for FusionPBX with FreeSWITCH
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170602/b1cb8cf8/attachment.html>

From infos at madovsky.org  Fri Jun  2 21:58:55 2017
From: infos at madovsky.org (Madovsky)
Date: Fri, 2 Jun 2017 14:58:55 -0700
Subject: [Freeswitch-users] trouble with instalation of mod_fail2ban
 module in FreeSwitch.
In-Reply-To: <010a01d2daf0$36c6cc50$a45464f0$@smartic.es>
References: <010a01d2daf0$36c6cc50$a45464f0$@smartic.es>
Message-ID: <52bacbc4-bc14-e9c9-d938-da1cc789f357@madovsky.org>


On 6/1/2017 9:00 AM, Miguel Jesús López Valverde wrote:
>
> Hello:
>
> I am sending you this email because I am having problems installing 
> the module mod_fail2ban following the recipe offered at 
> https://freeswitch.org/confluence/display/FREESWITCH/mod_fail2ban
>
> because when I run make, I get this result:
>
> /usr/local/freeswitch/mod/applications/mod_fail2ban$ sudo make
>
> Makefile:2: ../../../../build/modmake.rules: No such file or directory
>
> make: *** No rule to make target '../../../../build/modmake.rules'.  Stop.
>
> Do you know in which folder the sentence “git clone” may to be 
> executed?. Do you know if there is anything else to keep in mind that 
> can cause this problem?
>
> Thank you very much.
>
> Miguel J. Lopez.
>
>
> <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=emailclient> 
> 	Libre de virus. www.avast.com 
> <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=emailclient> 
>
>
> <#DAB4FAD8-2DD7-40BB-A1B8-4E2AA1F9FDF2>
>
>
did you install fail2ban from your distro?
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170602/bea610e4/attachment.html>

From mike at jerris.com  Fri Jun  2 22:07:59 2017
From: mike at jerris.com (Michael Jerris)
Date: Fri, 2 Jun 2017 18:07:59 -0400
Subject: [Freeswitch-users] Not honoring ODBC in the core
In-Reply-To: <CAFLo2Qz569hQM35R_r+kbKOHrc3vcWmyQH_PbBxFbPCMnRrC-w@mail.gmail.com>
References: <CAFLo2Qz569hQM35R_r+kbKOHrc3vcWmyQH_PbBxFbPCMnRrC-w@mail.gmail.com>
Message-ID: <CF8B652F-8C16-47AC-BFB7-DECC2ACBD532@jerris.com>

take a look at the logs to see why it can’t connect to the db.

> On Jun 2, 2017, at 11:30 AM, Luis Daniel Lucio Quiroz <luis.daniel.lucio at gmail.com> wrote:
> 
> my FS 1.6.17 is not honoring my ODBC settings, it keeps creating core.db and other sqlite db's. I can confirm ODBC works, isql command connects without issues.
> 
>    <param name="core-db-dsn" value="freeswitch:USER:XXXXXXXXXXXX" />
>    <param name="core-recovery-db-dsn" value="freeswitch:USER:XXXXXXXX" />
> 



From mike at jerris.com  Fri Jun  2 22:09:18 2017
From: mike at jerris.com (Michael Jerris)
Date: Fri, 2 Jun 2017 18:09:18 -0400
Subject: [Freeswitch-users] trouble with instalation of mod_fail2ban
 module in FreeSwitch.
In-Reply-To: <010a01d2daf0$36c6cc50$a45464f0$@smartic.es>
References: <010a01d2daf0$36c6cc50$a45464f0$@smartic.es>
Message-ID: <84E0A820-13D2-47D7-9F7C-5AB5FB0617AC@jerris.com>

This is a new module only available in the master branch

> On Jun 1, 2017, at 12:00 PM, Miguel Jesús López Valverde <mjlopez at smartic.es> wrote:
> 
> Hello:
>  
> I am sending you this email because I am having problems installing the module mod_fail2ban following the recipe offered at https://freeswitch.org/confluence/display/FREESWITCH/mod_fail2ban <https://freeswitch.org/confluence/display/FREESWITCH/mod_fail2ban>
> because when I run make, I get this result:
>  
> /usr/local/freeswitch/mod/applications/mod_fail2ban$ sudo make
> Makefile:2: ../../../../build/modmake.rules: No such file or directory
> make: *** No rule to make target '../../../../build/modmake.rules'.  Stop.
>  
> Do you know in which folder the sentence “git clone” may to be executed?. Do you know if there is anything else to keep in mind that can cause this problem?
>  

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170602/5d632fb0/attachment.html>

From xxxman2008 at 126.com  Sat Jun  3 05:58:08 2017
From: xxxman2008 at 126.com (Raymond)
Date: Sat, 3 Jun 2017 13:58:08 +0800 (CST)
Subject: [Freeswitch-users] question about HA solution
In-Reply-To: <CADcrYG+RHGQ2fC7dSKK9x9WfVvYcTczOtbppCLgttRCfMGqArA@mail.gmail.com>
References: <CADcrYG+RHGQ2fC7dSKK9x9WfVvYcTczOtbppCLgttRCfMGqArA@mail.gmail.com>
Message-ID: <4c2d3f0d.1f3e.15c6c8841fa.Coremail.xxxman2008@126.com>

Hi, Denys


     Talking about HA ,it's complex , and have lots of detail.
     The "FS HA Solution" is very simple  . if you think deeper ,you will find more question,such as:
       A. if we have 5 servers in the group , what is the rule for transfer the calls ?
       B. Once the broken server have 1000 concurrent call ,  if  there's  new "performance issue"  when we transfer so many calls to a server.


If it really need an answer about your question -- "if it is possible to move calls back".   I think it's unnecessary.  Every time when we transfer the calls from one server to another, there's  a little "audio dropped" , maybe , not good for user experience.


So,  forgot your question , the "HA Solution" is just a demo . It need more work to do  for product situation.


Raymond





在 2017-06-01 06:28:42，"Denys Pozniak" <denys.pozniak at crazycall.com> 写道：

Hello!


I built FS HA solution based on keepalived and mysql master-master.
It works ok generally, but as I understand FS after restarting cleaning own database.


So when node1 fails calls jump to node2, after script restarts node1 it is not possible to move calls back.


Tried options in switch.conf.xml, but no luck:


    <param name="auto-create-schemas" value="false"/>
    <param name="auto-clear-sql" value="false"/>


Is there is a way to solve this?


BR,
Denys








 
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170603/856c28b6/attachment-0001.html>

From donguyenha at gmail.com  Sat Jun  3 06:11:37 2017
From: donguyenha at gmail.com (Do Nguyen Ha)
Date: Sat, 3 Jun 2017 13:11:37 +0700
Subject: [Freeswitch-users] question about HA solution
In-Reply-To: <4c2d3f0d.1f3e.15c6c8841fa.Coremail.xxxman2008@126.com>
References: <CADcrYG+RHGQ2fC7dSKK9x9WfVvYcTczOtbppCLgttRCfMGqArA@mail.gmail.com>
 <4c2d3f0d.1f3e.15c6c8841fa.Coremail.xxxman2008@126.com>
Message-ID: <CACgE3xt4ur_cBRUoxVSBgCtUoevph0sejn9WAm7KF0WoA2Uzog@mail.gmail.com>

+1 for Raymond

On Jun 3, 2017 12:58, "Raymond" <xxxman2008 at 126.com> wrote:

> Hi, Denys
>
>      Talking about HA ,it's complex , and have lots of detail.
>      The "FS HA Solution" is very simple  . if you think deeper ,you will
> find more question,such as:
>        A. if we have 5 servers in the group , what is the rule for
> transfer the calls ?
>        B. Once the broken server have 1000 concurrent call ,  if  there's
>  new "performance issue"  when we transfer so many calls to a server.
>
> If it really need an answer about your question -- "if it is possible to
> move calls back".   I think it's unnecessary.  Every time when we transfer
> the calls from one server to another, there's  a little "audio dropped" ,
> maybe , not good for user experience.
>
> So,  forgot your question , the "HA Solution" is just a demo . It need
> more work to do  for product situation.
>
> Raymond
>
>
>
>
> 在 2017-06-01 06:28:42，"Denys Pozniak" <denys.pozniak at crazycall.com> 写道：
>
> Hello!
>
> I built FS HA solution based on keepalived and mysql master-master.
> It works ok generally, but as I understand FS after restarting cleaning
> own database.
>
> So when node1 fails calls jump to node2, after script restarts node1 it is
> not possible to move calls back.
>
> Tried options in switch.conf.xml, but no luck:
>
>     <param name="auto-create-schemas" value="false"/>
>     <param name="auto-clear-sql" value="false"/>
>
> Is there is a way to solve this?
>
> BR,
> Denys
>
>
>
>
>
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170603/17126fb0/attachment.html>

From alexdruzhilov at gmail.com  Sat Jun  3 07:35:24 2017
From: alexdruzhilov at gmail.com (=?UTF-8?B?0JDQu9C10LrRgdCw0L3QtNGAINCU0YDRg9C20LjQu9C+0LI=?=)
Date: Sat, 3 Jun 2017 10:35:24 +0300
Subject: [Freeswitch-users] Fwd: Video conference with transcode mode is not
 working more than for 5-6 users
In-Reply-To: <CAMfvzkgEjyb7mD26VtuMX+Uwqzqddb32ANYT7Xj3dY8g3jZHiQ@mail.gmail.com>
References: <CAMfvzkgEjyb7mD26VtuMX+Uwqzqddb32ANYT7Xj3dY8g3jZHiQ@mail.gmail.com>
Message-ID: <CAMfvzkiHwiJVRtaojTuUX-cFE50UvxuLUrq51MGUPW4ScjRokg@mail.gmail.com>

OS: CentOS 7.2.1511
Freeswitch: 1.6.17

Steps:
1) create video conference (mod_sofia and conference-mode = transcode) with
one video stream from conference owner
2) add 4-6 members in this conference who will receive owner's video stream
3) everyone in this conference will see dramatic degradation of video
stream (incoming video stream bitrate falls from 1 Mbps to 0.5 Mbps, frame
rate falls down to 10 fps from 30 fps, lag between video and audio stream
appears)

But I don't see any problems with CPU, memory or network. So does anybody
knows whether it is an issue or it's how freeswitch 'transcode' mode should
works? And what to do to tune performance of this mode?
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170603/8502e1d4/attachment.html>

From mike at jerris.com  Sat Jun  3 13:57:55 2017
From: mike at jerris.com (Michael Jerris)
Date: Sat, 03 Jun 2017 13:57:55 +0000
Subject: [Freeswitch-users] Fwd: Video conference with transcode mode is
 not working more than for 5-6 users
In-Reply-To: <CAMfvzkiHwiJVRtaojTuUX-cFE50UvxuLUrq51MGUPW4ScjRokg@mail.gmail.com>
References: <CAMfvzkgEjyb7mD26VtuMX+Uwqzqddb32ANYT7Xj3dY8g3jZHiQ@mail.gmail.com>
 <CAMfvzkiHwiJVRtaojTuUX-cFE50UvxuLUrq51MGUPW4ScjRokg@mail.gmail.com>
Message-ID: <CAEwTsAgJM2GT2BZWBCeVZskazkCRUdbrrRuZTsVA9JhSrHFPsg@mail.gmail.com>

We develop on debian 8.  It's worth trying there
to see if it's an issue with versions of things in centos.

On Sat, Jun 3, 2017 at 3:37 AM Александр Дружилов <alexdruzhilov at gmail.com>
wrote:

> OS: CentOS 7.2.1511
> Freeswitch: 1.6.17
>
> Steps:
> 1) create video conference (mod_sofia and conference-mode = transcode)
> with one video stream from conference owner
> 2) add 4-6 members in this conference who will receive owner's video stream
> 3) everyone in this conference will see dramatic degradation of video
> stream (incoming video stream bitrate falls from 1 Mbps to 0.5 Mbps, frame
> rate falls down to 10 fps from 30 fps, lag between video and audio stream
> appears)
>
> But I don't see any problems with CPU, memory or network. So does anybody
> knows whether it is an issue or it's how freeswitch 'transcode' mode should
> works? And what to do to tune performance of this mode?
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170603/1aa99aba/attachment.html>

From anthony.minessale at gmail.com  Sat Jun  3 16:26:46 2017
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 03 Jun 2017 16:26:46 +0000
Subject: [Freeswitch-users] Fwd: Video conference with transcode mode is
 not working more than for 5-6 users
In-Reply-To: <CAEwTsAgJM2GT2BZWBCeVZskazkCRUdbrrRuZTsVA9JhSrHFPsg@mail.gmail.com>
References: <CAMfvzkgEjyb7mD26VtuMX+Uwqzqddb32ANYT7Xj3dY8g3jZHiQ@mail.gmail.com>
 <CAMfvzkiHwiJVRtaojTuUX-cFE50UvxuLUrq51MGUPW4ScjRokg@mail.gmail.com>
 <CAEwTsAgJM2GT2BZWBCeVZskazkCRUdbrrRuZTsVA9JhSrHFPsg@mail.gmail.com>
Message-ID: <CAKbxfG_2V49OXQV9Gja-i2rceZQcovGBm=Yak+skTdy1KvGYGA@mail.gmail.com>

Mux mode with 1x1 layout and minimize-video-encoding flag is better


On Sat, Jun 3, 2017 at 8:59 AM Michael Jerris <mike at jerris.com> wrote:

> We develop on debian 8.  It's worth trying there
> to see if it's an issue with versions of things in centos.
>
> On Sat, Jun 3, 2017 at 3:37 AM Александр Дружилов <alexdruzhilov at gmail.com>
> wrote:
>
>> OS: CentOS 7.2.1511
>> Freeswitch: 1.6.17
>>
>> Steps:
>> 1) create video conference (mod_sofia and conference-mode = transcode)
>> with one video stream from conference owner
>> 2) add 4-6 members in this conference who will receive owner's video
>> stream
>> 3) everyone in this conference will see dramatic degradation of video
>> stream (incoming video stream bitrate falls from 1 Mbps to 0.5 Mbps, frame
>> rate falls down to 10 fps from 30 fps, lag between video and audio stream
>> appears)
>>
>> But I don't see any problems with CPU, memory or network. So does anybody
>> knows whether it is an issue or it's how freeswitch 'transcode' mode should
>> works? And what to do to tune performance of this mode?
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.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
>> http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org

-- 
Anthony Minessale II       ♬ @anthmfs  ♬ @FreeSWITCH  ♬

☞ http://freeswitch.org/  ☞ http://cluecon.com/  ☞
http://twitter.com/FreeSWITCH
☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+
<http://freeswitch.org/g+>*

ClueCon Weekly Development Call
☎ sip:888 at conference.freeswitch.org  ☎ +19193869900

https://www.youtube.com/watch?v=9XXgW34t40s
https://www.youtube.com/watch?v=NLaDpGQuZDA
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170603/8611f137/attachment.html>

From d.mordovin at dwide.com  Sun Jun  4 08:10:13 2017
From: d.mordovin at dwide.com (Dmitry Mordovin)
Date: Sun, 4 Jun 2017 11:10:13 +0300
Subject: [Freeswitch-users] DTMF events
Message-ID: <91ba4d3c-b0a4-a255-2b17-75d91b7767fb@dwide.com>

Hello!

I want make application which will listen DTMF events and when it fire, 
send DTMF digit to web-url.

For example, I use API for handle call state, execute_on_originate, 
execute_on_ring, execute_on_answer...

Does exists API for DTMF?

Anyone knows how can do it?


Best regards, Dmitry Mordovin



From luis.daniel.lucio at gmail.com  Sat Jun  3 20:16:49 2017
From: luis.daniel.lucio at gmail.com (Luis Daniel Lucio Quiroz)
Date: Sat, 3 Jun 2017 16:16:49 -0400
Subject: [Freeswitch-users] question about HA solution
In-Reply-To: <CADcrYG+RHGQ2fC7dSKK9x9WfVvYcTczOtbppCLgttRCfMGqArA@mail.gmail.com>
References: <CADcrYG+RHGQ2fC7dSKK9x9WfVvYcTczOtbppCLgttRCfMGqArA@mail.gmail.com>
Message-ID: <CAFLo2QxxbZ5Ged0J0p9_VfQ2nkGtuV0CDp9G763EjjuvqUv8iw@mail.gmail.com>

You may want to read this article.

http://inside-out.xyz/technology/how-to-configure-freeswitch-for-ha.html

Le 31 mai 2017 6:29 PM, "Denys Pozniak" <denys.pozniak at crazycall.com> a
écrit :

Hello!

I built FS HA solution based on keepalived and mysql master-master.
It works ok generally, but as I understand FS after restarting cleaning own
database.

So when node1 fails calls jump to node2, after script restarts node1 it is
not possible to move calls back.

Tried options in switch.conf.xml, but no luck:

    <param name="auto-create-schemas" value="false"/>
    <param name="auto-clear-sql" value="false"/>

Is there is a way to solve this?

BR,
Denys



_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.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
http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170603/d9a49406/attachment.html>

From luis.daniel.lucio at gmail.com  Sun Jun  4 00:25:34 2017
From: luis.daniel.lucio at gmail.com (Luis Daniel Lucio Quiroz)
Date: Sat, 3 Jun 2017 20:25:34 -0400
Subject: [Freeswitch-users] Not honoring ODBC in the core
In-Reply-To: <CF8B652F-8C16-47AC-BFB7-DECC2ACBD532@jerris.com>
References: <CAFLo2Qz569hQM35R_r+kbKOHrc3vcWmyQH_PbBxFbPCMnRrC-w@mail.gmail.com>
 <CF8B652F-8C16-47AC-BFB7-DECC2ACBD532@jerris.com>
Message-ID: <CAFLo2QzBSmcMunBb2g+CKCHZQ16MhbhvN66D-bPvSXTAJYk9gg@mail.gmail.com>

I have managed to move all modules to ODBC but the core. If it helps, this
deployment is using AWS RDS.

I have turned on all the debug options, and i get the following. Please
note that the core.db file is still being created. This output is not
telling me what is missing or failing to fix it :(     Any pointer will be
appreciated.

2017-06-04 00:21:52.739199 [INFO] switch_event.c:685 Activate Eventing
Engine.
2017-06-04 00:21:52.749458 [WARNING] switch_event.c:656 Create additional
event dispatch thread 0
2017-06-04 00:21:52.783529 [INFO] switch_nat.c:417 Scanning for NAT
2017-06-04 00:21:52.783645 [DEBUG] switch_nat.c:170 Checking for PMP 1/5
2017-06-04 00:21:53.783805 [DEBUG] switch_nat.c:170 Checking for PMP 2/5
2017-06-04 00:21:54.783954 [DEBUG] switch_nat.c:170 Checking for PMP 3/5
2017-06-04 00:21:55.784111 [DEBUG] switch_nat.c:170 Checking for PMP 4/5
2017-06-04 00:21:56.784274 [DEBUG] switch_nat.c:170 Checking for PMP 5/5
2017-06-04 00:21:57.784405 [ERR] switch_nat.c:199 Error checking for PMP
[general error]
2017-06-04 00:21:57.784435 [DEBUG] switch_nat.c:422 Checking for UPnP
2017-06-04 00:22:09.785033 [INFO] switch_nat.c:438 No PMP or UPnP NAT
devices detected!
2017-06-04 00:22:09.786686 [NOTICE] switch_core.c:2326 Set switchname to
fs02.prostarsentertainment.com
2017-06-04 00:22:09.787097 [INFO] switch_core_sqldb.c:3396 Opening DB
2017-06-04 00:22:09.787131 [ERR] switch_core_sqldb.c:587 NATIVE SQL ERR [no
such table: channels]
drop table channels
2017-06-04 00:22:09.787149 [ERR] switch_core_sqldb.c:587 NATIVE SQL ERR [no
such table: calls]
drop table calls
2017-06-04 00:22:09.787164 [ERR] switch_core_sqldb.c:587 NATIVE SQL ERR [no
such view: detailed_calls]
drop view detailed_calls
2017-06-04 00:22:09.787179 [ERR] switch_core_sqldb.c:587 NATIVE SQL ERR [no
such view: basic_calls]
drop view basic_calls
2017-06-04 00:22:09.787193 [ERR] switch_core_sqldb.c:587 NATIVE SQL ERR [no
such table: interfaces]
drop table interfaces
2017-06-04 00:22:09.787218 [ERR] switch_core_sqldb.c:587 NATIVE SQL ERR [no
such table: tasks]
drop table tasks
2017-06-04 00:22:09.787393 [DEBUG] switch_core_sqldb.c:1347 SQL ERR [no
such table: aliases]
[select hostname from aliases]
Auto Generating Table!
2017-06-04 00:22:09.787412 [DEBUG] switch_core_sqldb.c:1354 Ignoring SQL
ERR [no such table: aliases]
[DROP TABLE aliases]
2017-06-04 00:22:09.787562 [DEBUG] switch_core_sqldb.c:1347 SQL ERR [no
such table: complete]
[select hostname from complete]
Auto Generating Table!
2017-06-04 00:22:09.787582 [DEBUG] switch_core_sqldb.c:1354 Ignoring SQL
ERR [no such table: complete]
[DROP TABLE complete]
2017-06-04 00:22:09.787709 [DEBUG] switch_core_sqldb.c:1347 SQL ERR [no
such table: nat]
[select hostname from nat]
Auto Generating Table!
2017-06-04 00:22:09.787728 [DEBUG] switch_core_sqldb.c:1354 Ignoring SQL
ERR [no such table: nat]
[DROP TABLE nat]
2017-06-04 00:22:09.787822 [DEBUG] switch_core_sqldb.c:1347 SQL ERR [no
such table: registrations]
[delete from registrations where reg_user='']
Auto Generating Table!
2017-06-04 00:22:09.787842 [DEBUG] switch_core_sqldb.c:1354 Ignoring SQL
ERR [no such table: registrations]
[DROP TABLE registrations]
2017-06-04 00:22:09.788012 [DEBUG] switch_core_sqldb.c:1347 SQL ERR [no
such table: recovery]
[select hostname from recovery]
Auto Generating Table!
2017-06-04 00:22:09.788033 [DEBUG] switch_core_sqldb.c:1354 Ignoring SQL
ERR [no such table: recovery]
[DROP TABLE recovery]
2017-06-04 00:22:09.792959 [INFO] switch_core_sqldb.c:1693 CORE Starting
SQL thread.
2017-06-04 00:22:09.797506 [DEBUG] switch_scheduler.c:249 Added task 1
heartbeat (core) to run at 1496535729
2017-06-04 00:22:09.797533 [DEBUG] switch_scheduler.c:249 Added task 2
check_ip (core) to run at 1496535729




--
Luis Daniel Lucio Quiroz
CISSP, CISM, CISA
Linux, VoIP and much more fun
www.okay.com.mx

Need LCR? Check out LCR for FusionPBX with FreeSWITCH
Need Billing? Check out Billing for FusionPBX with FreeSWITCH

On Fri, Jun 2, 2017 at 6:07 PM, Michael Jerris <mike at jerris.com> wrote:

> take a look at the logs to see why it can’t connect to the db.
>
> > On Jun 2, 2017, at 11:30 AM, Luis Daniel Lucio Quiroz <
> luis.daniel.lucio at gmail.com> wrote:
> >
> > my FS 1.6.17 is not honoring my ODBC settings, it keeps creating core.db
> and other sqlite db's. I can confirm ODBC works, isql command connects
> without issues.
> >
> >    <param name="core-db-dsn" value="freeswitch:USER:XXXXXXXXXXXX" />
> >    <param name="core-recovery-db-dsn" value="freeswitch:USER:XXXXXXXX"
> />
> >
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170603/f4164c36/attachment-0001.html>

From luis.daniel.lucio at gmail.com  Sun Jun  4 00:57:00 2017
From: luis.daniel.lucio at gmail.com (Luis Daniel Lucio Quiroz)
Date: Sat, 3 Jun 2017 20:57:00 -0400
Subject: [Freeswitch-users] Not honoring ODBC in the core
In-Reply-To: <CAFLo2QzBSmcMunBb2g+CKCHZQ16MhbhvN66D-bPvSXTAJYk9gg@mail.gmail.com>
References: <CAFLo2Qz569hQM35R_r+kbKOHrc3vcWmyQH_PbBxFbPCMnRrC-w@mail.gmail.com>
 <CF8B652F-8C16-47AC-BFB7-DECC2ACBD532@jerris.com>
 <CAFLo2QzBSmcMunBb2g+CKCHZQ16MhbhvN66D-bPvSXTAJYk9gg@mail.gmail.com>
Message-ID: <CAFLo2QyEy00-5cJRKWncFAMN9K3JSfbPXvpqecZcGnVnv8swdA@mail.gmail.com>

It is working. It sounds like a dummy error, but after reading the source,
specifically the function _switch_cache_db_get_db_handle_dsn in the
file src/switch_core_sqldb.c, first i found that it is better to prefix
with ,odbc://, second I found in the freeswitch.xml the big issue
with <X-PRE-PROCESS cmd="include" data="autoload_configs/*.xml"/>, it seems
that my syncthing was creating double files without taking out the .xml
extension, this makes the to process another "backup" file which it was
overwriting the variable.

Thanks for your time

--
Luis Daniel Lucio Quiroz
CISSP, CISM, CISA
Linux, VoIP and much more fun
www.okay.com.mx

Need LCR? Check out LCR for FusionPBX with FreeSWITCH
Need Billing? Check out Billing for FusionPBX with FreeSWITCH

On Sat, Jun 3, 2017 at 8:25 PM, Luis Daniel Lucio Quiroz <
luis.daniel.lucio at gmail.com> wrote:

> I have managed to move all modules to ODBC but the core. If it helps, this
> deployment is using AWS RDS.
>
> I have turned on all the debug options, and i get the following. Please
> note that the core.db file is still being created. This output is not
> telling me what is missing or failing to fix it :(     Any pointer will be
> appreciated.
>
> 2017-06-04 00:21:52.739199 [INFO] switch_event.c:685 Activate Eventing
> Engine.
> 2017-06-04 00:21:52.749458 [WARNING] switch_event.c:656 Create additional
> event dispatch thread 0
> 2017-06-04 00:21:52.783529 [INFO] switch_nat.c:417 Scanning for NAT
> 2017-06-04 00:21:52.783645 [DEBUG] switch_nat.c:170 Checking for PMP 1/5
> 2017-06-04 00:21:53.783805 [DEBUG] switch_nat.c:170 Checking for PMP 2/5
> 2017-06-04 00:21:54.783954 [DEBUG] switch_nat.c:170 Checking for PMP 3/5
> 2017-06-04 00:21:55.784111 [DEBUG] switch_nat.c:170 Checking for PMP 4/5
> 2017-06-04 00:21:56.784274 [DEBUG] switch_nat.c:170 Checking for PMP 5/5
> 2017-06-04 00:21:57.784405 [ERR] switch_nat.c:199 Error checking for PMP
> [general error]
> 2017-06-04 00:21:57.784435 [DEBUG] switch_nat.c:422 Checking for UPnP
> 2017-06-04 00:22:09.785033 [INFO] switch_nat.c:438 No PMP or UPnP NAT
> devices detected!
> 2017-06-04 00:22:09.786686 [NOTICE] switch_core.c:2326 Set switchname to
> fs02.prostarsentertainment.com
> 2017-06-04 00:22:09.787097 [INFO] switch_core_sqldb.c:3396 Opening DB
> 2017-06-04 00:22:09.787131 [ERR] switch_core_sqldb.c:587 NATIVE SQL ERR
> [no such table: channels]
> drop table channels
> 2017-06-04 00:22:09.787149 [ERR] switch_core_sqldb.c:587 NATIVE SQL ERR
> [no such table: calls]
> drop table calls
> 2017-06-04 00:22:09.787164 [ERR] switch_core_sqldb.c:587 NATIVE SQL ERR
> [no such view: detailed_calls]
> drop view detailed_calls
> 2017-06-04 00:22:09.787179 [ERR] switch_core_sqldb.c:587 NATIVE SQL ERR
> [no such view: basic_calls]
> drop view basic_calls
> 2017-06-04 00:22:09.787193 [ERR] switch_core_sqldb.c:587 NATIVE SQL ERR
> [no such table: interfaces]
> drop table interfaces
> 2017-06-04 00:22:09.787218 [ERR] switch_core_sqldb.c:587 NATIVE SQL ERR
> [no such table: tasks]
> drop table tasks
> 2017-06-04 00:22:09.787393 [DEBUG] switch_core_sqldb.c:1347 SQL ERR [no
> such table: aliases]
> [select hostname from aliases]
> Auto Generating Table!
> 2017-06-04 00:22:09.787412 [DEBUG] switch_core_sqldb.c:1354 Ignoring SQL
> ERR [no such table: aliases]
> [DROP TABLE aliases]
> 2017-06-04 00:22:09.787562 [DEBUG] switch_core_sqldb.c:1347 SQL ERR [no
> such table: complete]
> [select hostname from complete]
> Auto Generating Table!
> 2017-06-04 00:22:09.787582 [DEBUG] switch_core_sqldb.c:1354 Ignoring SQL
> ERR [no such table: complete]
> [DROP TABLE complete]
> 2017-06-04 00:22:09.787709 [DEBUG] switch_core_sqldb.c:1347 SQL ERR [no
> such table: nat]
> [select hostname from nat]
> Auto Generating Table!
> 2017-06-04 00:22:09.787728 [DEBUG] switch_core_sqldb.c:1354 Ignoring SQL
> ERR [no such table: nat]
> [DROP TABLE nat]
> 2017-06-04 00:22:09.787822 [DEBUG] switch_core_sqldb.c:1347 SQL ERR [no
> such table: registrations]
> [delete from registrations where reg_user='']
> Auto Generating Table!
> 2017-06-04 00:22:09.787842 [DEBUG] switch_core_sqldb.c:1354 Ignoring SQL
> ERR [no such table: registrations]
> [DROP TABLE registrations]
> 2017-06-04 00:22:09.788012 [DEBUG] switch_core_sqldb.c:1347 SQL ERR [no
> such table: recovery]
> [select hostname from recovery]
> Auto Generating Table!
> 2017-06-04 00:22:09.788033 [DEBUG] switch_core_sqldb.c:1354 Ignoring SQL
> ERR [no such table: recovery]
> [DROP TABLE recovery]
> 2017-06-04 00:22:09.792959 [INFO] switch_core_sqldb.c:1693 CORE Starting
> SQL thread.
> 2017-06-04 00:22:09.797506 [DEBUG] switch_scheduler.c:249 Added task 1
> heartbeat (core) to run at 1496535729
> 2017-06-04 00:22:09.797533 [DEBUG] switch_scheduler.c:249 Added task 2
> check_ip (core) to run at 1496535729
>
>
>
>
> --
> Luis Daniel Lucio Quiroz
> CISSP, CISM, CISA
> Linux, VoIP and much more fun
> www.okay.com.mx
>
> Need LCR? Check out LCR for FusionPBX with FreeSWITCH
> Need Billing? Check out Billing for FusionPBX with FreeSWITCH
>
> On Fri, Jun 2, 2017 at 6:07 PM, Michael Jerris <mike at jerris.com> wrote:
>
>> take a look at the logs to see why it can’t connect to the db.
>>
>> > On Jun 2, 2017, at 11:30 AM, Luis Daniel Lucio Quiroz <
>> luis.daniel.lucio at gmail.com> wrote:
>> >
>> > my FS 1.6.17 is not honoring my ODBC settings, it keeps creating
>> core.db and other sqlite db's. I can confirm ODBC works, isql command
>> connects without issues.
>> >
>> >    <param name="core-db-dsn" value="freeswitch:USER:XXXXXXXXXXXX" />
>> >    <param name="core-recovery-db-dsn" value="freeswitch:USER:XXXXXXXX"
>> />
>> >
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.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
>> http://www.freeswitch.org
>
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170603/8a23963c/attachment.html>

From royj at yandex.ru  Sun Jun  4 10:22:44 2017
From: royj at yandex.ru (royj at yandex.ru)
Date: Sun, 04 Jun 2017 13:22:44 +0300
Subject: [Freeswitch-users] DTMF events
In-Reply-To: <91ba4d3c-b0a4-a255-2b17-75d91b7767fb@dwide.com>
References: <91ba4d3c-b0a4-a255-2b17-75d91b7767fb@dwide.com>
Message-ID: <5532821496571764@web40j.yandex.ru>

May be using mod_event_socket ( https://freeswitch.org/confluence/display/FREESWITCH/mod_event_socket ) will be good for you. You can listen for DTMF events arrived and other events withal.

04.06.2017, 11:19, "Dmitry Mordovin" <d.mordovin at dwide.com>:
> Hello!
>
> I want make application which will listen DTMF events and when it fire,
> send DTMF digit to web-url.
>
> For example, I use API for handle call state, execute_on_originate,
> execute_on_ring, execute_on_answer...
>
> Does exists API for DTMF?
>
> Anyone knows how can do it?
>
> Best regards, Dmitry Mordovin
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org


From jurijs.ivolga at gmail.com  Sun Jun  4 11:39:40 2017
From: jurijs.ivolga at gmail.com (Jurijs Ivolga)
Date: Sun, 4 Jun 2017 14:39:40 +0300
Subject: [Freeswitch-users] DTMF events
In-Reply-To: <5532821496571764@web40j.yandex.ru>
References: <91ba4d3c-b0a4-a255-2b17-75d91b7767fb@dwide.com>
 <5532821496571764@web40j.yandex.ru>
Message-ID: <CAOrA2Uao9y3aro7ovpf7OfGfV41dJ-H+-RvQrmgLOPMLf-6Vtw@mail.gmail.com>

Hi,

I think this is what you are looking for:

https://wiki.freeswitch.org/wiki/Channel_Variables#api_on_tone_detect

With kind regards,

Jurijs

On Sun, Jun 4, 2017 at 1:22 PM, <royj at yandex.ru> wrote:

> May be using mod_event_socket ( https://freeswitch.org/
> confluence/display/FREESWITCH/mod_event_socket ) will be good for you.
> You can listen for DTMF events arrived and other events withal.
>
> 04.06.2017, 11:19, "Dmitry Mordovin" <d.mordovin at dwide.com>:
> > Hello!
> >
> > I want make application which will listen DTMF events and when it fire,
> > send DTMF digit to web-url.
> >
> > For example, I use API for handle call state, execute_on_originate,
> > execute_on_ring, execute_on_answer...
> >
> > Does exists API for DTMF?
> >
> > Anyone knows how can do it?
> >
> > Best regards, Dmitry Mordovin
> >
> > ____________________________________________________________
> _____________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://confluence.freeswitch.org
> > http://www.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
> > http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170604/9dd6576f/attachment-0001.html>

From alexdruzhilov at gmail.com  Mon Jun  5 09:40:08 2017
From: alexdruzhilov at gmail.com (=?UTF-8?B?0JDQu9C10LrRgdCw0L3QtNGAINCU0YDRg9C20LjQu9C+0LI=?=)
Date: Mon, 5 Jun 2017 12:40:08 +0300
Subject: [Freeswitch-users] Fwd: Video conference with transcode mode is
 not working more than for 5-6 users
In-Reply-To: <CAKbxfG_2V49OXQV9Gja-i2rceZQcovGBm=Yak+skTdy1KvGYGA@mail.gmail.com>
References: <CAMfvzkgEjyb7mD26VtuMX+Uwqzqddb32ANYT7Xj3dY8g3jZHiQ@mail.gmail.com>
 <CAMfvzkiHwiJVRtaojTuUX-cFE50UvxuLUrq51MGUPW4ScjRokg@mail.gmail.com>
 <CAEwTsAgJM2GT2BZWBCeVZskazkCRUdbrrRuZTsVA9JhSrHFPsg@mail.gmail.com>
 <CAKbxfG_2V49OXQV9Gja-i2rceZQcovGBm=Yak+skTdy1KvGYGA@mail.gmail.com>
Message-ID: <CAMfvzkig8cTi2vXYO1mszsc_j06-0Z6yjs60hApRVe5UfCph6A@mail.gmail.com>

My use case is to make video conference where anybody can share his video
from camera or share screen. Video from camera and screen could have
different resolutions and frame rates so I would prefer transcode because
mux forces me to use a fixed frame rate and canvas size.

2017-06-03 19:26 GMT+03:00 Anthony Minessale <anthony.minessale at gmail.com>:

> Mux mode with 1x1 layout and minimize-video-encoding flag is better
>
>
> On Sat, Jun 3, 2017 at 8:59 AM Michael Jerris <mike at jerris.com> wrote:
>
>> We develop on debian 8.  It's worth trying there
>> to see if it's an issue with versions of things in centos.
>>
>> On Sat, Jun 3, 2017 at 3:37 AM Александр Дружилов <
>> alexdruzhilov at gmail.com> wrote:
>>
>>> OS: CentOS 7.2.1511
>>> Freeswitch: 1.6.17
>>>
>>> Steps:
>>> 1) create video conference (mod_sofia and conference-mode = transcode)
>>> with one video stream from conference owner
>>> 2) add 4-6 members in this conference who will receive owner's video
>>> stream
>>> 3) everyone in this conference will see dramatic degradation of video
>>> stream (incoming video stream bitrate falls from 1 Mbps to 0.5 Mbps, frame
>>> rate falls down to 10 fps from 30 fps, lag between video and audio stream
>>> appears)
>>>
>>> But I don't see any problems with CPU, memory or network. So does
>>> anybody knows whether it is an issue or it's how freeswitch 'transcode'
>>> mode should works? And what to do to tune performance of this mode?
>>> ____________________________________________________________
>>> _____________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.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
>>> http://www.freeswitch.org
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.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
>> http://www.freeswitch.org
>
> --
> Anthony Minessale II       ♬ @anthmfs  ♬ @FreeSWITCH  ♬
>
> ☞ http://freeswitch.org/  ☞ http://cluecon.com/  ☞
> http://twitter.com/FreeSWITCH
> ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+
> <http://freeswitch.org/g+>*
>
> ClueCon Weekly Development Call
> ☎ sip:888 at conference.freeswitch.org  ☎ +19193869900
>
> https://www.youtube.com/watch?v=9XXgW34t40s
> https://www.youtube.com/watch?v=NLaDpGQuZDA
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170605/42bbd54c/attachment.html>

From loi.dangthanh at gmail.com  Mon Jun  5 09:46:46 2017
From: loi.dangthanh at gmail.com (=?UTF-8?B?TOG7o2kgxJDhurduZw==?=)
Date: Mon, 05 Jun 2017 09:46:46 +0000
Subject: [Freeswitch-users] About proxy-hold codec re-negotiation
	behavior.
In-Reply-To: <CAGsrq_vd06aJueTAXwsVPjqcZMznJ68PoVtWEVZ4AR9bt7cfdw@mail.gmail.com>
References: <CAGsrq_vd06aJueTAXwsVPjqcZMznJ68PoVtWEVZ4AR9bt7cfdw@mail.gmail.com>
Message-ID: <CAGsrq_uCZ15T+PwM2aeF_jVZrE38vBS=23=aZSauUs-7=PHB2Q@mail.gmail.com>

Hi guys, I resend this because I guess my first email does not reach at all.
Helps are appreciated.
rgds,
Loi Dang

On Mon, May 22, 2017 at 5:49 PM Lợi Đặng <loi.dangthanh at gmail.com> wrote:

> Hi list, I'm using FS with *proxy-hold*.
> Call flow is simple
> A -> FreeSWITCH -> B
> Assume FS, A and B support PCMA and PCMU, FS prefer PCMA in codec-prefs
> profile configuration.
> In affects of my configuration for *greedy early negotiation* and *disable
> transcoding*, PCMA is negotiated in both legs for initial INVITE, that's
> good, and expected.
>
> But then I remove PCMA from A and compose a re-INVITE for holding with
> PCMU only, the a leg is re-negotiated with PCMU, but the b leg have
> re-INVITE with PCMA due to *proxy-hold* variable, that causes transcoding
> happened after hold.
>
> My desire is to have FS to re-negotiate with b leg too, not so similar but
> as what I observed when I was using it with media proxy option ( A-A before
> hold, U-U after hold).
>
> I tried late negotiation in this case, but no luck.
> So is that not able for FS to re-negotiate codec on b leg, in using
> proxy-hold?
> Any advise is appreciated.
>
> rgds,
> Loi Dang
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170605/c7c56ce1/attachment.html>

From boesemar at gmail.com  Mon Jun  5 09:53:00 2017
From: boesemar at gmail.com (boesemar .)
Date: Mon, 5 Jun 2017 10:53:00 +0100
Subject: [Freeswitch-users] Enabling new languages for Say
Message-ID: <CAPLWHmjLiKSk_hobMzSbOGXq9iu4gMFaeNdq-SwN3QPJvpJq_g@mail.gmail.com>

Hi Brian,

mod_say_pt is definitely loaded:

> module_exists mod_say_pt
true

Can you think of any other cause for the "Invalid SAY Interface [pt]" ?

Martin


PS: sorry for breaking the thread - problem with email



Brian West brian at freeswitch.org
Fri Jun 2 14:51:25 UTC 2017

I would guess mod_say_pt is not loaded.

On Fri, Jun 2, 2017 at 4:54 AM, Martin Boese <martin at maxnet.ao
<http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>>
wrote:

>* Hi!
*>>* FreeSWITCH Version 1.6.15-32-bec4538~64bit (-32-bec4538 64bit)
*>* Debian Jessie. Vanilla config.
*>>* I am trying to enable portuguese to say numbers using dptools "Say".
*>>* This is what I did:
*>*  - Module mod_say_pt is loaded
*>*  - in freeswitch.xml
*>*    include lang/pt/pt_PT.xml
*>*  - Downloaded sounds from
*>*    https://github.com/jpawlowski/freeswitch-sounds-ttsand
<https://github.com/jpawlowski/freeswitch-sounds-ttsand> placed them
*>*    into /usr/share/freeswitch/sounds
*>*    Directory structure is now like: /pt/tts/google/ascii/16000/...
*>>* CLI> say_string t.wav pt NUMBER pronounced 123
*>* [ERR] switch_xml.c:3180 Can't find phrases tag
*>>* ..I found out that lang/pt_PT.xml seems to be missing the
*>* <phrases><macros> tags within the <language> tag (vanilla config). I
*>* fixed that.
*>>* Now:
*>* CLI> say_string t.wav pt NUMBER pronounced 123
*>* [ERR] switch_ivr.c:3726 Invalid SAY Interface [pt]!
*>>* BTW: English works fine:
*>* CLI> say_string t.wav en NUMBER pronounced 123
*>* file_string://digits/1.wav!digits/hundred.wav!digits/20.wav!digits/3.wav
*>>* I also tried other languages but have same error "Invalid SAY
*>* Interface".
*>>* What am I missing. Please help.
*>>* Thanks,
*>* Martin
*>>* _________________________________________________________________________
*>* Professional FreeSWITCH Consulting Services:
*>* consulting at freeswitch.org
<http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
*>* http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com>
*>>* Official FreeSWITCH Sites
*>* http://www.freeswitch.org <http://www.freeswitch.org>
*>* http://confluence.freeswitch.org <http://confluence.freeswitch.org>
*>* http://www.cluecon.com <http://www.cluecon.com>
*>>* FreeSWITCH-users mailing list
*>* FreeSWITCH-users at lists.freeswitch.org
<http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
*>* 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>
*>* http://www.freeswitch.org <http://www.freeswitch.org>
*



-- 

*Brian West*brian at freeswitch.org
<http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>

*Twitter: @FreeSWITCH , @briankwest*
http://www.freeswitchbook.comhttp://www.freeswitchcookbook.com

Book a phone call (CST) <https://freeswitch.com/appointment>

Allison prompts for FreeSWITCH:

*https://www.gofundme.com/allison-prompts-for-freeswitch*
<https://www.gofundme.com/allison-prompts-for-freeswitch>

Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
/r/freeswitch <https://www.reddit.com/r/freeswitch>

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*Skype:*briankwest
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170602/d72352fe/attachment.html>

------------------------------
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170605/9d84f4f3/attachment-0001.html>

From denys.pozniak at crazycall.com  Mon Jun  5 11:41:16 2017
From: denys.pozniak at crazycall.com (Denys Pozniak)
Date: Mon, 5 Jun 2017 13:41:16 +0200
Subject: [Freeswitch-users] question about HA solution
In-Reply-To: <CAFLo2QxxbZ5Ged0J0p9_VfQ2nkGtuV0CDp9G763EjjuvqUv8iw@mail.gmail.com>
References: <CADcrYG+RHGQ2fC7dSKK9x9WfVvYcTczOtbppCLgttRCfMGqArA@mail.gmail.com>
 <CAFLo2QxxbZ5Ged0J0p9_VfQ2nkGtuV0CDp9G763EjjuvqUv8iw@mail.gmail.com>
Message-ID: <CADcrYG+sO0+WYUtYVN6pkpU-5SwyfurfshdUdhxb-c4Az1s49Q@mail.gmail.com>

Hello!

Thank you *Raymond* about your explanation, but I dont agree with some
point:
*If it really need an answer about your question -- "if it is possible to
move calls back".   I think it's unnecessary.*  - in my case I have two not
equal servers, so I need to have only one as a master.
If switchover happens I need to have ability to restore master back.

Thank you *Luis* for your link, you can do simple test to understand what I
am talking about: do call -> check on master and slave #show channels ->
restart FS on slave -> check on master #show channels. In my case I dont
see any active calls after this, so restoring back is not possible.



On 3 June 2017 at 22:16, Luis Daniel Lucio Quiroz <
luis.daniel.lucio at gmail.com> wrote:

> You may want to read this article.
>
> http://inside-out.xyz/technology/how-to-configure-freeswitch-for-ha.html
>
> Le 31 mai 2017 6:29 PM, "Denys Pozniak" <denys.pozniak at crazycall.com> a
> écrit :
>
> Hello!
>
> I built FS HA solution based on keepalived and mysql master-master.
> It works ok generally, but as I understand FS after restarting cleaning
> own database.
>
> So when node1 fails calls jump to node2, after script restarts node1 it is
> not possible to move calls back.
>
> Tried options in switch.conf.xml, but no luck:
>
>     <param name="auto-create-schemas" value="false"/>
>     <param name="auto-clear-sql" value="false"/>
>
> Is there is a way to solve this?
>
> BR,
> Denys
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org
>
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170605/0e1fe695/attachment.html>

From vladislaus at gmail.com  Mon Jun  5 13:20:14 2017
From: vladislaus at gmail.com (Andres Gomez)
Date: Mon, 5 Jun 2017 08:20:14 -0500
Subject: [Freeswitch-users] Videoconference error with H263+ [ERR]
	avcodec.c:736
Message-ID: <CAHzfA-i2VJc9Pw=SXnY+NtY2xOgpXhRCfWFPz6RCBwkyjro83Q@mail.gmail.com>

Hello.

I see this console error  videcoconference  with H263+

2017-06-05 08:14:44.668608 [ERR] avcodec.c:736 len: 1200, mark:0 00 00 d2 ef
2017-06-05 08:14:44.668608 [ERR] avcodec.c:736 len: 1200, mark:0 00 00 f4 12
2017-06-05 08:14:44.668608 [ERR] avcodec.c:736 len: 382, mark:1 00 00 01 c0
2017-06-05 08:14:44.678607 [ERR] avcodec.c:736 len: 1198, mark:0 04 00 82 ca
2017-06-05 08:14:44.678607 [ERR] avcodec.c:736 len: 1200, mark:0 00 00 56 83
2017-06-05 08:14:44.678607 [ERR] avcodec.c:736 len: 1200, mark:0 00 00 24 4e
2017-06-05 08:14:44.678607 [ERR] avcodec.c:736 len: 1200, mark:0 00 00 e9 db
2017-06-05 08:14:44.678607 [ERR] avcodec.c:736 len: 1200, mark:0 00 00 9a 3a

Any Idea why?

Regards
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170605/4ee5a8bf/attachment.html>

From brian at freeswitch.org  Mon Jun  5 14:27:00 2017
From: brian at freeswitch.org (Brian West)
Date: Mon, 5 Jun 2017 09:27:00 -0500
Subject: [Freeswitch-users] About proxy-hold codec re-negotiation
	behavior.
In-Reply-To: <CAGsrq_uCZ15T+PwM2aeF_jVZrE38vBS=23=aZSauUs-7=PHB2Q@mail.gmail.com>
References: <CAGsrq_vd06aJueTAXwsVPjqcZMznJ68PoVtWEVZ4AR9bt7cfdw@mail.gmail.com>
 <CAGsrq_uCZ15T+PwM2aeF_jVZrE38vBS=23=aZSauUs-7=PHB2Q@mail.gmail.com>
Message-ID: <CAEJMVkAZVp120-c8b_3WM4=kpExj5o61zoerXqYnOKbEz_1WaA@mail.gmail.com>

Its working as designed.  You have to remember we're a B2BUA... so we have
some behaviors that may behave proxy like, but FreeSWITCH is still not a
proxy.

On Mon, Jun 5, 2017 at 4:46 AM, Lợi Đặng <loi.dangthanh at gmail.com> wrote:

> Hi guys, I resend this because I guess my first email does not reach at
> all.
> Helps are appreciated.
> rgds,
> Loi Dang
>
> On Mon, May 22, 2017 at 5:49 PM Lợi Đặng <loi.dangthanh at gmail.com> wrote:
>
>> Hi list, I'm using FS with *proxy-hold*.
>> Call flow is simple
>> A -> FreeSWITCH -> B
>> Assume FS, A and B support PCMA and PCMU, FS prefer PCMA in codec-prefs
>> profile configuration.
>> In affects of my configuration for *greedy early negotiation* and *disable
>> transcoding*, PCMA is negotiated in both legs for initial INVITE, that's
>> good, and expected.
>>
>> But then I remove PCMA from A and compose a re-INVITE for holding with
>> PCMU only, the a leg is re-negotiated with PCMU, but the b leg have
>> re-INVITE with PCMA due to *proxy-hold* variable, that causes
>> transcoding happened after hold.
>>
>> My desire is to have FS to re-negotiate with b leg too, not so similar
>> but as what I observed when I was using it with media proxy option ( A-A
>> before hold, U-U after hold).
>>
>> I tried late negotiation in this case, but no luck.
>> So is that not able for FS to re-negotiate codec on b leg, in using
>> proxy-hold?
>> Any advise is appreciated.
>>
>> rgds,
>> Loi Dang
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org
>



-- 

*Brian West*
brian at freeswitch.org

*Twitter: @FreeSWITCH , @briankwest*

http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

Book a phone call (CST) <https://freeswitch.com/appointment>

Allison prompts for FreeSWITCH:

*https://www.gofundme.com/allison-prompts-for-freeswitch*
<https://www.gofundme.com/allison-prompts-for-freeswitch>

Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
/r/freeswitch <https://www.reddit.com/r/freeswitch>

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*Skype:*briankwest
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170605/97d1cfe7/attachment.html>

From mike at jerris.com  Mon Jun  5 15:03:26 2017
From: mike at jerris.com (Michael Jerris)
Date: Mon, 5 Jun 2017 11:03:26 -0400
Subject: [Freeswitch-users] Enabling new languages for Say
In-Reply-To: <CAPLWHmjLiKSk_hobMzSbOGXq9iu4gMFaeNdq-SwN3QPJvpJq_g@mail.gmail.com>
References: <CAPLWHmjLiKSk_hobMzSbOGXq9iu4gMFaeNdq-SwN3QPJvpJq_g@mail.gmail.com>
Message-ID: <590ED298-8358-44F1-B391-B9FCA07AE6DA@jerris.com>

what log output do you get when you load mod_say?  what output do you get to “show say”

> On Jun 5, 2017, at 5:53 AM, boesemar . <boesemar at gmail.com> wrote:
> 
> Hi Brian,
> 
> mod_say_pt is definitely loaded:
> 
> > module_exists mod_say_pt
> true
> 
> Can you think of any other cause for the "Invalid SAY Interface [pt]" ?
> Martin
> 
> PS: sorry for breaking the thread - problem with email
> 
> 
> Brian West brian at freeswitch.org <http://freeswitch.org/>
> Fri Jun 2 14:51:25 UTC 2017
> 
> I would guess mod_say_pt is not loaded.
> 
> On Fri, Jun 2, 2017 at 4:54 AM, Martin Boese <martin at maxnet.ao <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>> wrote:
> 
> > Hi!
> >
> > FreeSWITCH Version 1.6.15-32-bec4538~64bit (-32-bec4538 64bit)
> > Debian Jessie. Vanilla config.
> >
> > I am trying to enable portuguese to say numbers using dptools "Say".
> >
> > This is what I did:
> >  - Module mod_say_pt is loaded
> >  - in freeswitch.xml
> >    include lang/pt/pt_PT.xml
> >  - Downloaded sounds from
> >    https://github.com/jpawlowski/freeswitch-sounds-ttsand <https://github.com/jpawlowski/freeswitch-sounds-ttsand> placed them
> >    into /usr/share/freeswitch/sounds
> >    Directory structure is now like: /pt/tts/google/ascii/16000/...
> >
> > CLI> say_string t.wav pt NUMBER pronounced 123
> > [ERR] switch_xml.c:3180 Can't find phrases tag
> >
> > ..I found out that lang/pt_PT.xml seems to be missing the
> > <phrases><macros> tags within the <language> tag (vanilla config). I
> > fixed that.
> >
> > Now:
> > CLI> say_string t.wav pt NUMBER pronounced 123
> > [ERR] switch_ivr.c:3726 Invalid SAY Interface [pt]!
> >
> > BTW: English works fine:
> > CLI> say_string t.wav en NUMBER pronounced 123
> > file_string://digits/1.wav!digits/hundred.wav!digits/20.wav!digits/3.wav <>
> >
> > I also tried other languages but have same error "Invalid SAY
> > Interface".
> >
> > What am I missing. Please help.
> >
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170605/ad5d43f5/attachment.html>

From mike at jerris.com  Mon Jun  5 15:04:25 2017
From: mike at jerris.com (Michael Jerris)
Date: Mon, 5 Jun 2017 11:04:25 -0400
Subject: [Freeswitch-users] question about HA solution
In-Reply-To: <CADcrYG+sO0+WYUtYVN6pkpU-5SwyfurfshdUdhxb-c4Az1s49Q@mail.gmail.com>
References: <CADcrYG+RHGQ2fC7dSKK9x9WfVvYcTczOtbppCLgttRCfMGqArA@mail.gmail.com>
 <CAFLo2QxxbZ5Ged0J0p9_VfQ2nkGtuV0CDp9G763EjjuvqUv8iw@mail.gmail.com>
 <CADcrYG+sO0+WYUtYVN6pkpU-5SwyfurfshdUdhxb-c4Az1s49Q@mail.gmail.com>
Message-ID: <8A13E0EC-FA13-4267-80F6-CE1A8E8360CF@jerris.com>

recovered calls will get new entries in the table.

> On Jun 5, 2017, at 7:41 AM, Denys Pozniak <denys.pozniak at crazycall.com> wrote:
> 
> Hello!
> 
> Thank you Raymond about your explanation, but I dont agree with some point: 
> If it really need an answer about your question -- "if it is possible to move calls back".   I think it's unnecessary.  - in my case I have two not equal servers, so I need to have only one as a master.
> If switchover happens I need to have ability to restore master back.
> 
> Thank you Luis for your link, you can do simple test to understand what I am talking about: do call -> check on master and slave #show channels -> restart FS on slave -> check on master #show channels. In my case I dont see any active calls after this, so restoring back is not possible.
> 
> 
> 
> On 3 June 2017 at 22:16, Luis Daniel Lucio Quiroz <luis.daniel.lucio at gmail.com <mailto:luis.daniel.lucio at gmail.com>> wrote:
> You may want to read this article.
> 
> http://inside-out.xyz/technology/how-to-configure-freeswitch-for-ha.html <http://inside-out.xyz/technology/how-to-configure-freeswitch-for-ha.html>
> 
> Le 31 mai 2017 6:29 PM, "Denys Pozniak" <denys.pozniak at crazycall.com <mailto:denys.pozniak at crazycall.com>> a écrit :
> Hello!
> 
> I built FS HA solution based on keepalived and mysql master-master.
> It works ok generally, but as I understand FS after restarting cleaning own database.
> 
> So when node1 fails calls jump to node2, after script restarts node1 it is not possible to move calls back.
> 
> Tried options in switch.conf.xml, but no luck:
> 
>     <param name="auto-create-schemas" value="false"/>
>     <param name="auto-clear-sql" value="false"/>
> 
> Is there is a way to solve this?
> 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170605/d2ae9da0/attachment-0001.html>

From denys.pozniak at crazycall.com  Mon Jun  5 17:50:18 2017
From: denys.pozniak at crazycall.com (Denys Pozniak)
Date: Mon, 5 Jun 2017 19:50:18 +0200
Subject: [Freeswitch-users] question about HA solution
In-Reply-To: <8A13E0EC-FA13-4267-80F6-CE1A8E8360CF@jerris.com>
References: <CADcrYG+RHGQ2fC7dSKK9x9WfVvYcTczOtbppCLgttRCfMGqArA@mail.gmail.com>
 <CAFLo2QxxbZ5Ged0J0p9_VfQ2nkGtuV0CDp9G763EjjuvqUv8iw@mail.gmail.com>
 <CADcrYG+sO0+WYUtYVN6pkpU-5SwyfurfshdUdhxb-c4Az1s49Q@mail.gmail.com>
 <8A13E0EC-FA13-4267-80F6-CE1A8E8360CF@jerris.com>
Message-ID: <CADcrYGLbN1Ut6riF1Lf7UvwG9ypvR2ekWxt1m=e5BmqOzP_c=w@mail.gmail.com>

Yes, correct. But when you restart FS on slave, it will erase database. And
option auto-clear-sql=false not working for me.

On Jun 5, 2017 6:32 PM, "Michael Jerris" <mike at jerris.com> wrote:

recovered calls will get new entries in the table.

On Jun 5, 2017, at 7:41 AM, Denys Pozniak <denys.pozniak at crazycall.com>
wrote:

Hello!

Thank you *Raymond* about your explanation, but I dont agree with some
point:
*If it really need an answer about your question -- "if it is possible to
move calls back".   I think it's unnecessary.*  - in my case I have two not
equal servers, so I need to have only one as a master.
If switchover happens I need to have ability to restore master back.

Thank you *Luis* for your link, you can do simple test to understand what I
am talking about: do call -> check on master and slave #show channels ->
restart FS on slave -> check on master #show channels. In my case I dont
see any active calls after this, so restoring back is not possible.



On 3 June 2017 at 22:16, Luis Daniel Lucio Quiroz <
luis.daniel.lucio at gmail.com> wrote:

> You may want to read this article.
>
> http://inside-out.xyz/technology/how-to-configure-freeswitch-for-ha.html
>
> Le 31 mai 2017 6:29 PM, "Denys Pozniak" <denys.pozniak at crazycall.com> a
> écrit :
>
> Hello!
>
> I built FS HA solution based on keepalived and mysql master-master.
> It works ok generally, but as I understand FS after restarting cleaning
> own database.
>
> So when node1 fails calls jump to node2, after script restarts node1 it is
> not possible to move calls back.
>
> Tried options in switch.conf.xml, but no luck:
>
>     <param name="auto-create-schemas" value="false"/>
>     <param name="auto-clear-sql" value="false"/>
>
> Is there is a way to solve this?
>
>

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.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
http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170605/be3b0202/attachment.html>

From brian at freeswitch.org  Mon Jun  5 19:39:03 2017
From: brian at freeswitch.org (Brian West)
Date: Mon, 5 Jun 2017 14:39:03 -0500
Subject: [Freeswitch-users] Calling on the community for Bug Marshals
Message-ID: <CAEJMVkChkspixRypT0rMCzawf4F14Tn3vUzYFzN-qtiqVmRnrg@mail.gmail.com>

FreeSWITCHers,

We are in need of a few good bug marshals, We are trying to get 1.8 ready
and out the door and the more help we have testing and working thru patches
on JIRA the quicker it will arrive.  If you're interested in helping us out
email me directly.  We are also considering bringing back a few days a week
we are sitting in 888 and helping the community out with issues pending in
JIRA.

Also we are only about 2600 short on the gofund me for the Allison prompts,
which will be delivered sometime this week.  ;)  So help us get over that
last little bit this week.

Thanks,

-- 

*Brian West*
brian at freeswitch.org

*Twitter: @FreeSWITCH , @briankwest*

http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

Book a phone call (CST) <https://freeswitch.com/appointment>

Allison prompts for FreeSWITCH:

*https://www.gofundme.com/allison-prompts-for-freeswitch*
<https://www.gofundme.com/allison-prompts-for-freeswitch>

Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
/r/freeswitch <https://www.reddit.com/r/freeswitch>

*T:*+19184209001 <(918)%20420-9001> | *F:*+19184209002 <(918)%20420-9002> |
*M:*+1918424WEST (9378)
*Skype:*briankwest
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170605/61ab25e6/attachment.html>

From loi.dangthanh at gmail.com  Tue Jun  6 02:31:28 2017
From: loi.dangthanh at gmail.com (=?UTF-8?B?TOG7o2kgxJDhurduZw==?=)
Date: Tue, 06 Jun 2017 02:31:28 +0000
Subject: [Freeswitch-users] About proxy-hold codec re-negotiation
	behavior.
In-Reply-To: <CAEJMVkAZVp120-c8b_3WM4=kpExj5o61zoerXqYnOKbEz_1WaA@mail.gmail.com>
References: <CAGsrq_vd06aJueTAXwsVPjqcZMznJ68PoVtWEVZ4AR9bt7cfdw@mail.gmail.com>
 <CAGsrq_uCZ15T+PwM2aeF_jVZrE38vBS=23=aZSauUs-7=PHB2Q@mail.gmail.com>
 <CAEJMVkAZVp120-c8b_3WM4=kpExj5o61zoerXqYnOKbEz_1WaA@mail.gmail.com>
Message-ID: <CAGsrq_uUoesVyPN8CfFm6M5oa9bXYi4NS+cv8F5=G_GxMHvJsg@mail.gmail.com>

yep, I'm not expecting too much this can be done, the design is good but
I'm just curious about profile variables
`renegotiate-codec-on-reinvite/hold` that was removed since 1.6.10, guess
the job is done in old days.
rgds,
Loi Dang

On Mon, Jun 5, 2017 at 9:30 PM Brian West <brian at freeswitch.org> wrote:

> Its working as designed.  You have to remember we're a B2BUA... so we have
> some behaviors that may behave proxy like, but FreeSWITCH is still not a
> proxy.
>
> On Mon, Jun 5, 2017 at 4:46 AM, Lợi Đặng <loi.dangthanh at gmail.com> wrote:
>
>> Hi guys, I resend this because I guess my first email does not reach at
>> all.
>> Helps are appreciated.
>> rgds,
>> Loi Dang
>>
>> On Mon, May 22, 2017 at 5:49 PM Lợi Đặng <loi.dangthanh at gmail.com> wrote:
>>
>>> Hi list, I'm using FS with *proxy-hold*.
>>> Call flow is simple
>>> A -> FreeSWITCH -> B
>>> Assume FS, A and B support PCMA and PCMU, FS prefer PCMA in codec-prefs
>>> profile configuration.
>>> In affects of my configuration for *greedy early negotiation* and *disable
>>> transcoding*, PCMA is negotiated in both legs for initial INVITE,
>>> that's good, and expected.
>>>
>>> But then I remove PCMA from A and compose a re-INVITE for holding with
>>> PCMU only, the a leg is re-negotiated with PCMU, but the b leg have
>>> re-INVITE with PCMA due to *proxy-hold* variable, that causes
>>> transcoding happened after hold.
>>>
>>> My desire is to have FS to re-negotiate with b leg too, not so similar
>>> but as what I observed when I was using it with media proxy option ( A-A
>>> before hold, U-U after hold).
>>>
>>> I tried late negotiation in this case, but no luck.
>>> So is that not able for FS to re-negotiate codec on b leg, in using
>>> proxy-hold?
>>> Any advise is appreciated.
>>>
>>> rgds,
>>> Loi Dang
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.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
>> http://www.freeswitch.org
>>
>
>
>
> --
>
> *Brian West*
> brian at freeswitch.org
>
> *Twitter: @FreeSWITCH , @briankwest*
>
> http://www.freeswitchbook.com
> http://www.freeswitchcookbook.com
>
> Book a phone call (CST) <https://freeswitch.com/appointment>
>
> Allison prompts for FreeSWITCH:
>
> *https://www.gofundme.com/allison-prompts-for-freeswitch*
> <https://www.gofundme.com/allison-prompts-for-freeswitch>
>
> Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
> /r/freeswitch <https://www.reddit.com/r/freeswitch>
>
> *T:*+19184209001 <(918)%20420-9001> | *F:*+19184209002 <(918)%20420-9002>
> | *M:*+1918424WEST (9378)
> *Skype:*briankwest
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170606/af217068/attachment-0001.html>

From hunterj91 at hotmail.com  Tue Jun  6 06:59:43 2017
From: hunterj91 at hotmail.com (Jonathan Hunter)
Date: Tue, 6 Jun 2017 06:59:43 +0000
Subject: [Freeswitch-users] Monitoring call quality using
	variable_rtp_audio_xxx
In-Reply-To: <DB3PR02MB04275464611C567EBFBCA711BDF10@DB3PR02MB0427.eurprd02.prod.outlook.com>
References: <DB3PR02MB04275464611C567EBFBCA711BDF10@DB3PR02MB0427.eurprd02.prod.outlook.com>
Message-ID: <DB3PR02MB04274C1149CD867FA5D8714ABDCB0@DB3PR02MB0427.eurprd02.prod.outlook.com>

Hi Guys,


To be more specific, what is the meaning of this parameter?


in_flaw_total


I am becoming clear on the other parameters!


Thanks


Jon


________________________________
From: Jonathan Hunter <hunterj91 at hotmail.com>
Sent: 31 May 2017 19:24
To: freeswitch-users at lists.freeswitch.org
Subject: Monitoring call quality using variable_rtp_audio_xxx


Hi Guys,

Sorry for the noise, we are looking to poll calls in progress to grab the rtp_audio variables when a call is in progress, and we want to understand if this is a good approach and to that end what values/variables we should consider and what ranges should we be working with?

I understand mos and quality percentage but what other values are a good indicator?

Again Im clear on jitter and its meaning just want to understand whats what as not everything understandably is not documented.

Many thanks

Jon

    "variable_rtp_audio_recv_pt": "8",
    "variable_rtp_audio_in_raw_bytes": "374960",
    "variable_rtp_audio_in_media_bytes": "374616",
    "variable_rtp_audio_in_packet_count": "2180",
    "variable_rtp_audio_in_media_packet_count": "2178",
    "variable_rtp_audio_in_skip_packet_count": "92",
    "variable_rtp_audio_in_jitter_packet_count": "0",
    "variable_rtp_audio_in_dtmf_packet_count": "0",
    "variable_rtp_audio_in_cng_packet_count": "0",
    "variable_rtp_audio_in_flush_packet_count": "2",
    "variable_rtp_audio_in_largest_jb_size": "0",
    "variable_rtp_audio_in_jitter_min_variance": "28.57",
    "variable_rtp_audio_in_jitter_max_variance": "116.33",
    "variable_rtp_audio_in_jitter_loss_rate": "0.02",
    "variable_rtp_audio_in_jitter_burst_rate": "0.98",
    "variable_rtp_audio_in_mean_interval": "20.39",
    "variable_rtp_audio_in_flaw_total": "43",
    "variable_rtp_audio_in_quality_percentage": "97.00",
    "variable_rtp_audio_in_mos": "4.47",
    "variable_rtp_audio_out_raw_bytes": "210012",
    "variable_rtp_audio_out_media_bytes": "210012",
    "variable_rtp_audio_out_packet_count": "1221",
    "variable_rtp_audio_out_media_packet_count": "1221",
    "variable_rtp_audio_out_skip_packet_count": "0",
    "variable_rtp_audio_out_dtmf_packet_count": "0",
    "variable_rtp_audio_out_cng_packet_count": "0",
    "variable_rtp_audio_rtcp_packet_count": "0",
    "variable_rtp_audio_rtcp_octet_count": "0",

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170606/7613741b/attachment.html>

From boesemar at gmail.com  Tue Jun  6 07:19:38 2017
From: boesemar at gmail.com (boesemar .)
Date: Tue, 6 Jun 2017 08:19:38 +0100
Subject: [Freeswitch-users] Enabling new languages for Say
In-Reply-To: <590ED298-8358-44F1-B391-B9FCA07AE6DA@jerris.com>
References: <CAPLWHmjLiKSk_hobMzSbOGXq9iu4gMFaeNdq-SwN3QPJvpJq_g@mail.gmail.com>
 <590ED298-8358-44F1-B391-B9FCA07AE6DA@jerris.com>
Message-ID: <CAPLWHmjA5H_G2eSabCJX5CFhj6K7vJDwP3q30xxeAk8oj6v0fg@mail.gmail.com>

Hi Michael,

> show say
type,name,ikey
say,en,mod_say_en
say,pt,mod_say_pt

2 total.

Thx,
Martin




On Mon, Jun 5, 2017 at 4:03 PM, Michael Jerris <mike at jerris.com> wrote:

> what log output do you get when you load mod_say?  what output do you get
> to “show say”
>
> On Jun 5, 2017, at 5:53 AM, boesemar . <boesemar at gmail.com> wrote:
>
> Hi Brian,
>
> mod_say_pt is definitely loaded:
>
> > module_exists mod_say_pt
> true
>
> Can you think of any other cause for the "Invalid SAY Interface [pt]" ?
>
> Martin
>
>
> PS: sorry for breaking the thread - problem with email
>
>
>
> Brian West brian at freeswitch.org
> Fri Jun 2 14:51:25 UTC 2017
>
> I would guess mod_say_pt is not loaded.
>
> On Fri, Jun 2, 2017 at 4:54 AM, Martin Boese <martin at maxnet.ao <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>> wrote:
>
> >* Hi!
> *>>* FreeSWITCH Version 1.6.15-32-bec4538~64bit (-32-bec4538 64bit)
> *>* Debian Jessie. Vanilla config.
> *>>* I am trying to enable portuguese to say numbers using dptools "Say".
> *>>* This is what I did:
> *>*  - Module mod_say_pt is loaded
> *>*  - in freeswitch.xml
> *>*    include lang/pt/pt_PT.xml
> *>*  - Downloaded sounds from
> *>*    https://github.com/jpawlowski/freeswitch-sounds-ttsand <https://github.com/jpawlowski/freeswitch-sounds-ttsand> placed them
> *>*    into /usr/share/freeswitch/sounds
> *>*    Directory structure is now like: /pt/tts/google/ascii/16000/...
> *>>* CLI> say_string t.wav pt NUMBER pronounced 123
> *>* [ERR] switch_xml.c:3180 Can't find phrases tag
> *>>* ..I found out that lang/pt_PT.xml seems to be missing the
> *>* <phrases><macros> tags within the <language> tag (vanilla config). I
> *>* fixed that.
> *>>* Now:
> *>* CLI> say_string t.wav pt NUMBER pronounced 123
> *>* [ERR] switch_ivr.c:3726 Invalid SAY Interface [pt]!
> *>>* BTW: English works fine:
> *>* CLI> say_string t.wav en NUMBER pronounced 123
> *>* file_string://digits/1.wav!digits/hundred.wav!digits/20.wav!digits/3.wav
> *>>* I also tried other languages but have same error "Invalid SAY
> *>* Interface".
> *>>* What am I missing. Please help.
> *>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170606/6d448a02/attachment.html>

From hardikitpl at gmail.com  Tue Jun  6 11:11:33 2017
From: hardikitpl at gmail.com (Hardik Patel)
Date: Tue, 6 Jun 2017 16:41:33 +0530
Subject: [Freeswitch-users] Fax receive issue with t30 codec
Message-ID: <CAHdn4vjxr5vVoO7Oc1P0u-xg6dJNYBhjxU=y5iz6VP5_JEcUjA@mail.gmail.com>

Hello,

I am using opensips as entry point using dispatcher. opensips( 127.0.0.1),
i am routing call to freeswitch server (127.0.0.3).

Now I am trying to receive fax, my issue is when i try to send fax in
softphone(Zoiper) from the log i am seeing that it is sending fax using t30
codec. and i am not receiving the fax at destination, is it because of
codec, should it only work with t38 codec? if that is the issue than how am
i be able to send the fax using t38 from zoiper?

Here i am attaching the fs log with loglevel 9 and sip trace is also
enabled.

127.0.0.2  => carrier/provider IP
123456789 => Fax number
test at gamil.com => Email Address
127.0.0.4 =>UI IP

Pastebin link:https://pastebin.freeswitch.org/view/9ec52715
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170606/9f7e0396/attachment.html>

From deepikay at iiitd.ac.in  Tue Jun  6 12:34:51 2017
From: deepikay at iiitd.ac.in (Deepika Yadav)
Date: Tue, 6 Jun 2017 18:04:51 +0530
Subject: [Freeswitch-users] Fwd: How to get if of the playing media in a
	conference
In-Reply-To: <CAM99_Y=PnKDNU6tKgBpY5O+agVzJ1J-yMyQMhsSGGLj0Tp3SmA@mail.gmail.com>
References: <CAM99_Ykp2YCsoacCVEorN6ZQNrWNqw8+XU57m_7_c+Eh8_fYwg@mail.gmail.com>
 <CAM99_Y=PnKDNU6tKgBpY5O+agVzJ1J-yMyQMhsSGGLj0Tp3SmA@mail.gmail.com>
Message-ID: <CAM99_Yn2kV5PvRNh6uTJB2qSLN7eyn_9Xjja4Yddx8Tp47WBYA@mail.gmail.com>

---------- Forwarded message ----------
From: "Deepika Yadav" <deepikay at iiitd.ac.in>
Date: Jun 6, 2017 4:29 PM
Subject: How to get if of the playing media in a conference
To: <freeswitch-users-owner at lists.freeswitch.org>
Cc:

Hi,

I need to detect the start and stop of media in a conference call through
ESL. Event named as "play-file-done" and "play-file" are for detection of
any kind of media related activity. However, I could get find a method to
get the ID of the media to identify them.

e.g. event "play-file-done" is detected multiple times in a conference  -
if someone is added and also when any playing media is stopped.


-- 
Regards
Deepika
https://www.iiitd.edu.in/~deepikay/
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170606/8fb250ad/attachment.html>

From wsimon at stratusvideo.com  Tue Jun  6 13:21:36 2017
From: wsimon at stratusvideo.com (William Simon)
Date: Tue, 6 Jun 2017 13:21:36 +0000
Subject: [Freeswitch-users] X-RTP-STAT or P-RTP-STAT
Message-ID: <0362BB4B-BFDA-4BF5-B650-690AB6E35601@stratusvideo.com>

Can anyone explain how to add the X/P-RTP-STAT header to the BYE for summarizing RTP statistics at the end of the call?

Seems like there's something built in for this, but it's not clear how to use it.  https://wiki.freeswitch.org/wiki/Variable_sip_p_rtp_stat




"The information transmitted is intended only for the person or entity to which it is addressed and may contain proprietary, business-confidential and/or privileged material. If you are not the intended recipient of this message you are hereby notified that any use, review, retransmission, dissemination, distribution, reproduction or any action taken in reliance upon this message is prohibited. If you received this in error, please contact the sender and delete the material from any computer."
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170606/f5166733/attachment.html>

From brian at freeswitch.org  Tue Jun  6 14:00:06 2017
From: brian at freeswitch.org (Brian West)
Date: Tue, 6 Jun 2017 09:00:06 -0500
Subject: [Freeswitch-users] Fax receive issue with t30 codec
In-Reply-To: <CAHdn4vjxr5vVoO7Oc1P0u-xg6dJNYBhjxU=y5iz6VP5_JEcUjA@mail.gmail.com>
References: <CAHdn4vjxr5vVoO7Oc1P0u-xg6dJNYBhjxU=y5iz6VP5_JEcUjA@mail.gmail.com>
Message-ID: <CAEJMVkDtY8VeG+14cB42wHee5fjgzsXE5trA31dtp=F4rxoeRg@mail.gmail.com>

You'll have to use 1.6.17 if you ever want any faxing to work in all test
cases.

/b

On Tue, Jun 6, 2017 at 6:11 AM, Hardik Patel <hardikitpl at gmail.com> wrote:

> Hello,
>
> I am using opensips as entry point using dispatcher. opensips( 127.0.0.1),
> i am routing call to freeswitch server (127.0.0.3).
>
> Now I am trying to receive fax, my issue is when i try to send fax in
> softphone(Zoiper) from the log i am seeing that it is sending fax using t30
> codec. and i am not receiving the fax at destination, is it because of
> codec, should it only work with t38 codec? if that is the issue than how am
> i be able to send the fax using t38 from zoiper?
>
> Here i am attaching the fs log with loglevel 9 and sip trace is also
> enabled.
>
> 127.0.0.2  => carrier/provider IP
> 123456789 => Fax number
> test at gamil.com => Email Address
> 127.0.0.4 =>UI IP
>
> Pastebin link:https://pastebin.freeswitch.org/view/9ec52715
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org
>



-- 

*Brian West*
brian at freeswitch.org

*Twitter: @FreeSWITCH , @briankwest*

http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

Book a phone call (CST) <https://freeswitch.com/appointment>

Allison prompts for FreeSWITCH:

*https://www.gofundme.com/allison-prompts-for-freeswitch*
<https://www.gofundme.com/allison-prompts-for-freeswitch>

Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
/r/freeswitch <https://www.reddit.com/r/freeswitch>

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*Skype:*briankwest
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170606/595e10b5/attachment.html>

From mike at jerris.com  Tue Jun  6 14:08:19 2017
From: mike at jerris.com (Michael Jerris)
Date: Tue, 6 Jun 2017 10:08:19 -0400
Subject: [Freeswitch-users] X-RTP-STAT or P-RTP-STAT
In-Reply-To: <0362BB4B-BFDA-4BF5-B650-690AB6E35601@stratusvideo.com>
References: <0362BB4B-BFDA-4BF5-B650-690AB6E35601@stratusvideo.com>
Message-ID: <E97C7330-7E9D-47B7-9096-55D0AFD23FCE@jerris.com>

Those vars are for getting those stats that devices send us in a BYE

> On Jun 6, 2017, at 9:21 AM, William Simon <wsimon at stratusvideo.com> wrote:
> 
> Can anyone explain how to add the X/P-RTP-STAT header to the BYE for summarizing RTP statistics at the end of the call?
> 
> Seems like there's something built in for this, but it's not clear how to use it.  https://wiki.freeswitch.org/wiki/Variable_sip_p_rtp_stat <https://wiki.freeswitch.org/wiki/Variable_sip_p_rtp_stat>
> 
> 
> 
> 
> “The information transmitted is intended only for the person or entity to which it is addressed and may contain proprietary, business-confidential and/or privileged material. If you are not the intended recipient of this message you are hereby notified that any use, review, retransmission, dissemination, distribution, reproduction or any action taken in reliance upon this message is prohibited. If you received this in error, please contact the sender and delete the material from any computer.”

Sorry, nothing you send to the list is private.  it will be distributed.  Note, we will NOT remove messages from the list archives if requested (yes, its been requested repeatedly in the past).

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170606/43992858/attachment-0001.html>

From wsimon at stratusvideo.com  Tue Jun  6 14:25:43 2017
From: wsimon at stratusvideo.com (William Simon)
Date: Tue, 6 Jun 2017 14:25:43 +0000
Subject: [Freeswitch-users] X-RTP-STAT or P-RTP-STAT
In-Reply-To: <E97C7330-7E9D-47B7-9096-55D0AFD23FCE@jerris.com>
References: <0362BB4B-BFDA-4BF5-B650-690AB6E35601@stratusvideo.com>
 <E97C7330-7E9D-47B7-9096-55D0AFD23FCE@jerris.com>
Message-ID: <C7FA7441-A4B1-40DC-A530-070D8F80DB99@stratusvideo.com>

Does FreeSWITCH have the capability to generate the stats? We would like to get a summary RTP quality report from FS's perspective.

On Jun 6, 2017, at 10:08 AM, Michael Jerris <mike at jerris.com<mailto:mike at jerris.com>> wrote:

Those vars are for getting those stats that devices send us in a BYE

On Jun 6, 2017, at 9:21 AM, William Simon <wsimon at stratusvideo.com<mailto:wsimon at stratusvideo.com>> wrote:

Can anyone explain how to add the X/P-RTP-STAT header to the BYE for summarizing RTP statistics at the end of the call?

Seems like there's something built in for this, but it's not clear how to use it.  https://wiki.freeswitch.org/wiki/Variable_sip_p_rtp_stat




“The information transmitted is intended only for the person or entity to which it is addressed and may contain proprietary, business-confidential and/or privileged material. If you are not the intended recipient of this message you are hereby notified that any use, review, retransmission, dissemination, distribution, reproduction or any action taken in reliance upon this message is prohibited. If you received this in error, please contact the sender and delete the material from any computer.”

Sorry, nothing you send to the list is private.  it will be distributed.  Note, we will NOT remove messages from the list archives if requested (yes, its been requested repeatedly in the past).



“The information transmitted is intended only for the person or entity to which it is addressed and may contain proprietary, business-confidential and/or privileged material. If you are not the intended recipient of this message you are hereby notified that any use, review, retransmission, dissemination, distribution, reproduction or any action taken in reliance upon this message is prohibited. If you received this in error, please contact the sender and delete the material from any computer.”
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.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
http://www.freeswitch.org



“The information transmitted is intended only for the person or entity to which it is addressed and may contain proprietary, business-confidential and/or privileged material. If you are not the intended recipient of this message you are hereby notified that any use, review, retransmission, dissemination, distribution, reproduction or any action taken in reliance upon this message is prohibited. If you received this in error, please contact the sender and delete the material from any computer.”
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170606/fad177de/attachment.html>

From acheraime at gmail.com  Tue Jun  6 14:30:59 2017
From: acheraime at gmail.com (acheraime at gmail.com)
Date: Tue, 6 Jun 2017 10:30:59 -0400
Subject: [Freeswitch-users] X-RTP-STAT or P-RTP-STAT
In-Reply-To: <C7FA7441-A4B1-40DC-A530-070D8F80DB99@stratusvideo.com>
References: <0362BB4B-BFDA-4BF5-B650-690AB6E35601@stratusvideo.com>
 <E97C7330-7E9D-47B7-9096-55D0AFD23FCE@jerris.com>
 <C7FA7441-A4B1-40DC-A530-070D8F80DB99@stratusvideo.com>
Message-ID: <2D844020-558B-4E1E-90CE-3B1B95322BA3@gmail.com>

The CDR generated after each call contains a lot of RTP related information include the MOS.


Sent from my iPhone

> On Jun 6, 2017, at 10:25 AM, William Simon <wsimon at stratusvideo.com> wrote:
> 
> Does FreeSWITCH have the capability to generate the stats? We would like to get a summary RTP quality report from FS's perspective.
> 
>> On Jun 6, 2017, at 10:08 AM, Michael Jerris <mike at jerris.com> wrote:
>> 
>> Those vars are for getting those stats that devices send us in a BYE
>> 
>>> On Jun 6, 2017, at 9:21 AM, William Simon <wsimon at stratusvideo.com> wrote:
>>> 
>>> Can anyone explain how to add the X/P-RTP-STAT header to the BYE for summarizing RTP statistics at the end of the call?
>>> 
>>> Seems like there's something built in for this, but it's not clear how to use it.  https://wiki.freeswitch.org/wiki/Variable_sip_p_rtp_stat
>>> 
>>> 
>>> 
>>> 
>>> “The information transmitted is intended only for the person or entity to which it is addressed and may contain proprietary, business-confidential and/or privileged material. If you are not the intended recipient of this message you are hereby notified that any use, review, retransmission, dissemination, distribution, reproduction or any action taken in reliance upon this message is prohibited. If you received this in error, please contact the sender and delete the material from any computer.”
>> 
>> Sorry, nothing you send to the list is private.  it will be distributed.  Note, we will NOT remove messages from the list archives if requested (yes, its been requested repeatedly in the past).
>> 
>> 
>> 
>> “The information transmitted is intended only for the person or entity to which it is addressed and may contain proprietary, business-confidential and/or privileged material. If you are not the intended recipient of this message you are hereby notified that any use, review, retransmission, dissemination, distribution, reproduction or any action taken in reliance upon this message is prohibited. If you received this in error, please contact the sender and delete the material from any computer.”
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.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
>> http://www.freeswitch.org
> 
> 
> 
> “The information transmitted is intended only for the person or entity to which it is addressed and may contain proprietary, business-confidential and/or privileged material. If you are not the intended recipient of this message you are hereby notified that any use, review, retransmission, dissemination, distribution, reproduction or any action taken in reliance upon this message is prohibited. If you received this in error, please contact the sender and delete the material from any computer.”
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170606/ea93328d/attachment.html>

From mike at jerris.com  Tue Jun  6 14:31:50 2017
From: mike at jerris.com (Michael Jerris)
Date: Tue, 6 Jun 2017 10:31:50 -0400
Subject: [Freeswitch-users] X-RTP-STAT or P-RTP-STAT
In-Reply-To: <C7FA7441-A4B1-40DC-A530-070D8F80DB99@stratusvideo.com>
References: <0362BB4B-BFDA-4BF5-B650-690AB6E35601@stratusvideo.com>
 <E97C7330-7E9D-47B7-9096-55D0AFD23FCE@jerris.com>
 <C7FA7441-A4B1-40DC-A530-070D8F80DB99@stratusvideo.com>
Message-ID: <A3160963-4FA3-4447-90F0-7020A00E5996@jerris.com>

We don’t send those in BYE, some stats are recorded in cdr.

> On Jun 6, 2017, at 10:25 AM, William Simon <wsimon at stratusvideo.com> wrote:
> 
> Does FreeSWITCH have the capability to generate the stats? We would like to get a summary RTP quality report from FS's perspective.
> 
>> On Jun 6, 2017, at 10:08 AM, Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>> wrote:
>> 
>> Those vars are for getting those stats that devices send us in a BYE
>> 
>>> On Jun 6, 2017, at 9:21 AM, William Simon <wsimon at stratusvideo.com <mailto:wsimon at stratusvideo.com>> wrote:
>>> 
>>> Can anyone explain how to add the X/P-RTP-STAT header to the BYE for summarizing RTP statistics at the end of the call?
>>> 
>>> Seems like there's something built in for this, but it's not clear how to use it.  https://wiki.freeswitch.org/wiki/Variable_sip_p_rtp_stat <https://wiki.freeswitch.org/wiki/Variable_sip_p_rtp_stat>
>>> 
>>> 
>>> 
>>> 
>>> “The information transmitted is intended only for the person or entity to which it is addressed and may contain proprietary, business-confidential and/or privileged material. If you are not the intended recipient of this message you are hereby notified that any use, review, retransmission, dissemination, distribution, reproduction or any action taken in reliance upon this message is prohibited. If you received this in error, please contact the sender and delete the material from any computer.”
>> 
>> Sorry, nothing you send to the list is private.  it will be distributed.  Note, we will NOT remove messages from the list archives if requested (yes, its been requested repeatedly in the past).
>> 
>> 
>> 
>> “The information transmitted is intended only for the person or entity to which it is addressed and may contain proprietary, business-confidential and/or privileged material. If you are not the intended recipient of this message you are hereby notified that any use, review, retransmission, dissemination, distribution, reproduction or any action taken in reliance upon this message is prohibited. If you received this in error, please contact the sender and delete the material from any computer.”
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>> http://www.freeswitchsolutions.com
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.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
>> http://www.freeswitch.org
> 
> 
> 
> “The information transmitted is intended only for the person or entity to which it is addressed and may contain proprietary, business-confidential and/or privileged material. If you are not the intended recipient of this message you are hereby notified that any use, review, retransmission, dissemination, distribution, reproduction or any action taken in reliance upon this message is prohibited. If you received this in error, please contact the sender and delete the material from any computer.”
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170606/73115f12/attachment-0001.html>

From anthony.minessale at gmail.com  Tue Jun  6 16:26:50 2017
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 6 Jun 2017 11:26:50 -0500
Subject: [Freeswitch-users] Monitoring call quality using
	variable_rtp_audio_xxx
In-Reply-To: <DB3PR02MB04274C1149CD867FA5D8714ABDCB0@DB3PR02MB0427.eurprd02.prod.outlook.com>
References: <DB3PR02MB04275464611C567EBFBCA711BDF10@DB3PR02MB0427.eurprd02.prod.outlook.com>
 <DB3PR02MB04274C1149CD867FA5D8714ABDCB0@DB3PR02MB0427.eurprd02.prod.outlook.com>
Message-ID: <CAKbxfG_9jxHCWvwnxyBX75bb+fTYmAXkcixwa16aud5Sf8c-_Q@mail.gmail.com>

flaws are missing or out-of-order packets.
If you get subsequent missing packets it counts as 2 flaws.

The number of flaws out of the total number of packets is the quality
percentage.  The quality percentage rounded to a scale of 4.5 is the mos.


On Tue, Jun 6, 2017 at 1:59 AM, Jonathan Hunter <hunterj91 at hotmail.com>
wrote:

> Hi Guys,
>
>
> To be more specific, what is the meaning of this parameter?
>
>
> in_flaw_total
>
>
> I am becoming clear on the other parameters!
>
>
> Thanks
>
>
> Jon
>
>
> ------------------------------
> *From:* Jonathan Hunter <hunterj91 at hotmail.com>
> *Sent:* 31 May 2017 19:24
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Monitoring call quality using variable_rtp_audio_xxx
>
>
> Hi Guys,
>
> Sorry for the noise, we are looking to poll calls in progress to grab the
> rtp_audio variables when a call is in progress, and we want to understand
> if this is a good approach and to that end what values/variables we should
> consider and what ranges should we be working with?
>
> I understand mos and quality percentage but what other values are a good
> indicator?
>
> Again Im clear on jitter and its meaning just want to understand whats
> what as not everything understandably is not documented.
>
> Many thanks
>
> Jon
>
>     "variable_rtp_audio_recv_pt": "8",
>     "variable_rtp_audio_in_raw_bytes": "374960",
>     "variable_rtp_audio_in_media_bytes": "374616",
>     "variable_rtp_audio_in_packet_count": "2180",
>     "variable_rtp_audio_in_media_packet_count": "2178",
>     "variable_rtp_audio_in_skip_packet_count": "92",
>     "variable_rtp_audio_in_jitter_packet_count": "0",
>     "variable_rtp_audio_in_dtmf_packet_count": "0",
>     "variable_rtp_audio_in_cng_packet_count": "0",
>     "variable_rtp_audio_in_flush_packet_count": "2",
>     "variable_rtp_audio_in_largest_jb_size": "0",
>     "variable_rtp_audio_in_jitter_min_variance": "28.57",
>     "variable_rtp_audio_in_jitter_max_variance": "116.33",
>     "variable_rtp_audio_in_jitter_loss_rate": "0.02",
>     "variable_rtp_audio_in_jitter_burst_rate": "0.98",
>     "variable_rtp_audio_in_mean_interval": "20.39",
>     "variable_rtp_audio_in_flaw_total": "43",
>     "variable_rtp_audio_in_quality_percentage": "97.00",
>     "variable_rtp_audio_in_mos": "4.47",
>     "variable_rtp_audio_out_raw_bytes": "210012",
>     "variable_rtp_audio_out_media_bytes": "210012",
>     "variable_rtp_audio_out_packet_count": "1221",
>     "variable_rtp_audio_out_media_packet_count": "1221",
>     "variable_rtp_audio_out_skip_packet_count": "0",
>     "variable_rtp_audio_out_dtmf_packet_count": "0",
>     "variable_rtp_audio_out_cng_packet_count": "0",
>     "variable_rtp_audio_rtcp_packet_count": "0",
>     "variable_rtp_audio_rtcp_octet_count": "0",
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org
>



-- 
Anthony Minessale II       ♬ @anthmfs  ♬ @FreeSWITCH  ♬

☞ http://freeswitch.org/  ☞ http://cluecon.com/  ☞
http://twitter.com/FreeSWITCH
☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+
<http://freeswitch.org/g+>*

ClueCon Weekly Development Call
☎ sip:888 at conference.freeswitch.org  ☎ +19193869900

https://www.youtube.com/watch?v=9XXgW34t40s
https://www.youtube.com/watch?v=NLaDpGQuZDA
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170606/060e37af/attachment.html>

From mjlopez at smartic.es  Tue Jun  6 17:38:30 2017
From: mjlopez at smartic.es (=?iso-8859-1?Q?Miguel_Jes=FAs_L=F3pez_Valverde?=)
Date: Tue, 6 Jun 2017 19:38:30 +0200
Subject: [Freeswitch-users] trouble with instalation of mod_fail2ban
	module in FreeSwitch.
In-Reply-To: <52bacbc4-bc14-e9c9-d938-da1cc789f357@madovsky.org>
References: <010a01d2daf0$36c6cc50$a45464f0$@smartic.es>
 <52bacbc4-bc14-e9c9-d938-da1cc789f357@madovsky.org>
Message-ID: <0ff201d2deeb$b6fb5090$24f1f1b0$@smartic.es>

Thank you very much. Now is running right.

 

 

 

De: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org]
En nombre de Madovsky
Enviado el: viernes, 02 de junio de 2017 23:59
Para: freeswitch-users at lists.freeswitch.org
Asunto: Re: [Freeswitch-users] trouble with instalation of mod_fail2ban
module in FreeSwitch.

 

 

On 6/1/2017 9:00 AM, Miguel Jesús López Valverde wrote:

Hello:

 

I am sending you this email because I am having problems installing the
module mod_fail2ban following the recipe offered at
https://freeswitch.org/confluence/display/FREESWITCH/mod_fail2ban

because when I run make, I get this result:

 

/usr/local/freeswitch/mod/applications/mod_fail2ban$ sudo make

Makefile:2: ../../../../build/modmake.rules: No such file or directory

make: *** No rule to make target '../../../../build/modmake.rules'.  Stop.

 

Do you know in which folder the sentence “git clone” may to be executed?. Do
you know if there is anything else to keep in mind that can cause this
problem?

 

Thank you very much.

 

Miguel J. Lopez.

 


 
<https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campai
gn=sig-email&utm_content=emailclient> 

Libre de virus.
<https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campai
gn=sig-email&utm_content=emailclient> www.avast.com 





did you install fail2ban from your distro?



---
El software de antivirus Avast ha analizado este correo electrónico en busca de virus.
https://www.avast.com/antivirus
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170606/88f31194/attachment.html>

From xxxman2008 at 126.com  Wed Jun  7 07:42:04 2017
From: xxxman2008 at 126.com (Raymond)
Date: Wed, 7 Jun 2017 15:42:04 +0800 (CST)
Subject: [Freeswitch-users] question about HA solution
In-Reply-To: <CADcrYGLbN1Ut6riF1Lf7UvwG9ypvR2ekWxt1m=e5BmqOzP_c=w@mail.gmail.com>
References: <CADcrYG+RHGQ2fC7dSKK9x9WfVvYcTczOtbppCLgttRCfMGqArA@mail.gmail.com>
 <CAFLo2QxxbZ5Ged0J0p9_VfQ2nkGtuV0CDp9G763EjjuvqUv8iw@mail.gmail.com>
 <CADcrYG+sO0+WYUtYVN6pkpU-5SwyfurfshdUdhxb-c4Az1s49Q@mail.gmail.com>
 <8A13E0EC-FA13-4267-80F6-CE1A8E8360CF@jerris.com>
 <CADcrYGLbN1Ut6riF1Lf7UvwG9ypvR2ekWxt1m=e5BmqOzP_c=w@mail.gmail.com>
Message-ID: <55e4ea34.7356.15c8180dc29.Coremail.xxxman2008@126.com>

Ok, Denys , I understand you have an "Master - Slave" situation.  So ,you  must move the call back when Master server come back.


I'm sorry ,no help for u ,but ,maybe ,use an " master - master - master -...."  architecture is the fastest way to resolve the problem.  :-)) .


The key of your question is "auto-clear-sql" option  not functioning normally . plz double check your config ,and make sure , all servers have same option.


Raymond


在 2017-06-06 01:50:18，"Denys Pozniak" <denys.pozniak at crazycall.com> 写道：
Yes, correct. But when you restart FS on slave, it will erase database. And option auto-clear-sql=false not working for me.



On Jun 5, 2017 6:32 PM, "Michael Jerris" <mike at jerris.com> wrote:

recovered calls will get new entries in the table.

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170607/89f3b8f3/attachment-0001.html>

From hunterj91 at hotmail.com  Wed Jun  7 08:50:09 2017
From: hunterj91 at hotmail.com (Jonathan Hunter)
Date: Wed, 7 Jun 2017 08:50:09 +0000
Subject: [Freeswitch-users] Monitoring call quality
	using	variable_rtp_audio_xxx
In-Reply-To: <CAKbxfG_9jxHCWvwnxyBX75bb+fTYmAXkcixwa16aud5Sf8c-_Q@mail.gmail.com>
References: <DB3PR02MB04275464611C567EBFBCA711BDF10@DB3PR02MB0427.eurprd02.prod.outlook.com>
 <DB3PR02MB04274C1149CD867FA5D8714ABDCB0@DB3PR02MB0427.eurprd02.prod.outlook.com>,
 <CAKbxfG_9jxHCWvwnxyBX75bb+fTYmAXkcixwa16aud5Sf8c-_Q@mail.gmail.com>
Message-ID: <DB3PR02MB042700D27FD408DD27F3D8B1BDC80@DB3PR02MB0427.eurprd02.prod.outlook.com>

Perfect thank you!


________________________________
From: FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> on behalf of Anthony Minessale <anthony.minessale at gmail.com>
Sent: 06 June 2017 16:26
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Monitoring call quality using variable_rtp_audio_xxx

flaws are missing or out-of-order packets.
If you get subsequent missing packets it counts as 2 flaws.

The number of flaws out of the total number of packets is the quality percentage.  The quality percentage rounded to a scale of 4.5 is the mos.


On Tue, Jun 6, 2017 at 1:59 AM, Jonathan Hunter <hunterj91 at hotmail.com<mailto:hunterj91 at hotmail.com>> wrote:

Hi Guys,


To be more specific, what is the meaning of this parameter?


in_flaw_total


I am becoming clear on the other parameters!


Thanks


Jon


________________________________
From: Jonathan Hunter <hunterj91 at hotmail.com<mailto:hunterj91 at hotmail.com>>
Sent: 31 May 2017 19:24
To: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
Subject: Monitoring call quality using variable_rtp_audio_xxx


Hi Guys,

Sorry for the noise, we are looking to poll calls in progress to grab the rtp_audio variables when a call is in progress, and we want to understand if this is a good approach and to that end what values/variables we should consider and what ranges should we be working with?

I understand mos and quality percentage but what other values are a good indicator?

Again Im clear on jitter and its meaning just want to understand whats what as not everything understandably is not documented.

Many thanks

Jon

    "variable_rtp_audio_recv_pt": "8",
    "variable_rtp_audio_in_raw_bytes": "374960",
    "variable_rtp_audio_in_media_bytes": "374616",
    "variable_rtp_audio_in_packet_count": "2180",
    "variable_rtp_audio_in_media_packet_count": "2178",
    "variable_rtp_audio_in_skip_packet_count": "92",
    "variable_rtp_audio_in_jitter_packet_count": "0",
    "variable_rtp_audio_in_dtmf_packet_count": "0",
    "variable_rtp_audio_in_cng_packet_count": "0",
    "variable_rtp_audio_in_flush_packet_count": "2",
    "variable_rtp_audio_in_largest_jb_size": "0",
    "variable_rtp_audio_in_jitter_min_variance": "28.57",
    "variable_rtp_audio_in_jitter_max_variance": "116.33",
    "variable_rtp_audio_in_jitter_loss_rate": "0.02",
    "variable_rtp_audio_in_jitter_burst_rate": "0.98",
    "variable_rtp_audio_in_mean_interval": "20.39",
    "variable_rtp_audio_in_flaw_total": "43",
    "variable_rtp_audio_in_quality_percentage": "97.00",
    "variable_rtp_audio_in_mos": "4.47",
    "variable_rtp_audio_out_raw_bytes": "210012",
    "variable_rtp_audio_out_media_bytes": "210012",
    "variable_rtp_audio_out_packet_count": "1221",
    "variable_rtp_audio_out_media_packet_count": "1221",
    "variable_rtp_audio_out_skip_packet_count": "0",
    "variable_rtp_audio_out_dtmf_packet_count": "0",
    "variable_rtp_audio_out_cng_packet_count": "0",
    "variable_rtp_audio_rtcp_packet_count": "0",
    "variable_rtp_audio_rtcp_octet_count": "0",


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.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
http://www.freeswitch.org



--
Anthony Minessale II       ♬ @anthmfs  ♬ @FreeSWITCH  ♬

☞ http://freeswitch.org/  ☞ http://cluecon.com/  ☞ http://twitter.com/FreeSWITCH
☞ irc.freenode.net<http://irc.freenode.net> #freeswitch ☞ http://freeswitch.org/g+

ClueCon Weekly Development Call
☎ sip:888 at conference.freeswitch.org<mailto:sip%3A888 at conference.freeswitch.org>  ☎ +19193869900

https://www.youtube.com/watch?v=9XXgW34t40s
https://www.youtube.com/watch?v=NLaDpGQuZDA
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170607/b2df6c53/attachment.html>

From boesemar at gmail.com  Wed Jun  7 09:22:45 2017
From: boesemar at gmail.com (boesemar .)
Date: Wed, 7 Jun 2017 10:22:45 +0100
Subject: [Freeswitch-users] Enabling new languages for Say
In-Reply-To: <CAPLWHmjLiKSk_hobMzSbOGXq9iu4gMFaeNdq-SwN3QPJvpJq_g@mail.gmail.com>
References: <CAPLWHmjLiKSk_hobMzSbOGXq9iu4gMFaeNdq-SwN3QPJvpJq_g@mail.gmail.com>
Message-ID: <CAPLWHmhnm+GfVX4DnnnQAUXw-MGq-UaND63S8yt8qRBBSYj5EQ@mail.gmail.com>

Fixed!

It seems that just say_string on the CLI is only available for en and ru.
Also mentioned here:
https://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_say

Application 'say' from the dialplan is in fact working... all good :)

Martin

On Mon, Jun 5, 2017 at 10:53 AM, boesemar . <boesemar at gmail.com> wrote:

> Hi Brian,
>
> mod_say_pt is definitely loaded:
>
> > module_exists mod_say_pt
> true
>
> Can you think of any other cause for the "Invalid SAY Interface [pt]" ?
>
> Martin
>
>
> PS: sorry for breaking the thread - problem with email
>
>
>
> Brian West brian at freeswitch.org
> Fri Jun 2 14:51:25 UTC 2017
>
> I would guess mod_say_pt is not loaded.
>
> On Fri, Jun 2, 2017 at 4:54 AM, Martin Boese <martin at maxnet.ao <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>> wrote:
>
> >* Hi!
> *>>* FreeSWITCH Version 1.6.15-32-bec4538~64bit (-32-bec4538 64bit)
> *>* Debian Jessie. Vanilla config.
> *>>* I am trying to enable portuguese to say numbers using dptools "Say".
> *>>* This is what I did:
> *>*  - Module mod_say_pt is loaded
> *>*  - in freeswitch.xml
> *>*    include lang/pt/pt_PT.xml
> *>*  - Downloaded sounds from
> *>*    https://github.com/jpawlowski/freeswitch-sounds-ttsand <https://github.com/jpawlowski/freeswitch-sounds-ttsand> placed them
> *>*    into /usr/share/freeswitch/sounds
> *>*    Directory structure is now like: /pt/tts/google/ascii/16000/...
> *>>* CLI> say_string t.wav pt NUMBER pronounced 123
> *>* [ERR] switch_xml.c:3180 Can't find phrases tag
> *>>* ..I found out that lang/pt_PT.xml seems to be missing the
> *>* <phrases><macros> tags within the <language> tag (vanilla config). I
> *>* fixed that.
> *>>* Now:
> *>* CLI> say_string t.wav pt NUMBER pronounced 123
> *>* [ERR] switch_ivr.c:3726 Invalid SAY Interface [pt]!
> *>>* BTW: English works fine:
> *>* CLI> say_string t.wav en NUMBER pronounced 123
> *>* file_string://digits/1.wav!digits/hundred.wav!digits/20.wav!digits/3.wav
> *>>* I also tried other languages but have same error "Invalid SAY
> *>* Interface".
> *>>* What am I missing. Please help.
> *>>* Thanks,
> *>* Martin
> *>>* _________________________________________________________________________
> *>* Professional FreeSWITCH Consulting Services:
> *>* consulting at freeswitch.org <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> *>* http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com>
> *>>* Official FreeSWITCH Sites
> *>* http://www.freeswitch.org <http://www.freeswitch.org>
> *>* http://confluence.freeswitch.org <http://confluence.freeswitch.org>
> *>* http://www.cluecon.com <http://www.cluecon.com>
> *>>* FreeSWITCH-users mailing list
> *>* FreeSWITCH-users at lists.freeswitch.org <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> *>* 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>
> *>* http://www.freeswitch.org <http://www.freeswitch.org>
> *
>
>
>
> --
>
> *Brian West*brian at freeswitch.org <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>
> *Twitter: @FreeSWITCH , @briankwest*
> http://www.freeswitchbook.comhttp://www.freeswitchcookbook.com
>
> Book a phone call (CST) <https://freeswitch.com/appointment>
>
> Allison prompts for FreeSWITCH:
>
> *https://www.gofundme.com/allison-prompts-for-freeswitch*
> <https://www.gofundme.com/allison-prompts-for-freeswitch>
>
> Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
> /r/freeswitch <https://www.reddit.com/r/freeswitch>
>
> *T:*+19184209001 <%28918%29%20420-9001> | *F:*+19184209002 <%28918%29%20420-9002> | *M:*+1918424WEST (9378)
> *Skype:*briankwest
> -------------- next part --------------
> An HTML attachment was scrubbed...
> URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170602/d72352fe/attachment.html>
>
> ------------------------------
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170607/10fb3373/attachment-0001.html>

From Alexander.Haugg at c4b.de  Wed Jun  7 10:49:30 2017
From: Alexander.Haugg at c4b.de (Alexander Haugg)
Date: Wed, 7 Jun 2017 10:49:30 +0000
Subject: [Freeswitch-users] Turn configuration mod_sofia
Message-ID: <26962017ab7e4ba88fbd2db5a551a34d@c4b.de>

Hi,

my question:
how can I configure an ICE server (turn) in the sip profile?

My think for this id case I need the possibility to configure the turn server with credentials in the sip profile like the stun configuration too.

-          STUN configuration "<param name="ext-rtp-ip" value=" stun:stun.freeswitch.org"/>" -> that's working!

-          ICE configuration could be? "<param name="ext-rtp-ip" value=" { 'stun:stun.freeswitch.org', 'turn:turn.myserver.com;username=xxx;password=123'}"/>"

Is there a possibility?

Thanks a lot

Von: Alexander Haugg
Gesendet: Donnerstag, 1. Juni 2017 15:24
An: 'freeswitch-users at lists.freeswitch.org' <freeswitch-users at lists.freeswitch.org>
Betreff: WG: Turn configuration mod_sofia

Hi Brian,

we use the Freeswitch as "Man in the Middle" for WebRTC.

The WebRTC Clients (ICE Link Stack from Frozen Mountain) registered via SIP on the Freeswitch.
The SIP signalling is a special szenario and works successfully in several LAN WAN setups.
But now we need srflx (that's fine with teh STUN configuration) an relay candidates in the SDP that's ganerated by the Freeswitch.

My think is, for this case i need the possibility to configure the turn server with credentials in the sip profile like the stun configuration too.

-          STUN configuration "<param name="ext-rtp-ip" value=" stun:stun.freeswitch.org"/>"

-          ICE configuration could be? "<param name="ext-rtp-ip" value=" { 'stun:stun.freeswitch.org', 'turn:turn.myserver.com;username=xxx;password=123'}"/>"

Thanks a lot


>From brian at freeswitch.org  Tue May 30 13:50:32 2017
From: brian at freeswitch.org (Brian West)
Date: Tue, 30 May 2017 08:50:32 -0500
Subject: [Freeswitch-users] Turn configuration mod_sofia
In-Reply-To: <a9e067a18e104afbb53e207712467baf at c4b.de<mailto:a9e067a18e104afbb53e207712467baf at c4b.de>>
References: <a9e067a18e104afbb53e207712467baf at c4b.de<mailto:a9e067a18e104afbb53e207712467baf at c4b.de>>
Message-ID: <CAEJMVkDy5e2COC8cH=xV=ozM0gepBS3cQ_YxUPQk4twMoaSVmg at mail.gmail.com<mailto:CAEJMVkDy5e2COC8cH=xV=ozM0gepBS3cQ_YxUPQk4twMoaSVmg at mail.gmail.com>>

There is little reason to use TURN when speaking to FreeSWITCH, What issue
are you trying to solve?


Von: Alexander Haugg
Gesendet: Montag, 29. Mai 2017 18:25
An: 'freeswitch-users at lists.freeswitch.org' <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
Betreff: Turn configuration mod_sofia

Hi all,

the stun configuration for the mod_sofia profile is very easy "<param name="ext-rtp-ip" value=" stun:stun.freeswitch.org"/>"
but what is to do, if i need a relay candidate in the sdp? How can i set the turn address and the login credentials?

Thanks a lot
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170607/deb1d07d/attachment.html>

From khorsmann at gmail.com  Wed Jun  7 11:47:03 2017
From: khorsmann at gmail.com (Karsten Horsmann)
Date: Wed, 7 Jun 2017 13:47:03 +0200
Subject: [Freeswitch-users] Multiple FreeSWITCH servers behind
	kamailio-websocket
Message-ID: <CAFArqsZjjVP63_OscW=qVOh_skaddndKuXzKaWJp2WgHg9KaEw@mail.gmail.com>

Hello List,


is there any howto about webrtc loadbalance in combination with kamailio
and FreeSWITCH?

I want to share one WSS address/endpoint to multiple FreeSWITCH backends.
Or is there any other best practice?

My callflow is mostly that my internal SIP Servers called my registered
webrtc clients.

Would be nice to get some input.

-- 
Kind Regards
*Karsten Horsmann*
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170607/1c352af1/attachment.html>

From mike at jerris.com  Wed Jun  7 15:35:25 2017
From: mike at jerris.com (Michael Jerris)
Date: Wed, 7 Jun 2017 11:35:25 -0400
Subject: [Freeswitch-users] question about HA solution
In-Reply-To: <CADcrYGLbN1Ut6riF1Lf7UvwG9ypvR2ekWxt1m=e5BmqOzP_c=w@mail.gmail.com>
References: <CADcrYG+RHGQ2fC7dSKK9x9WfVvYcTczOtbppCLgttRCfMGqArA@mail.gmail.com>
 <CAFLo2QxxbZ5Ged0J0p9_VfQ2nkGtuV0CDp9G763EjjuvqUv8iw@mail.gmail.com>
 <CADcrYG+sO0+WYUtYVN6pkpU-5SwyfurfshdUdhxb-c4Az1s49Q@mail.gmail.com>
 <8A13E0EC-FA13-4267-80F6-CE1A8E8360CF@jerris.com>
 <CADcrYGLbN1Ut6riF1Lf7UvwG9ypvR2ekWxt1m=e5BmqOzP_c=w@mail.gmail.com>
Message-ID: <E55363B0-487F-4F65-8F3F-78620AB1AB73@jerris.com>

That param should keep it from doing so, if its not you are not setting it somehow or something else is wiping the db.

> On Jun 5, 2017, at 1:50 PM, Denys Pozniak <denys.pozniak at crazycall.com> wrote:
> 
> Yes, correct. But when you restart FS on slave, it will erase database. And option auto-clear-sql=false not working for me.
> 
> On Jun 5, 2017 6:32 PM, "Michael Jerris" <mike at jerris.com <mailto:mike at jerris.com>> wrote:
> recovered calls will get new entries in the table.
> 
>> On Jun 5, 2017, at 7:41 AM, Denys Pozniak <denys.pozniak at crazycall.com <mailto:denys.pozniak at crazycall.com>> wrote:
>> 
>> Hello!
>> 
>> Thank you Raymond about your explanation, but I dont agree with some point: 
>> If it really need an answer about your question -- "if it is possible to move calls back".   I think it's unnecessary.  - in my case I have two not equal servers, so I need to have only one as a master.
>> If switchover happens I need to have ability to restore master back.
>> 
>> Thank you Luis for your link, you can do simple test to understand what I am talking about: do call -> check on master and slave #show channels -> restart FS on slave -> check on master #show channels. In my case I dont see any active calls after this, so restoring back is not possible.
>> 
>> 
>> 
>> On 3 June 2017 at 22:16, Luis Daniel Lucio Quiroz <luis.daniel.lucio at gmail.com <mailto:luis.daniel.lucio at gmail.com>> wrote:
>> You may want to read this article.
>> 
>> http://inside-out.xyz/technology/how-to-configure-freeswitch-for-ha.html <http://inside-out.xyz/technology/how-to-configure-freeswitch-for-ha.html>
>> 
>> Le 31 mai 2017 6:29 PM, "Denys Pozniak" <denys.pozniak at crazycall.com <mailto:denys.pozniak at crazycall.com>> a écrit :
>> Hello!
>> 
>> I built FS HA solution based on keepalived and mysql master-master.
>> It works ok generally, but as I understand FS after restarting cleaning own database.
>> 
>> So when node1 fails calls jump to node2, after script restarts node1 it is not possible to move calls back.
>> 
>> Tried options in switch.conf.xml, but no luck:
>> 
>>     <param name="auto-create-schemas" value="false"/>
>>     <param name="auto-clear-sql" value="false"/>
>> 
>> Is there is a way to solve this?
>> 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170607/34fb35dc/attachment.html>

From mike at jerris.com  Wed Jun  7 15:41:53 2017
From: mike at jerris.com (Michael Jerris)
Date: Wed, 7 Jun 2017 11:41:53 -0400
Subject: [Freeswitch-users] Enabling new languages for Say
In-Reply-To: <CAPLWHmhnm+GfVX4DnnnQAUXw-MGq-UaND63S8yt8qRBBSYj5EQ@mail.gmail.com>
References: <CAPLWHmjLiKSk_hobMzSbOGXq9iu4gMFaeNdq-SwN3QPJvpJq_g@mail.gmail.com>
 <CAPLWHmhnm+GfVX4DnnnQAUXw-MGq-UaND63S8yt8qRBBSYj5EQ@mail.gmail.com>
Message-ID: <B6E1690F-B406-44CA-B45B-77B9906AF096@jerris.com>

Thats weird, its in more than en and ru, its also in es_ar, he, pl, and sv.

Not sure why whoever wrote the other language mods never added that part.  If anyone wants to toss me a pull request on fixing this, I’d be happy to look at it.

> On Jun 7, 2017, at 5:22 AM, boesemar . <boesemar at gmail.com> wrote:
> 
> Fixed!
> 
> It seems that just say_string on the CLI is only available for en and ru. Also mentioned here:
> https://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_say <https://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_say>
> 
> Application 'say' from the dialplan is in fact working... all good :)
> 
> Martin
> 
> On Mon, Jun 5, 2017 at 10:53 AM, boesemar . <boesemar at gmail.com <mailto:boesemar at gmail.com>> wrote:
> Hi Brian,
> 
> mod_say_pt is definitely loaded:
> 
> > module_exists mod_say_pt
> true
> 
> Can you think of any other cause for the "Invalid SAY Interface [pt]" ?
> Martin
> 
> PS: sorry for breaking the thread - problem with email
> 
> 
> Brian West brian at freeswitch.org <http://freeswitch.org/>
> Fri Jun 2 14:51:25 UTC 2017
> 
> I would guess mod_say_pt is not loaded.
> 
> On Fri, Jun 2, 2017 at 4:54 AM, Martin Boese <martin at maxnet.ao <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>> wrote:
> 
> > Hi!
> >
> > FreeSWITCH Version 1.6.15-32-bec4538~64bit (-32-bec4538 64bit)
> > Debian Jessie. Vanilla config.
> >
> > I am trying to enable portuguese to say numbers using dptools "Say".
> >
> > This is what I did:
> >  - Module mod_say_pt is loaded
> >  - in freeswitch.xml
> >    include lang/pt/pt_PT.xml
> >  - Downloaded sounds from
> >    https://github.com/jpawlowski/freeswitch-sounds-ttsand <https://github.com/jpawlowski/freeswitch-sounds-ttsand> placed them
> >    into /usr/share/freeswitch/sounds
> >    Directory structure is now like: /pt/tts/google/ascii/16000/...
> >
> > CLI> say_string t.wav pt NUMBER pronounced 123
> > [ERR] switch_xml.c:3180 Can't find phrases tag
> >
> > ..I found out that lang/pt_PT.xml seems to be missing the
> > <phrases><macros> tags within the <language> tag (vanilla config). I
> > fixed that.
> >
> > Now:
> > CLI> say_string t.wav pt NUMBER pronounced 123
> > [ERR] switch_ivr.c:3726 Invalid SAY Interface [pt]!
> >
> > BTW: English works fine:
> > CLI> say_string t.wav en NUMBER pronounced 123
> > file_string://digits/1.wav!digits/hundred.wav!digits/20.wav!digits/3.wav <>
> >
> > I also tried other languages but have same error "Invalid SAY
> > Interface".
> >
> > What am I missing. Please help.
> >
> > Thanks,
> > Martin
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> > http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org <http://www.freeswitch.org/>
> > http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> > http://www.cluecon.com <http://www.cluecon.com/>
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> > 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>
> > http://www.freeswitch.org <http://www.freeswitch.org/>
> 
> 
> 
> 
> -- 
> 
> *Brian West*
> brian at freeswitch.org <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> 
> *Twitter: @FreeSWITCH , @briankwest*
> 
> http://www.freeswitchbook.com <http://www.freeswitchbook.com/>
> http://www.freeswitchcookbook.com <http://www.freeswitchcookbook.com/>
> 
> Book a phone call (CST) <https://freeswitch.com/appointment <https://freeswitch.com/appointment>>
> 
> Allison prompts for FreeSWITCH:
> 
> *https://www.gofundme.com/allison-prompts-for-freeswitch* <https://www.gofundme.com/allison-prompts-for-freeswitch*>
> <https://www.gofundme.com/allison-prompts-for-freeswitch <https://www.gofundme.com/allison-prompts-for-freeswitch>>
> 
> Got Bugs? Report them here <https://freeswitch.org/jira <https://freeswitch.org/jira>>! | Reddit:
> /r/freeswitch <https://www.reddit.com/r/freeswitch <https://www.reddit.com/r/freeswitch>>
> 
> *T:*+19184209001 <tel:%28918%29%20420-9001> | *F:*+19184209002 <tel:%28918%29%20420-9002> | *M:*+1918424WEST (9378)
> *Skype:*briankwest
> -------------- next part --------------
> An HTML attachment was scrubbed...
> URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170602/d72352fe/attachment.html <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170602/d72352fe/attachment.html>>
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170607/ef1e25eb/attachment-0001.html>

From mase2hot at gmail.com  Wed Jun  7 15:49:39 2017
From: mase2hot at gmail.com (Jason Bedward)
Date: Wed, 7 Jun 2017 16:49:39 +0100
Subject: [Freeswitch-users] Re-invite Glare issue - FS ending call
Message-ID: <CANHADwNmo1FV6GO6xfhBhK8SfwGDZVbOXhzmBPr8JaGng71k7A@mail.gmail.com>

Hi,

I have an issue with re-invites on some calls. I'm using 1.6.15 and these
calls are using bypass media. I have Kamailio as inbound SBC but FS
connecting directly to my provider for B leg.


   - After 5 minutes plus on some calls my provider sends re-invite on A leg
   - FS then sends this re-invite to the B leg
   - At the same time the B leg sends a Re-invite
   - FS replys 491
   - B leg provider replys 100, then 500 (with retry in the 500)
   - FS send ACK and then BYE

Not sure what setting I need to change or can change infact to either retry
the invite in accordance with the 500 retry request. Or something else to
stop the call ending...

[image: Inline image 1]

Thanks
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170607/8485e141/attachment-0001.html>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: image.png
Type: image/png
Size: 45647 bytes
Desc: not available
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170607/8485e141/attachment-0001.png>

From mike at jerris.com  Wed Jun  7 15:53:00 2017
From: mike at jerris.com (Michael Jerris)
Date: Wed, 7 Jun 2017 11:53:00 -0400
Subject: [Freeswitch-users] Turn configuration mod_sofia
In-Reply-To: <26962017ab7e4ba88fbd2db5a551a34d@c4b.de>
References: <26962017ab7e4ba88fbd2db5a551a34d@c4b.de>
Message-ID: <F7B3EF2C-9B90-4248-8B4F-AAEF9DA078D1@jerris.com>

you would configure a TURN server on the client.  we will relay through a turn server when supplied by the client, but as we are a server stack, we have no support for a turn server on the server side, as Brain said, there is little reason for it on the server side.

> On Jun 7, 2017, at 6:49 AM, Alexander Haugg <Alexander.Haugg at c4b.de> wrote:
> 
> Hi,
>  
> my question:
> how can I configure an ICE server (turn) in the sip profile?
>  
> My think for this id case I need the possibility to configure the turn server with credentials in the sip profile like the stun configuration too.
> -          STUN configuration „<param name="ext-rtp-ip" value=" stun:stun.freeswitch.org"/>“ -> that‘s working!
> -          ICE configuration could be? „<param name="ext-rtp-ip" value=" { ‘stun:stun.freeswitch.org <http://stun.freeswitch.org/>‘, ‘turn:turn.myserver.com <http://turn.myserver.com/>;username=xxx;password=123‘}"/>“
>  
> Is there a possibility?
>  
> Thanks a lot
>  
> Von: Alexander Haugg 
> Gesendet: Donnerstag, 1. Juni 2017 15:24
> An: 'freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org>' <freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org>>
> Betreff: WG: Turn configuration mod_sofia 
>  
> Hi Brian,
>  
> we use the Freeswitch as „Man in the Middle“ for WebRTC.
>  
> The WebRTC Clients (ICE Link Stack from Frozen Mountain) registered via SIP on the Freeswitch.
> The SIP signalling is a special szenario and works successfully in several LAN WAN setups.
> But now we need srflx (that’s fine with teh STUN configuration) an relay candidates in the SDP that’s ganerated by the Freeswitch.
>  
> My think is, for this case i need the possibility to configure the turn server with credentials in the sip profile like the stun configuration too.
> -          STUN configuration „<param name="ext-rtp-ip" value=" stun:stun.freeswitch.org"/>“
> -          ICE configuration could be? „<param name="ext-rtp-ip" value=" { ‘stun:stun.freeswitch.org <http://stun.freeswitch.org/>‘, ‘turn:turn.myserver.com <http://turn.myserver.com/>;username=xxx;password=123‘}"/>“
>  
> Thanks a lot
>  
>  
> From brian at freeswitch.org <http://freeswitch.org/>  Tue May 30 13:50:32 2017
> From: brian at freeswitch.org <http://freeswitch.org/> (Brian West)
> Date: Tue, 30 May 2017 08:50:32 -0500
> Subject: [Freeswitch-users] Turn configuration mod_sofia
> In-Reply-To: <a9e067a18e104afbb53e207712467baf at c4b.de <mailto:a9e067a18e104afbb53e207712467baf at c4b.de>>
> References: <a9e067a18e104afbb53e207712467baf at c4b.de <mailto:a9e067a18e104afbb53e207712467baf at c4b.de>>
> Message-ID: <CAEJMVkDy5e2COC8cH=xV=ozM0gepBS3cQ_YxUPQk4twMoaSVmg at mail.gmail.com <mailto:CAEJMVkDy5e2COC8cH=xV=ozM0gepBS3cQ_YxUPQk4twMoaSVmg at mail.gmail.com>>
>  
> There is little reason to use TURN when speaking to FreeSWITCH, What issue
> are you trying to solve?
>  
>  
> Von: Alexander Haugg 
> Gesendet: Montag, 29. Mai 2017 18:25
> An: 'freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org>' <freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org>>
> Betreff: Turn configuration mod_sofia 
>  
> Hi all,
>  
> the stun configuration for the mod_sofia profile is very easy „<param name="ext-rtp-ip" value=" stun:stun.freeswitch.org"/>“
> but what is to do, if i need a relay candidate in the sdp? How can i set the turn address and the login credentials?
>  
> Thanks a lot
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> http://www.cluecon.com <http://www.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>
> http://www.freeswitch.org <http://www.freeswitch.org/>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170607/4c066b7f/attachment-0001.html>

From mike at jerris.com  Wed Jun  7 15:57:18 2017
From: mike at jerris.com (Michael Jerris)
Date: Wed, 7 Jun 2017 11:57:18 -0400
Subject: [Freeswitch-users] Re-invite Glare issue - FS ending call
In-Reply-To: <CANHADwNmo1FV6GO6xfhBhK8SfwGDZVbOXhzmBPr8JaGng71k7A@mail.gmail.com>
References: <CANHADwNmo1FV6GO6xfhBhK8SfwGDZVbOXhzmBPr8JaGng71k7A@mail.gmail.com>
Message-ID: <72A53E3A-47FC-498E-AF90-2B39448DE52A@jerris.com>

why are they both sending a re-invite at the same time?  This is called glare, we seem to be handling it properly, the provider seems not to be.  The easiest fix here is probably to figure out why we are both sending re-invite at the same time, maybe session timers and the provider is broken and trying to send it when we said we would.
> On Jun 7, 2017, at 11:49 AM, Jason Bedward <mase2hot at gmail.com> wrote:
> 
> Hi,
> 
> I have an issue with re-invites on some calls. I'm using 1.6.15 and these calls are using bypass media. I have Kamailio as inbound SBC but FS connecting directly to my provider for B leg.
> 
> After 5 minutes plus on some calls my provider sends re-invite on A leg
> FS then sends this re-invite to the B leg
> At the same time the B leg sends a Re-invite
> FS replys 491
> B leg provider replys 100, then 500 (with retry in the 500)
> FS send ACK and then BYE
> Not sure what setting I need to change or can change infact to either retry the invite in accordance with the 500 retry request. Or something else to stop the call ending...
> 
> <image.png>
> 
> Thanks
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170607/44529a39/attachment.html>

From grcamauer at gmail.com  Wed Jun  7 20:17:33 2017
From: grcamauer at gmail.com (Guillermo Ruiz Camauer)
Date: Wed, 7 Jun 2017 17:17:33 -0300
Subject: [Freeswitch-users] Open Source SIP\WebRTC clients compatible
	with FreeSWITCH
In-Reply-To: <10cd01d236a8$425136b0$c6f3a410$@freeswitch.org>
References: <6FD2F8B5BB72834E9939AEDF9FB802A901E860B047@mbx-01.sysconfig.co.uk>
 <10c301d236a7$1ea0f8f0$5be2ead0$@freeswitch.org>
 <CALR0W69hJEpLqwgnq2FE0hBi6E3BHOOWEE=TANcjFQ8SEMRBoA@mail.gmail.com>
 <10cd01d236a8$425136b0$c6f3a410$@freeswitch.org>
Message-ID: <CAPNTnnr9FjeR4HGPzkUaU_mG6b326gZ_-RJk4Z=PX1C5XSco-g@mail.gmail.com>

Apple announces WebRTC support in iOS11 / Safari:
https://apple.slashdot.org/story/17/06/07/1958242/apple-announces-support-for-webrtc-in-safari-11



On Fri, Nov 4, 2016 at 11:32 AM, Ken Rice <krice at freeswitch.org> wrote:

> No one supports Native WebRTC on iOS at this time except for people using
> their own private SDKs that they are not allowing to get out there…
>
>
>
> Apple does not have webRTC in webkit (Safari) or iOS at this time. Chrome
> on iOS is not even really Chrome, its just a wrapper around the WebKIT APIs
> and is effectively just safari with a few extra functions and built to look
> like chrome.
>
>
>
>
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Chandramouli
> P
> *Sent:* Friday, November 4, 2016 9:29 AM
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Subject:* Re: [Freeswitch-users] Open Source SIP\WebRTC clients
> compatible with FreeSWITCH
>
>
>
> Hello Ken,
>
>
>
> We worked with Google native WebRTC on Firefox, Chrome, and Opera browsers
> on Windows OS. I recently noticed that Google has added the WebRTC support
> for Android, and iOS platforms (webrtc.org). Now, we are planning to
> develop video calling module using Google native WebRTC on these new
> platforms. Can anybody give me the information about my below queries:
>
>
>
> 1) Does Google native WebRTC supports Apple iOS platform (native mobile
> app)?
>
> 2) Does Google native WebRTC supports Apple OS X platform?
>
> 3) Is it possible to develop video calling module using native WebRTC on
> Safari, and Chrome browsers on Apple OS X platform?
>
> 4) Does Google native WebRTC supports Android platform (native mobile app)?
>
> 5) If it supports, I could not find any documentation for Apple iOS, Apple
> OS X, and Android platforms specifically. Could you please send some
> referral links?
>
> 6) I could not able to find the referral examples also for Apple iOS,
> Apple OS X, and Android platforms specifically. Could you please send some
> referral links?
>
>
>
> Please do needful.
>
>
>
> Thank you,
>
> Chandramouli.
>
>
>
>
>
> On Fri, Nov 4, 2016 at 7:54 PM, Ken Rice <krice at freeswitch.org> wrote:
>
> Verto Works on pretty much any platform that has native webrtc support
> now... unfortunately things like iOS and don’t have native iOS support yet…
>
>
>
> If you are looking to build something you might contact
> consulting at freeswitch.org and see if you can work with the FSS Team to
> develop something
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Shaun Stokes
> *Sent:* Friday, November 4, 2016 8:40 AM
> *To:* 'FreeSWITCH Users Help' <freeswitch-users at lists.freeswitch.org>
> *Subject:* [Freeswitch-users] Open Source SIP\WebRTC clients compatible
> with FreeSWITCH
>
>
>
> Hi All,
>
>
>
> Does anyone have any recommendations on a good open source SIP\WebRTC
> client which works on multiple platforms (Windows, Mac, Linux, Mobiles) to
> provide presence, voice, video, instant messaging, screen sharing and file
> sharing? This must be capable of integrating with FreeSWITCH for voice and
> video (presence via FreeSWITCH would be an advantage).
>
>
>
> Many Thanks,
>
> Shaun
>
> Shaun Stokes - Infrastructure Analyst
>
> T :
>
> 01453 700713
>
> E :
>
> shaun.stokes at itec-support.co.uk
>
> W :
>
> www.itec-support.co.uk
>
> Registered Address :- ITEC Support, Suite 2 Prospect House, Bath Road,
> Stroud, Gloucestershire GL5 3QF
> Company No. 06908001
>
>
> CONFIDENTIALITY NOTICE
> This communication and the information it contains are intended for the
> person or organisation to which it is addressed. Its contents are
> confidential and may be protected in law. Unauthorised use, copying or
> disclosure of any of it may be unlawful. If you are not the intended
> recipient, please contact us immediately.
> The contents of any attachments in this e-mail may contain software
> viruses, which could damage your own computer system. While ITEC Support
> has taken every reasonable precaution to minimise this risk, we cannot
> accept liability for any damage which you sustain as a result of software
> viruses. You should carry out your own virus checking procedure before
> opening any attachment.
>
>
> ______________________________________________________________________
> This message has been checked for all known viruses by MessageLabs Virus
> Scanning Service.
> ______________________________________________________________________
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org
>



-- 
Guillermo Ruiz Camauer
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170607/71b30872/attachment-0001.html>

From savolainen at erinaco.ru  Wed Jun  7 18:16:45 2017
From: savolainen at erinaco.ru (Dmitri Savolainen)
Date: Wed, 7 Jun 2017 21:16:45 +0300
Subject: [Freeswitch-users] [SR-Users] Multiple FreeSWITCH servers
	behind kamailio-websocket
In-Reply-To: <CAFArqsZjjVP63_OscW=qVOh_skaddndKuXzKaWJp2WgHg9KaEw@mail.gmail.com>
References: <CAFArqsZjjVP63_OscW=qVOh_skaddndKuXzKaWJp2WgHg9KaEw@mail.gmail.com>
Message-ID: <CAGYSiDgvPp-BMtvK5C=xpjB=hZkG-KA-Jt4zhSr9=T93EXHewA@mail.gmail.com>

webrtc kamailio  for example here https://github.com/havfo/WEBRTC-to-SIP

By the way rtpengine is not mandatory with FreeSwitch. It is possible to
use a set of FS(1.6) and balancing by dispatcher module

2017-06-07 14:47 GMT+03:00 Karsten Horsmann <khorsmann at gmail.com>:

> Hello List,
>
>
> is there any howto about webrtc loadbalance in combination with kamailio
> and FreeSWITCH?
>
> I want to share one WSS address/endpoint to multiple FreeSWITCH backends.
> Or is there any other best practice?
>
> My callflow is mostly that my internal SIP Servers called my registered
> webrtc clients.
>
> Would be nice to get some input.
>
> --
> Kind Regards
> *Karsten Horsmann*
>
> _______________________________________________
> Kamailio (SER) - Users Mailing List
> sr-users at lists.kamailio.org
> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>
>


-- 
Savolainen Dmitri
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170607/762a57ce/attachment.html>

From anthony.minessale at gmail.com  Thu Jun  8 00:44:32 2017
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 08 Jun 2017 00:44:32 +0000
Subject: [Freeswitch-users] question about HA solution
In-Reply-To: <E55363B0-487F-4F65-8F3F-78620AB1AB73@jerris.com>
References: <CADcrYG+RHGQ2fC7dSKK9x9WfVvYcTczOtbppCLgttRCfMGqArA@mail.gmail.com>
 <CAFLo2QxxbZ5Ged0J0p9_VfQ2nkGtuV0CDp9G763EjjuvqUv8iw@mail.gmail.com>
 <CADcrYG+sO0+WYUtYVN6pkpU-5SwyfurfshdUdhxb-c4Az1s49Q@mail.gmail.com>
 <8A13E0EC-FA13-4267-80F6-CE1A8E8360CF@jerris.com>
 <CADcrYGLbN1Ut6riF1Lf7UvwG9ypvR2ekWxt1m=e5BmqOzP_c=w@mail.gmail.com>
 <E55363B0-487F-4F65-8F3F-78620AB1AB73@jerris.com>
Message-ID: <CAKbxfG-EaPcFF_JXAiSP-f0sbti2AYxZQdwiHhbJ12uUnpUr2A@mail.gmail.com>

Which tables in particular?  The one related to call recovery is the only
one that matters and it never auto-clears


On Wed, Jun 7, 2017 at 10:36 AM Michael Jerris <mike at jerris.com> wrote:

> That param should keep it from doing so, if its not you are not setting it
> somehow or something else is wiping the db.
>
> On Jun 5, 2017, at 1:50 PM, Denys Pozniak <denys.pozniak at crazycall.com>
> wrote:
>
> Yes, correct. But when you restart FS on slave, it will erase database.
> And option auto-clear-sql=false not working for me.
>
> On Jun 5, 2017 6:32 PM, "Michael Jerris" <mike at jerris.com> wrote:
>
> recovered calls will get new entries in the table.
>
> On Jun 5, 2017, at 7:41 AM, Denys Pozniak <denys.pozniak at crazycall.com>
> wrote:
>
> Hello!
>
> Thank you *Raymond* about your explanation, but I dont agree with some
> point:
> *If it really need an answer about your question -- "if it is possible to
> move calls back".   I think it's unnecessary.*  - in my case I have two
> not equal servers, so I need to have only one as a master.
> If switchover happens I need to have ability to restore master back.
>
> Thank you *Luis* for your link, you can do simple test to understand what
> I am talking about: do call -> check on master and slave #show channels ->
> restart FS on slave -> check on master #show channels. In my case I dont
> see any active calls after this, so restoring back is not possible.
>
>
>
> On 3 June 2017 at 22:16, Luis Daniel Lucio Quiroz <
> luis.daniel.lucio at gmail.com> wrote:
>
>> You may want to read this article.
>>
>> http://inside-out.xyz/technology/how-to-configure-freeswitch-for-ha.html
>>
>> Le 31 mai 2017 6:29 PM, "Denys Pozniak" <denys.pozniak at crazycall.com> a
>> écrit :
>>
>> Hello!
>>
>> I built FS HA solution based on keepalived and mysql master-master.
>> It works ok generally, but as I understand FS after restarting cleaning
>> own database.
>>
>> So when node1 fails calls jump to node2, after script restarts node1 it
>> is not possible to move calls back.
>>
>> Tried options in switch.conf.xml, but no luck:
>>
>>     <param name="auto-create-schemas" value="false"/>
>>     <param name="auto-clear-sql" value="false"/>
>>
>> Is there is a way to solve this?
>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org

-- 
Anthony Minessale II       ♬ @anthmfs  ♬ @FreeSWITCH  ♬

☞ http://freeswitch.org/  ☞ http://cluecon.com/  ☞
http://twitter.com/FreeSWITCH
☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+
<http://freeswitch.org/g+>*

ClueCon Weekly Development Call
☎ sip:888 at conference.freeswitch.org  ☎ +19193869900

https://www.youtube.com/watch?v=9XXgW34t40s
https://www.youtube.com/watch?v=NLaDpGQuZDA
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170608/67c573ee/attachment.html>

From rmundkowsky at ets.org  Wed Jun  7 21:50:31 2017
From: rmundkowsky at ets.org (Mundkowsky, Robert)
Date: Wed, 7 Jun 2017 21:50:31 +0000
Subject: [Freeswitch-users] Open Source SIP\WebRTC clients
 compatible	with FreeSWITCH
In-Reply-To: <CAPNTnnr9FjeR4HGPzkUaU_mG6b326gZ_-RJk4Z=PX1C5XSco-g@mail.gmail.com>
References: <6FD2F8B5BB72834E9939AEDF9FB802A901E860B047@mbx-01.sysconfig.co.uk>
 <10c301d236a7$1ea0f8f0$5be2ead0$@freeswitch.org>
 <CALR0W69hJEpLqwgnq2FE0hBi6E3BHOOWEE=TANcjFQ8SEMRBoA@mail.gmail.com>
 <10cd01d236a8$425136b0$c6f3a410$@freeswitch.org>
 <CAPNTnnr9FjeR4HGPzkUaU_mG6b326gZ_-RJk4Z=PX1C5XSco-g@mail.gmail.com>
Message-ID: <BN6PR07MB355689F93B56219553B97C31BEC80@BN6PR07MB3556.namprd07.prod.outlook.com>

Cool, not if they just support getUserMedia!

Robert Mundkowsky

From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Guillermo Ruiz Camauer
Sent: Wednesday, June 7, 2017 4:18 PM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Open Source SIP\WebRTC clients compatible with FreeSWITCH

Apple announces WebRTC support in iOS11 / Safari:  https://apple.slashdot.org/story/17/06/07/1958242/apple-announces-support-for-webrtc-in-safari-11



On Fri, Nov 4, 2016 at 11:32 AM, Ken Rice <krice at freeswitch.org<mailto:krice at freeswitch.org>> wrote:
No one supports Native WebRTC on iOS at this time except for people using their own private SDKs that they are not allowing to get out there…

Apple does not have webRTC in webkit (Safari) or iOS at this time. Chrome on iOS is not even really Chrome, its just a wrapper around the WebKIT APIs and is effectively just safari with a few extra functions and built to look like chrome.



From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Chandramouli P
Sent: Friday, November 4, 2016 9:29 AM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
Subject: Re: [Freeswitch-users] Open Source SIP\WebRTC clients compatible with FreeSWITCH

Hello Ken,

We worked with Google native WebRTC on Firefox, Chrome, and Opera browsers on Windows OS. I recently noticed that Google has added the WebRTC support for Android, and iOS platforms (webrtc.org<http://webrtc.org>). Now, we are planning to develop video calling module using Google native WebRTC on these new platforms. Can anybody give me the information about my below queries:

1) Does Google native WebRTC supports Apple iOS platform (native mobile app)?
2) Does Google native WebRTC supports Apple OS X platform?
3) Is it possible to develop video calling module using native WebRTC on Safari, and Chrome browsers on Apple OS X platform?
4) Does Google native WebRTC supports Android platform (native mobile app)?
5) If it supports, I could not find any documentation for Apple iOS, Apple OS X, and Android platforms specifically. Could you please send some referral links?
6) I could not able to find the referral examples also for Apple iOS, Apple OS X, and Android platforms specifically. Could you please send some referral links?

Please do needful.

Thank you,
Chandramouli.


On Fri, Nov 4, 2016 at 7:54 PM, Ken Rice <krice at freeswitch.org<mailto:krice at freeswitch.org>> wrote:
Verto Works on pretty much any platform that has native webrtc support now... unfortunately things like iOS and don’t have native iOS support yet…

If you are looking to build something you might contact consulting at freeswitch.org<mailto:consulting at freeswitch.org> and see if you can work with the FSS Team to develop something

From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Shaun Stokes
Sent: Friday, November 4, 2016 8:40 AM
To: 'FreeSWITCH Users Help' <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
Subject: [Freeswitch-users] Open Source SIP\WebRTC clients compatible with FreeSWITCH

Hi All,

Does anyone have any recommendations on a good open source SIP\WebRTC client which works on multiple platforms (Windows, Mac, Linux, Mobiles) to provide presence, voice, video, instant messaging, screen sharing and file sharing? This must be capable of integrating with FreeSWITCH for voice and video (presence via FreeSWITCH would be an advantage).

Many Thanks,
Shaun
[http://www.itec-support.co.uk/wp-content/uploads/2016/07/email_logo.jpg]

Shaun Stokes - Infrastructure Analyst

T :

01453 700713

E :

shaun.stokes at itec-support.co.uk<mailto:shaun.stokes at itec-support.co.uk>

W :

www.itec-support.co.uk<http://www.itec-support.co.uk>


Registered Address :- ITEC Support, Suite 2 Prospect House, Bath Road, Stroud, Gloucestershire GL5 3QF
Company No. 06908001

CONFIDENTIALITY NOTICE
This communication and the information it contains are intended for the person or organisation to which it is addressed. Its contents are confidential and may be protected in law. Unauthorised use, copying or disclosure of any of it may be unlawful. If you are not the intended recipient, please contact us immediately.
The contents of any attachments in this e-mail may contain software viruses, which could damage your own computer system. While ITEC Support has taken every reasonable precaution to minimise this risk, we cannot accept liability for any damage which you sustain as a result of software viruses. You should carry out your own virus checking procedure before opening any attachment.

______________________________________________________________________
This message has been checked for all known viruses by MessageLabs Virus Scanning Service.
______________________________________________________________________

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.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
http://www.freeswitch.org


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.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
http://www.freeswitch.org



--
Guillermo Ruiz Camauer

________________________________

This e-mail and any files transmitted with it may contain privileged or confidential information. It is solely for use by the individual for whom it is intended, even if addressed incorrectly. If you received this e-mail in error, please notify the sender; do not disclose, copy, distribute, or take any action in reliance on the contents of this information; and delete it from your system. Any other use of this e-mail is prohibited.


Thank you for your compliance.

________________________________
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170607/28d21861/attachment-0001.html>

From Alexander.Haugg at c4b.de  Thu Jun  8 06:34:23 2017
From: Alexander.Haugg at c4b.de (Alexander Haugg)
Date: Thu, 8 Jun 2017 06:34:23 +0000
Subject: [Freeswitch-users] Turn configuration mod_sofia
In-Reply-To: <F7B3EF2C-9B90-4248-8B4F-AAEF9DA078D1@jerris.com>
References: <26962017ab7e4ba88fbd2db5a551a34d@c4b.de>
 <F7B3EF2C-9B90-4248-8B4F-AAEF9DA078D1@jerris.com>
Message-ID: <0b9ea6953aef4324b9c478933e5235aa@c4b.de>

Hi,

now i know what you mean.
Thanks for the right direction!

„Mastering Freeswitch“ is a pretty nice book!

Von: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Michael Jerris
Gesendet: Mittwoch, 7. Juni 2017 17:53
An: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Betreff: Re: [Freeswitch-users] Turn configuration mod_sofia

you would configure a TURN server on the client.  we will relay through a turn server when supplied by the client, but as we are a server stack, we have no support for a turn server on the server side, as Brain said, there is little reason for it on the server side.

On Jun 7, 2017, at 6:49 AM, Alexander Haugg <Alexander.Haugg at c4b.de<mailto:Alexander.Haugg at c4b.de>> wrote:

Hi,

my question:
how can I configure an ICE server (turn) in the sip profile?

My think for this id case I need the possibility to configure the turn server with credentials in the sip profile like the stun configuration too.
-          STUN configuration „<param name="ext-rtp-ip" value=" stun:stun.freeswitch.org<http://stun.freeswitch.org>"/>“ -> that‘s working!
-          ICE configuration could be? „<param name="ext-rtp-ip" value=" { ‘stun:stun.freeswitch.org<http://stun.freeswitch.org/>‘, ‘turn:turn.myserver.com<http://turn.myserver.com/>;username=xxx;password=123‘}"/>“

Is there a possibility?

Thanks a lot

Von: Alexander Haugg
Gesendet: Donnerstag, 1. Juni 2017 15:24
An: 'freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>' <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
Betreff: WG: Turn configuration mod_sofia

Hi Brian,

we use the Freeswitch as „Man in the Middle“ for WebRTC.

The WebRTC Clients (ICE Link Stack from Frozen Mountain) registered via SIP on the Freeswitch.
The SIP signalling is a special szenario and works successfully in several LAN WAN setups.
But now we need srflx (that’s fine with teh STUN configuration) an relay candidates in the SDP that’s ganerated by the Freeswitch.

My think is, for this case i need the possibility to configure the turn server with credentials in the sip profile like the stun configuration too.
-          STUN configuration „<param name="ext-rtp-ip" value=" stun:stun.freeswitch.org<http://stun.freeswitch.org>"/>“
-          ICE configuration could be? „<param name="ext-rtp-ip" value=" { ‘stun:stun.freeswitch.org<http://stun.freeswitch.org/>‘, ‘turn:turn.myserver.com<http://turn.myserver.com/>;username=xxx;password=123‘}"/>“

Thanks a lot


From brian at freeswitch.org<http://freeswitch.org/>  Tue May 30 13:50:32 2017
From: brian at freeswitch.org<http://freeswitch.org/> (Brian West)
Date: Tue, 30 May 2017 08:50:32 -0500
Subject: [Freeswitch-users] Turn configuration mod_sofia
In-Reply-To: <a9e067a18e104afbb53e207712467baf at c4b.de<mailto:a9e067a18e104afbb53e207712467baf at c4b.de>>
References: <a9e067a18e104afbb53e207712467baf at c4b.de<mailto:a9e067a18e104afbb53e207712467baf at c4b.de>>
Message-ID: <CAEJMVkDy5e2COC8cH=xV=ozM0gepBS3cQ_YxUPQk4twMoaSVmg at mail.gmail.com<mailto:CAEJMVkDy5e2COC8cH=xV=ozM0gepBS3cQ_YxUPQk4twMoaSVmg at mail.gmail.com>>

There is little reason to use TURN when speaking to FreeSWITCH, What issue
are you trying to solve?


Von: Alexander Haugg
Gesendet: Montag, 29. Mai 2017 18:25
An: 'freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>' <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
Betreff: Turn configuration mod_sofia

Hi all,

the stun configuration for the mod_sofia profile is very easy „<param name="ext-rtp-ip" value=" stun:stun.freeswitch.org<http://stun.freeswitch.org>"/>“
but what is to do, if i need a relay candidate in the sdp? How can i set the turn address and the login credentials?

Thanks a lot
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com<http://www.freeswitchsolutions.com/>

Official FreeSWITCH Sites
http://www.freeswitch.org<http://www.freeswitch.org/>
http://confluence.freeswitch.org<http://confluence.freeswitch.org/>
http://www.cluecon.com<http://www.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
http://www.freeswitch.org<http://www.freeswitch.org/>

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170608/ab018c27/attachment-0001.html>

From mase2hot at gmail.com  Thu Jun  8 07:14:00 2017
From: mase2hot at gmail.com (Jason Bedward)
Date: Thu, 8 Jun 2017 08:14:00 +0100
Subject: [Freeswitch-users] Re-invite Glare issue - FS ending call
In-Reply-To: <72A53E3A-47FC-498E-AF90-2B39448DE52A@jerris.com>
References: <CANHADwNmo1FV6GO6xfhBhK8SfwGDZVbOXhzmBPr8JaGng71k7A@mail.gmail.com>
 <72A53E3A-47FC-498E-AF90-2B39448DE52A@jerris.com>
Message-ID: <CANHADwPZtVGd0c5n7TvWcdJMJt9SZKxPQde7fHFBk-Xon4o3UQ@mail.gmail.com>

My provider is sending the reinvite on the A leg FS the sends this on to
the B leg. At same time the B leg provider sends a reinvite.

I dont believe FS is sending the reinvite on its own accord. Its only
because of the A leg and that we are using bypass media. So I dont think
session timers will make a difference.

FYI A leg and B leg provider are the same company although different
servers.

On 7 Jun 2017 16:57, "Michael Jerris" <mike at jerris.com> wrote:

> why are they both sending a re-invite at the same time?  This is called
> glare, we seem to be handling it properly, the provider seems not to be.
> The easiest fix here is probably to figure out why we are both sending
> re-invite at the same time, maybe session timers and the provider is broken
> and trying to send it when we said we would.
>
> On Jun 7, 2017, at 11:49 AM, Jason Bedward <mase2hot at gmail.com> wrote:
>
> Hi,
>
> I have an issue with re-invites on some calls. I'm using 1.6.15 and these
> calls are using bypass media. I have Kamailio as inbound SBC but FS
> connecting directly to my provider for B leg.
>
>
>    - After 5 minutes plus on some calls my provider sends re-invite on A
>    leg
>    - FS then sends this re-invite to the B leg
>    - At the same time the B leg sends a Re-invite
>    - FS replys 491
>    - B leg provider replys 100, then 500 (with retry in the 500)
>    - FS send ACK and then BYE
>
> Not sure what setting I need to change or can change infact to either
> retry the invite in accordance with the 500 retry request. Or something
> else to stop the call ending...
>
> <image.png>
>
> Thanks
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170608/caffab6f/attachment.html>

From s.safarov at gmail.com  Thu Jun  8 07:15:38 2017
From: s.safarov at gmail.com (Sergey Safarov)
Date: Thu, 08 Jun 2017 07:15:38 +0000
Subject: [Freeswitch-users] BugHunt
Message-ID: <CAHtxdDddZkiByD-VEJaKD03Pn2WkZzkvqpqN2xhx2g79y9dGTQ@mail.gmail.com>

Is BugHunt will be today?
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170608/9f6c0600/attachment.html>

From denys.pozniak at crazycall.com  Thu Jun  8 08:07:18 2017
From: denys.pozniak at crazycall.com (Denys Pozniak)
Date: Thu, 8 Jun 2017 10:07:18 +0200
Subject: [Freeswitch-users] question about HA solution
In-Reply-To: <E55363B0-487F-4F65-8F3F-78620AB1AB73@jerris.com>
References: <CADcrYG+RHGQ2fC7dSKK9x9WfVvYcTczOtbppCLgttRCfMGqArA@mail.gmail.com>
 <CAFLo2QxxbZ5Ged0J0p9_VfQ2nkGtuV0CDp9G763EjjuvqUv8iw@mail.gmail.com>
 <CADcrYG+sO0+WYUtYVN6pkpU-5SwyfurfshdUdhxb-c4Az1s49Q@mail.gmail.com>
 <8A13E0EC-FA13-4267-80F6-CE1A8E8360CF@jerris.com>
 <CADcrYGLbN1Ut6riF1Lf7UvwG9ypvR2ekWxt1m=e5BmqOzP_c=w@mail.gmail.com>
 <E55363B0-487F-4F65-8F3F-78620AB1AB73@jerris.com>
Message-ID: <CADcrYGKK=cTmVzG5oY0MV3b+4CLqg0=YzGvCiDZNZdBqQe3mMA@mail.gmail.com>

Hello!

My configs:

*switch.conf.xml*

    <param name="switchname" value="freeswitch01"/>
    <param name="core-recovery-db-dsn"
value="odbc://freeswitch:root:ubuntu"/>
    <param name="core-db-dsn" value="odbc://freeswitch:root:ubuntu"/>
    <param name="auto-create-schemas" value="false"/>
    <param name="auto-clear-sql" value="false"/>


*external.conf.xml*

   <param name="track-calls" value="true"/>
   <param name="odbc-dsn" value="odbc://freeswitch:root:ubuntu"/>


On 7 June 2017 at 17:35, Michael Jerris <mike at jerris.com> wrote:

> That param should keep it from doing so, if its not you are not setting it
> somehow or something else is wiping the db.
>
> On Jun 5, 2017, at 1:50 PM, Denys Pozniak <denys.pozniak at crazycall.com>
> wrote:
>
> Yes, correct. But when you restart FS on slave, it will erase database.
> And option auto-clear-sql=false not working for me.
>
> On Jun 5, 2017 6:32 PM, "Michael Jerris" <mike at jerris.com> wrote:
>
> recovered calls will get new entries in the table.
>
> On Jun 5, 2017, at 7:41 AM, Denys Pozniak <denys.pozniak at crazycall.com>
> wrote:
>
> Hello!
>
> Thank you *Raymond* about your explanation, but I dont agree with some
> point:
> *If it really need an answer about your question -- "if it is possible to
> move calls back".   I think it's unnecessary.*  - in my case I have two
> not equal servers, so I need to have only one as a master.
> If switchover happens I need to have ability to restore master back.
>
> Thank you *Luis* for your link, you can do simple test to understand what
> I am talking about: do call -> check on master and slave #show channels ->
> restart FS on slave -> check on master #show channels. In my case I dont
> see any active calls after this, so restoring back is not possible.
>
>
>
> On 3 June 2017 at 22:16, Luis Daniel Lucio Quiroz <
> luis.daniel.lucio at gmail.com> wrote:
>
>> You may want to read this article.
>>
>> http://inside-out.xyz/technology/how-to-configure-freeswitch-for-ha.html
>>
>> Le 31 mai 2017 6:29 PM, "Denys Pozniak" <denys.pozniak at crazycall.com> a
>> écrit :
>>
>> Hello!
>>
>> I built FS HA solution based on keepalived and mysql master-master.
>> It works ok generally, but as I understand FS after restarting cleaning
>> own database.
>>
>> So when node1 fails calls jump to node2, after script restarts node1 it
>> is not possible to move calls back.
>>
>> Tried options in switch.conf.xml, but no luck:
>>
>>     <param name="auto-create-schemas" value="false"/>
>>     <param name="auto-clear-sql" value="false"/>
>>
>> Is there is a way to solve this?
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170608/7c1a4097/attachment.html>

From v.zakhozhai at gmail.com  Thu Jun  8 09:07:26 2017
From: v.zakhozhai at gmail.com (Vladyslav Zakhozhai)
Date: Thu, 8 Jun 2017 12:07:26 +0300
Subject: [Freeswitch-users] XML curl "not found" response
Message-ID: <CANsY4DJ3mg0K5vzf1KyOEQXMA6w4Emu2vzWkT709APVJ2j7PEg@mail.gmail.com>

Hi,

I am curious what HTTP status xml curl server must provide when no result
is found for request with body

<?xml version="1.0" encoding="UTF-8" standalone="no"?>
<document type="freeswitch/xml">
  <section name="result">
    <result status="not found" />
  </section>
</document>


I mean HTTP 200 or HTTP 404.


My problem is the following:

1. When freeswitch gets 404 for directory request it reports error:

[ERR] mod_xml_curl.c:315 Received HTTP error 404 trying to fetch


2. When freeswitch gets 200 with status "not found" on dialplan request
failover to file xml config does not occure.

Thank you in advance.

-- 
С уважением,
Владислав Захожай
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170608/1f1d2f51/attachment-0001.html>

From gmaruzz at gmail.com  Thu Jun  8 09:18:43 2017
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Thu, 8 Jun 2017 11:18:43 +0200
Subject: [Freeswitch-users] XML curl "not found" response
In-Reply-To: <CANsY4DJ3mg0K5vzf1KyOEQXMA6w4Emu2vzWkT709APVJ2j7PEg@mail.gmail.com>
References: <CANsY4DJ3mg0K5vzf1KyOEQXMA6w4Emu2vzWkT709APVJ2j7PEg@mail.gmail.com>
Message-ID: <CALXCt0pjCHuwOAY-Z-VahG1JPKVDdQo1oo6HR4WPC=pCMAvxoQ@mail.gmail.com>

"""If it receives a valid response from your web application, then it will
load the configuration just like it would if you had put it into the FreeSWITCH
Configuration File <https://wiki.freeswitch.org/wiki/Freeswitch.xml>. If it
receives an invalid or 404 *not found* response, then it will attempt to
look for the file on disk instead."""

"""https://freeswitch.org/confluence/display/FREESWITCH/
mod_xml_curl#mod_xml_curl-Section:notfound"""

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


On 8 June 2017 at 11:07, Vladyslav Zakhozhai <v.zakhozhai at gmail.com> wrote:

> Hi,
>
> I am curious what HTTP status xml curl server must provide when no result
> is found for request with body
>
> <?xml version="1.0" encoding="UTF-8" standalone="no"?>
> <document type="freeswitch/xml">
>   <section name="result">
>     <result status="not found" />
>   </section>
> </document>
>
>
> I mean HTTP 200 or HTTP 404.
>
>
> My problem is the following:
>
> 1. When freeswitch gets 404 for directory request it reports error:
>
> [ERR] mod_xml_curl.c:315 Received HTTP error 404 trying to fetch
>
>
> 2. When freeswitch gets 200 with status "not found" on dialplan request
> failover to file xml config does not occure.
>
> Thank you in advance.
>
> --
> С уважением,
> Владислав Захожай
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org
>



-- 

Sincerely,

Giovanni Maruzzelli
OpenTelecom.IT
cell: +39 347 266 56 18
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170608/a37cd18c/attachment.html>

From mase2hot at gmail.com  Thu Jun  8 09:40:48 2017
From: mase2hot at gmail.com (Jason Bedward)
Date: Thu, 8 Jun 2017 10:40:48 +0100
Subject: [Freeswitch-users] Re-invite Glare issue - FS ending call
In-Reply-To: <CANHADwPZtVGd0c5n7TvWcdJMJt9SZKxPQde7fHFBk-Xon4o3UQ@mail.gmail.com>
References: <CANHADwNmo1FV6GO6xfhBhK8SfwGDZVbOXhzmBPr8JaGng71k7A@mail.gmail.com>
 <72A53E3A-47FC-498E-AF90-2B39448DE52A@jerris.com>
 <CANHADwPZtVGd0c5n7TvWcdJMJt9SZKxPQde7fHFBk-Xon4o3UQ@mail.gmail.com>
Message-ID: <CANHADwNyoAX+nSwRADCXeLMxJx_+T=-=4bJxNbOUVyTPoo7WTQ@mail.gmail.com>

Looking again as this according to RFC 3261 Freeswitch is doing this wrong?

For the B leg my FS is the UAC, and if so when the invite comes in from the
B leg it shouldn't respond 491 as that should only be done from the UAS.

Unless I'm wrong with this?

On Thu, Jun 8, 2017 at 8:14 AM, Jason Bedward <mase2hot at gmail.com> wrote:

> My provider is sending the reinvite on the A leg FS the sends this on to
> the B leg. At same time the B leg provider sends a reinvite.
>
> I dont believe FS is sending the reinvite on its own accord. Its only
> because of the A leg and that we are using bypass media. So I dont think
> session timers will make a difference.
>
> FYI A leg and B leg provider are the same company although different
> servers.
>
> On 7 Jun 2017 16:57, "Michael Jerris" <mike at jerris.com> wrote:
>
>> why are they both sending a re-invite at the same time?  This is called
>> glare, we seem to be handling it properly, the provider seems not to be.
>> The easiest fix here is probably to figure out why we are both sending
>> re-invite at the same time, maybe session timers and the provider is broken
>> and trying to send it when we said we would.
>>
>> On Jun 7, 2017, at 11:49 AM, Jason Bedward <mase2hot at gmail.com> wrote:
>>
>> Hi,
>>
>> I have an issue with re-invites on some calls. I'm using 1.6.15 and these
>> calls are using bypass media. I have Kamailio as inbound SBC but FS
>> connecting directly to my provider for B leg.
>>
>>
>>    - After 5 minutes plus on some calls my provider sends re-invite on A
>>    leg
>>    - FS then sends this re-invite to the B leg
>>    - At the same time the B leg sends a Re-invite
>>    - FS replys 491
>>    - B leg provider replys 100, then 500 (with retry in the 500)
>>    - FS send ACK and then BYE
>>
>> Not sure what setting I need to change or can change infact to either
>> retry the invite in accordance with the 500 retry request. Or something
>> else to stop the call ending...
>>
>> <image.png>
>>
>> Thanks
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.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
>> http://www.freeswitch.org
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.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
>> http://www.freeswitch.org
>>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170608/791c9b7b/attachment.html>

From v.zakhozhai at gmail.com  Thu Jun  8 11:43:47 2017
From: v.zakhozhai at gmail.com (Vladyslav Zakhozhai)
Date: Thu, 8 Jun 2017 14:43:47 +0300
Subject: [Freeswitch-users] XML curl "not found" response
In-Reply-To: <CALXCt0pjCHuwOAY-Z-VahG1JPKVDdQo1oo6HR4WPC=pCMAvxoQ@mail.gmail.com>
References: <CANsY4DJ3mg0K5vzf1KyOEQXMA6w4Emu2vzWkT709APVJ2j7PEg@mail.gmail.com>
 <CALXCt0pjCHuwOAY-Z-VahG1JPKVDdQo1oo6HR4WPC=pCMAvxoQ@mail.gmail.com>
Message-ID: <CANsY4DLd3XADNk6yjFgXg-W2TEVi+RkP3rJMkJ5z-1FUq7Ktug@mail.gmail.com>

Giovanni, thank you for your reply.

Yes of course, I've read it. And the main question is - what is appropriate
way to inform freeswitch with "not found". And once more why freeswitch
treats 404 response as an error?

E.g. freeswitch requests configuration for non-existent user. Here we have
two options:
  1. Freeswitch replies with XML not found and HTTP 200. Freeswitch
understands it pretty correct and in logs I can see that freeswitch warns
me it was not able to find user in some domain. This is ok and correct.

 2. Freeswitch replies with XML not found and HTTP 404. Freeswitch reports
an error. It do not wait HTTP 404 status code.

And once more about p.2 from freeswitch's log:

"[ERR] mod_xml_curl.c:315 Received HTTP error 404 trying to fetch ....."


2017-06-08 12:18 GMT+03:00 Giovanni Maruzzelli <gmaruzz at gmail.com>:

> """If it receives a valid response from your web application, then it will
> load the configuration just like it would if you had put it into the FreeSWITCH
> Configuration File <https://wiki.freeswitch.org/wiki/Freeswitch.xml>. If
> it receives an invalid or 404 *not found* response, then it will attempt
> to look for the file on disk instead."""
>
> """https://freeswitch.org/confluence/display/FREESWITCH/mod_
> xml_curl#mod_xml_curl-Section:notfound"""
>
> https://freeswitch.org/confluence/display/FREESWITCH/mod_xml_curl
>
>
> On 8 June 2017 at 11:07, Vladyslav Zakhozhai <v.zakhozhai at gmail.com>
> wrote:
>
>> Hi,
>>
>> I am curious what HTTP status xml curl server must provide when no result
>> is found for request with body
>>
>> <?xml version="1.0" encoding="UTF-8" standalone="no"?>
>> <document type="freeswitch/xml">
>>   <section name="result">
>>     <result status="not found" />
>>   </section>
>> </document>
>>
>>
>> I mean HTTP 200 or HTTP 404.
>>
>>
>> My problem is the following:
>>
>> 1. When freeswitch gets 404 for directory request it reports error:
>>
>> [ERR] mod_xml_curl.c:315 Received HTTP error 404 trying to fetch
>>
>>
>> 2. When freeswitch gets 200 with status "not found" on dialplan request
>> failover to file xml config does not occure.
>>
>> Thank you in advance.
>>
>> --
>> С уважением,
>> Владислав Захожай
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.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
>> http://www.freeswitch.org
>>
>
>
>
> --
>
> Sincerely,
>
> Giovanni Maruzzelli
> OpenTelecom.IT
> cell: +39 347 266 56 18
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org
>



-- 
С уважением,
Владислав Захожай
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170608/45d80559/attachment-0001.html>

From vma at vallimamod.org  Thu Jun  8 12:53:15 2017
From: vma at vallimamod.org (Vallimamod Abdullah)
Date: Thu, 8 Jun 2017 14:53:15 +0200
Subject: [Freeswitch-users] XML curl "not found" response
In-Reply-To: <CANsY4DLd3XADNk6yjFgXg-W2TEVi+RkP3rJMkJ5z-1FUq7Ktug@mail.gmail.com>
References: <CANsY4DJ3mg0K5vzf1KyOEQXMA6w4Emu2vzWkT709APVJ2j7PEg@mail.gmail.com>
 <CALXCt0pjCHuwOAY-Z-VahG1JPKVDdQo1oo6HR4WPC=pCMAvxoQ@mail.gmail.com>
 <CANsY4DLd3XADNk6yjFgXg-W2TEVi+RkP3rJMkJ5z-1FUq7Ktug@mail.gmail.com>
Message-ID: <F5E1A93A-DC47-4B6B-B3F5-CEC5A0C57CA1@vallimamod.org>

Hi,

The correct reply is HTTP 200.
Any non 200 body is discarded by mod_xml_curl and triggers a failover to a backup url if you have defined one or the static conf.

If the failover to static conf does not occur in your case with a 200 'not found' reply, you need to investigate why. Can you post any full debug log?

Best Regards,
-- 
Vallimamod Abdullah
SIP Solutions
VOIP Consulting
vma at sipsolutions.fr
.

> On 8 Jun 2017, at 13:43, Vladyslav Zakhozhai <v.zakhozhai at gmail.com> wrote:
> 
> Giovanni, thank you for your reply.
> 
> Yes of course, I've read it. And the main question is - what is appropriate way to inform freeswitch with "not found". And once more why freeswitch treats 404 response as an error?
> 
> E.g. freeswitch requests configuration for non-existent user. Here we have two options:
>   1. Freeswitch replies with XML not found and HTTP 200. Freeswitch understands it pretty correct and in logs I can see that freeswitch warns me it was not able to find user in some domain. This is ok and correct.
> 
>  2. Freeswitch replies with XML not found and HTTP 404. Freeswitch reports an error. It do not wait HTTP 404 status code.
> 
> And once more about p.2 from freeswitch's log:
> 
> "[ERR] mod_xml_curl.c:315 Received HTTP error 404 trying to fetch ....."
> 
> 
> 2017-06-08 12:18 GMT+03:00 Giovanni Maruzzelli <gmaruzz at gmail.com>:
> """If it receives a valid response from your web application, then it will load the configuration just like it would if you had put it into the FreeSWITCH Configuration File. If it receives an invalid or 404 not found response, then it will attempt to look for the file on disk instead."""
> 
> """https://freeswitch.org/confluence/display/FREESWITCH/mod_xml_curl#mod_xml_curl-Section:notfound"""
> 
> https://freeswitch.org/confluence/display/FREESWITCH/mod_xml_curl
> 
> 
> On 8 June 2017 at 11:07, Vladyslav Zakhozhai <v.zakhozhai at gmail.com> wrote:
> Hi,
> 
> I am curious what HTTP status xml curl server must provide when no result is found for request with body
> 
> <?xml version="1.0" encoding="UTF-8" standalone="no"?>
> <document type="freeswitch/xml">
>   <section name="result">
>     <result status="not found" />
>   </section>
> </document>
> 
> 
> I mean HTTP 200 or HTTP 404.
> 
> 
> My problem is the following:
> 
> 1. When freeswitch gets 404 for directory request it reports error:
> 
> [ERR] mod_xml_curl.c:315 Received HTTP error 404 trying to fetch
> 
> 
> 2. When freeswitch gets 200 with status "not found" on dialplan request failover to file xml config does not occure.
> 
> Thank you in advance.
> 
> -- 
> С уважением,
> Владислав Захожай



From v.zakhozhai at gmail.com  Thu Jun  8 14:30:39 2017
From: v.zakhozhai at gmail.com (Vladyslav Zakhozhai)
Date: Thu, 8 Jun 2017 17:30:39 +0300
Subject: [Freeswitch-users] XML curl "not found" response
In-Reply-To: <F5E1A93A-DC47-4B6B-B3F5-CEC5A0C57CA1@vallimamod.org>
References: <CANsY4DJ3mg0K5vzf1KyOEQXMA6w4Emu2vzWkT709APVJ2j7PEg@mail.gmail.com>
 <CALXCt0pjCHuwOAY-Z-VahG1JPKVDdQo1oo6HR4WPC=pCMAvxoQ@mail.gmail.com>
 <CANsY4DLd3XADNk6yjFgXg-W2TEVi+RkP3rJMkJ5z-1FUq7Ktug@mail.gmail.com>
 <F5E1A93A-DC47-4B6B-B3F5-CEC5A0C57CA1@vallimamod.org>
Message-ID: <CANsY4DKNQVp9BiL_WBG4zKtKnmEGS9-ZO2VKAmJxH+evtmBYJw@mail.gmail.com>

Vallimamod,

Thank you for you clear answer. I appreciate that. I though that 200 "not
found" is correct but I had some doubts. You cleared out this question for
me.

I'll play with dialplan and failover to XML static config little bit later
and give you feedback.


2017-06-08 15:53 GMT+03:00 Vallimamod Abdullah <vma at vallimamod.org>:

> Hi,
>
> The correct reply is HTTP 200.
> Any non 200 body is discarded by mod_xml_curl and triggers a failover to a
> backup url if you have defined one or the static conf.
>
> If the failover to static conf does not occur in your case with a 200 'not
> found' reply, you need to investigate why. Can you post any full debug log?
>
> Best Regards,
> --
> Vallimamod Abdullah
> SIP Solutions
> VOIP Consulting
> vma at sipsolutions.fr
> .
>
> > On 8 Jun 2017, at 13:43, Vladyslav Zakhozhai <v.zakhozhai at gmail.com>
> wrote:
> >
> > Giovanni, thank you for your reply.
> >
> > Yes of course, I've read it. And the main question is - what is
> appropriate way to inform freeswitch with "not found". And once more why
> freeswitch treats 404 response as an error?
> >
> > E.g. freeswitch requests configuration for non-existent user. Here we
> have two options:
> >   1. Freeswitch replies with XML not found and HTTP 200. Freeswitch
> understands it pretty correct and in logs I can see that freeswitch warns
> me it was not able to find user in some domain. This is ok and correct.
> >
> >  2. Freeswitch replies with XML not found and HTTP 404. Freeswitch
> reports an error. It do not wait HTTP 404 status code.
> >
> > And once more about p.2 from freeswitch's log:
> >
> > "[ERR] mod_xml_curl.c:315 Received HTTP error 404 trying to fetch ....."
> >
> >
> > 2017-06-08 12:18 GMT+03:00 Giovanni Maruzzelli <gmaruzz at gmail.com>:
> > """If it receives a valid response from your web application, then it
> will load the configuration just like it would if you had put it into the
> FreeSWITCH Configuration File. If it receives an invalid or 404 not found
> response, then it will attempt to look for the file on disk instead."""
> >
> > """https://freeswitch.org/confluence/display/FREESWITCH/
> mod_xml_curl#mod_xml_curl-Section:notfound"""
> >
> > https://freeswitch.org/confluence/display/FREESWITCH/mod_xml_curl
> >
> >
> > On 8 June 2017 at 11:07, Vladyslav Zakhozhai <v.zakhozhai at gmail.com>
> wrote:
> > Hi,
> >
> > I am curious what HTTP status xml curl server must provide when no
> result is found for request with body
> >
> > <?xml version="1.0" encoding="UTF-8" standalone="no"?>
> > <document type="freeswitch/xml">
> >   <section name="result">
> >     <result status="not found" />
> >   </section>
> > </document>
> >
> >
> > I mean HTTP 200 or HTTP 404.
> >
> >
> > My problem is the following:
> >
> > 1. When freeswitch gets 404 for directory request it reports error:
> >
> > [ERR] mod_xml_curl.c:315 Received HTTP error 404 trying to fetch
> >
> >
> > 2. When freeswitch gets 200 with status "not found" on dialplan request
> failover to file xml config does not occure.
> >
> > Thank you in advance.
> >
> > --
> > С уважением,
> > Владислав Захожай
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org
>



-- 
С уважением,
Владислав Захожай
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170608/d5b2aed5/attachment.html>

From richardm at cellularity.co.uk  Thu Jun  8 10:33:44 2017
From: richardm at cellularity.co.uk (Richard Melville)
Date: Thu, 8 Jun 2017 11:33:44 +0100
Subject: [Freeswitch-users] Freeswitch Dependencies
Message-ID: <CAPeGcy8gMcdbc3YkEBnHxe0mgYgS+xWTt-PmXdGGtxY=oPjpGA@mail.gmail.com>

I haven't used or built Freeswitch yet, but I'm getting closer to that
point.  I will be building from source.  Having looked at the documentation
I can see that Debian is preferred, and the use of distros in general.
However, I don't use distros, but rather build my own systems from scratch.

I've looked at the Centos page ( https://freeswitch.org/
confluence/display/FREESWITCH/CentOS+7+and+RHEL+7 ) and under "Building
from source" there appears to be a list of dependencies.  I already have
most of those dependencies installed (other than the codecs) but I have
three questions which maybe somebody can answer.

The first is: can openssl be substituted with libressl, which I use?

The second is: "mongo-c-driver-devel" suggests that mongodb is a dependency
of Freeswitch.  I've seen no mention of mongodb anywhere in either the
book, or the documentation generally, so why is this listed as a dependency?

The third is: there is no mention of package version numbers anywhere, so
how can I find if there are any issues with particular versions?

Any help gratefully received.

-- 
Richard Melville
Systems Architect
cellularity.co.uk
stellarsystem.wordpress.com
+44 20 33 555 305
+44 7957 836330
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170608/ef9318ad/attachment.html>

From mike at jerris.com  Thu Jun  8 15:15:47 2017
From: mike at jerris.com (Michael Jerris)
Date: Thu, 08 Jun 2017 15:15:47 +0000
Subject: [Freeswitch-users] BugHunt
In-Reply-To: <CAHtxdDddZkiByD-VEJaKD03Pn2WkZzkvqpqN2xhx2g79y9dGTQ@mail.gmail.com>
References: <CAHtxdDddZkiByD-VEJaKD03Pn2WkZzkvqpqN2xhx2g79y9dGTQ@mail.gmail.com>
Message-ID: <CAEwTsAh9HR3=KMwpuTeTDaV87HwMxxG9SArdWBB7u_zT-L_GRQ@mail.gmail.com>

if you have things to discuss we can... ping me on hipchat and i'll call in
On Thu, Jun 8, 2017 at 3:16 AM Sergey Safarov <s.safarov at gmail.com> wrote:

> Is BugHunt will be today?
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170608/862a4dfe/attachment-0001.html>

From mike at jerris.com  Thu Jun  8 15:20:29 2017
From: mike at jerris.com (Michael Jerris)
Date: Thu, 08 Jun 2017 15:20:29 +0000
Subject: [Freeswitch-users] Re-invite Glare issue - FS ending call
In-Reply-To: <CANHADwNyoAX+nSwRADCXeLMxJx_+T=-=4bJxNbOUVyTPoo7WTQ@mail.gmail.com>
References: <CANHADwNmo1FV6GO6xfhBhK8SfwGDZVbOXhzmBPr8JaGng71k7A@mail.gmail.com>
 <72A53E3A-47FC-498E-AF90-2B39448DE52A@jerris.com>
 <CANHADwPZtVGd0c5n7TvWcdJMJt9SZKxPQde7fHFBk-Xon4o3UQ@mail.gmail.com>
 <CANHADwNyoAX+nSwRADCXeLMxJx_+T=-=4bJxNbOUVyTPoo7WTQ@mail.gmail.com>
Message-ID: <CAEwTsAigBQTbJu8zcnX_sNkLe20MTj2V-9CKBb7E4tC_y-e4ag@mail.gmail.com>

the 491 is in response to their re-invite, in which we are the UAS.  For
our re-invite we send we are UAC.

On Thu, Jun 8, 2017 at 5:41 AM Jason Bedward <mase2hot at gmail.com> wrote:

> Looking again as this according to RFC 3261 Freeswitch is doing this wrong?
>
> For the B leg my FS is the UAC, and if so when the invite comes in from
> the B leg it shouldn't respond 491 as that should only be done from the
> UAS.
>
> Unless I'm wrong with this?
>
> On Thu, Jun 8, 2017 at 8:14 AM, Jason Bedward <mase2hot at gmail.com> wrote:
>
>> My provider is sending the reinvite on the A leg FS the sends this on to
>> the B leg. At same time the B leg provider sends a reinvite.
>>
>> I dont believe FS is sending the reinvite on its own accord. Its only
>> because of the A leg and that we are using bypass media. So I dont think
>> session timers will make a difference.
>>
>> FYI A leg and B leg provider are the same company although different
>> servers.
>>
>> On 7 Jun 2017 16:57, "Michael Jerris" <mike at jerris.com> wrote:
>>
>>> why are they both sending a re-invite at the same time?  This is called
>>> glare, we seem to be handling it properly, the provider seems not to be.
>>> The easiest fix here is probably to figure out why we are both sending
>>> re-invite at the same time, maybe session timers and the provider is broken
>>> and trying to send it when we said we would.
>>>
>>> On Jun 7, 2017, at 11:49 AM, Jason Bedward <mase2hot at gmail.com> wrote:
>>>
>>> Hi,
>>>
>>> I have an issue with re-invites on some calls. I'm using 1.6.15 and
>>> these calls are using bypass media. I have Kamailio as inbound SBC but FS
>>> connecting directly to my provider for B leg.
>>>
>>>
>>>    - After 5 minutes plus on some calls my provider sends re-invite on
>>>    A leg
>>>    - FS then sends this re-invite to the B leg
>>>    - At the same time the B leg sends a Re-invite
>>>    - FS replys 491
>>>    - B leg provider replys 100, then 500 (with retry in the 500)
>>>    - FS send ACK and then BYE
>>>
>>> Not sure what setting I need to change or can change infact to either
>>> retry the invite in accordance with the 500 retry request. Or something
>>> else to stop the call ending...
>>>
>>> <image.png>
>>>
>>> Thanks
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.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
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.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
>>> http://www.freeswitch.org
>>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170608/d970e491/attachment.html>

From bipin at xbipin.com  Thu Jun  8 15:41:52 2017
From: bipin at xbipin.com (Bipin Patel)
Date: Thu, 8 Jun 2017 19:41:52 +0400
Subject: [Freeswitch-users] weird dtmf forwarding issue
Message-ID: <f823cdd8-a354-3698-d8c5-484b1b4cc8b4@xbipin.com>

hi,

i have 2 setups running FS with the exact same profile and dialplan, one 
running on windows and the other on a raspberry pi, both receive calls 
from registered clients and forward to a remote gateway, the problem is 
when client sends dtmf, the windows FS forwards to gateway without 
issues but on the rpi most of the times fs isnt able to detect dtmf or 
at times detects but doesnt or partially sends to gateway.
i have been banging my head with this from a few days but no idea whats 
going wrong, the client sending to rpi FS i even made him send call to 
gateway directly and then that works so this rules out client issue, no 
idea whats wrong with FS, both are on the latest build

im using g711u throughout

-- 
Regards,
Bipin


------------------------------------------------------------------------
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170608/7604a6ba/attachment.html>

From mike at jerris.com  Thu Jun  8 15:48:57 2017
From: mike at jerris.com (Michael Jerris)
Date: Thu, 08 Jun 2017 15:48:57 +0000
Subject: [Freeswitch-users] question about HA solution
In-Reply-To: <CADcrYGKK=cTmVzG5oY0MV3b+4CLqg0=YzGvCiDZNZdBqQe3mMA@mail.gmail.com>
References: <CADcrYG+RHGQ2fC7dSKK9x9WfVvYcTczOtbppCLgttRCfMGqArA@mail.gmail.com>
 <CAFLo2QxxbZ5Ged0J0p9_VfQ2nkGtuV0CDp9G763EjjuvqUv8iw@mail.gmail.com>
 <CADcrYG+sO0+WYUtYVN6pkpU-5SwyfurfshdUdhxb-c4Az1s49Q@mail.gmail.com>
 <8A13E0EC-FA13-4267-80F6-CE1A8E8360CF@jerris.com>
 <CADcrYGLbN1Ut6riF1Lf7UvwG9ypvR2ekWxt1m=e5BmqOzP_c=w@mail.gmail.com>
 <E55363B0-487F-4F65-8F3F-78620AB1AB73@jerris.com>
 <CADcrYGKK=cTmVzG5oY0MV3b+4CLqg0=YzGvCiDZNZdBqQe3mMA@mail.gmail.com>
Message-ID: <CAEwTsAhpLWcAoK8jfaEZ+HS2NeNLLEgbyAZkNRiTJ4RMNRet4Q@mail.gmail.com>

check your db logs as nothing we are doing should be clearing those.

On Thu, Jun 8, 2017 at 4:08 AM Denys Pozniak <denys.pozniak at crazycall.com>
wrote:

> Hello!
>
> My configs:
>
> *switch.conf.xml*
>
>     <param name="switchname" value="freeswitch01"/>
>     <param name="core-recovery-db-dsn"
> value="odbc://freeswitch:root:ubuntu"/>
>     <param name="core-db-dsn" value="odbc://freeswitch:root:ubuntu"/>
>     <param name="auto-create-schemas" value="false"/>
>     <param name="auto-clear-sql" value="false"/>
>
>
> *external.conf.xml*
>
>    <param name="track-calls" value="true"/>
>    <param name="odbc-dsn" value="odbc://freeswitch:root:ubuntu"/>
>
>
> On 7 June 2017 at 17:35, Michael Jerris <mike at jerris.com> wrote:
>
>> That param should keep it from doing so, if its not you are not setting
>> it somehow or something else is wiping the db.
>>
>> On Jun 5, 2017, at 1:50 PM, Denys Pozniak <denys.pozniak at crazycall.com>
>> wrote:
>>
>> Yes, correct. But when you restart FS on slave, it will erase database.
>> And option auto-clear-sql=false not working for me.
>>
>> On Jun 5, 2017 6:32 PM, "Michael Jerris" <mike at jerris.com> wrote:
>>
>> recovered calls will get new entries in the table.
>>
>> On Jun 5, 2017, at 7:41 AM, Denys Pozniak <denys.pozniak at crazycall.com>
>> wrote:
>>
>> Hello!
>>
>> Thank you *Raymond* about your explanation, but I dont agree with some
>> point:
>> *If it really need an answer about your question -- "if it is possible to
>> move calls back".   I think it's unnecessary.*  - in my case I have two
>> not equal servers, so I need to have only one as a master.
>> If switchover happens I need to have ability to restore master back.
>>
>> Thank you *Luis* for your link, you can do simple test to understand
>> what I am talking about: do call -> check on master and slave #show
>> channels -> restart FS on slave -> check on master #show channels. In my
>> case I dont see any active calls after this, so restoring back is not
>> possible.
>>
>>
>>
>> On 3 June 2017 at 22:16, Luis Daniel Lucio Quiroz <
>> luis.daniel.lucio at gmail.com> wrote:
>>
>>> You may want to read this article.
>>>
>>> http://inside-out.xyz/technology/how-to-configure-freeswitch-for-ha.html
>>>
>>> Le 31 mai 2017 6:29 PM, "Denys Pozniak" <denys.pozniak at crazycall.com> a
>>> écrit :
>>>
>>> Hello!
>>>
>>> I built FS HA solution based on keepalived and mysql master-master.
>>> It works ok generally, but as I understand FS after restarting cleaning
>>> own database.
>>>
>>> So when node1 fails calls jump to node2, after script restarts node1 it
>>> is not possible to move calls back.
>>>
>>> Tried options in switch.conf.xml, but no luck:
>>>
>>>     <param name="auto-create-schemas" value="false"/>
>>>     <param name="auto-clear-sql" value="false"/>
>>>
>>> Is there is a way to solve this?
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.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
>> http://www.freeswitch.org
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170608/13995ba7/attachment-0001.html>

From mike at jerris.com  Thu Jun  8 16:01:54 2017
From: mike at jerris.com (Michael Jerris)
Date: Thu, 8 Jun 2017 12:01:54 -0400
Subject: [Freeswitch-users] Freeswitch Dependencies
In-Reply-To: <CAPeGcy8gMcdbc3YkEBnHxe0mgYgS+xWTt-PmXdGGtxY=oPjpGA@mail.gmail.com>
References: <CAPeGcy8gMcdbc3YkEBnHxe0mgYgS+xWTt-PmXdGGtxY=oPjpGA@mail.gmail.com>
Message-ID: <091C458B-44F6-42F9-90A8-FFACEFC534D5@jerris.com>


> On Jun 8, 2017, at 6:33 AM, Richard Melville <richardm at cellularity.co.uk> wrote:
> 
> I haven't used or built Freeswitch yet, but I'm getting closer to that point.  I will be building from source.  Having looked at the documentation I can see that Debian is preferred, and the use of distros in general.  However, I don't use distros, but rather build my own systems from scratch.
> 
> I've looked at the Centos page ( https://freeswitch.org/confluence/display/FREESWITCH/CentOS+7+and+RHEL+7 <https://freeswitch.org/confluence/display/FREESWITCH/CentOS+7+and+RHEL+7> ) and under "Building from source" there appears to be a list of dependencies.  I already have most of those dependencies installed (other than the codecs) but I have three questions which maybe somebody can answer.
> 
> The first is: can openssl be substituted with libressl, which I use?

Not sure, depends if libressl has the required pieces we need for dtls-srtp and all the required ciphers required by the browsers for webrtc.  

> 
> The second is: "mongo-c-driver-devel" suggests that mongodb is a dependency of Freeswitch.  I've seen no mention of mongodb anywhere in either the book, or the documentation generally, so why is this listed as a dependency?

Its a module, if you don’t want that module, its not needed.

> 
> The third is: there is no mention of package version numbers anywhere, so how can I find if there are any issues with particular versions?

We don’t test a vast array of different package versions, we do testing based on the ones for the distros we package for.  The versions in Debian 8 are well tested, other versions are much less well tested or not tested at all.  As for other libs, use the ones in our stash project for dep libs when not otherwise available.  Creating extensive documentation for building on your own distro would be far more work than even adding support for a new distro, and we don’t have any plans to create that.  I’m happy to respond to some specific questions, but there are limits to the amount of time that it makes sense for us to spend on issues like this for a single person.

> 
> Any help gratefully received.
> 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170608/4b37b124/attachment.html>

From v.zakhozhai at gmail.com  Thu Jun  8 16:05:31 2017
From: v.zakhozhai at gmail.com (Vladyslav Zakhozhai)
Date: Thu, 8 Jun 2017 19:05:31 +0300
Subject: [Freeswitch-users] XML curl "not found" response
In-Reply-To: <CANsY4DKNQVp9BiL_WBG4zKtKnmEGS9-ZO2VKAmJxH+evtmBYJw@mail.gmail.com>
References: <CANsY4DJ3mg0K5vzf1KyOEQXMA6w4Emu2vzWkT709APVJ2j7PEg@mail.gmail.com>
 <CALXCt0pjCHuwOAY-Z-VahG1JPKVDdQo1oo6HR4WPC=pCMAvxoQ@mail.gmail.com>
 <CANsY4DLd3XADNk6yjFgXg-W2TEVi+RkP3rJMkJ5z-1FUq7Ktug@mail.gmail.com>
 <F5E1A93A-DC47-4B6B-B3F5-CEC5A0C57CA1@vallimamod.org>
 <CANsY4DKNQVp9BiL_WBG4zKtKnmEGS9-ZO2VKAmJxH+evtmBYJw@mail.gmail.com>
Message-ID: <CANsY4D+PoybQzS2mNZNfdSSjCxtW8q6ara8c6RRpSjjRptsBaw@mail.gmail.com>

That was my mistake. With HTTP 404 dialplan fails over to local static XML
Dialplan.
Thank you all.

2017-06-08 17:30 GMT+03:00 Vladyslav Zakhozhai <v.zakhozhai at gmail.com>:

> Vallimamod,
>
> Thank you for you clear answer. I appreciate that. I though that 200 "not
> found" is correct but I had some doubts. You cleared out this question for
> me.
>
> I'll play with dialplan and failover to XML static config little bit later
> and give you feedback.
>
>
> 2017-06-08 15:53 GMT+03:00 Vallimamod Abdullah <vma at vallimamod.org>:
>
>> Hi,
>>
>> The correct reply is HTTP 200.
>> Any non 200 body is discarded by mod_xml_curl and triggers a failover to
>> a backup url if you have defined one or the static conf.
>>
>> If the failover to static conf does not occur in your case with a 200
>> 'not found' reply, you need to investigate why. Can you post any full debug
>> log?
>>
>> Best Regards,
>> --
>> Vallimamod Abdullah
>> SIP Solutions
>> VOIP Consulting
>> vma at sipsolutions.fr
>> .
>>
>> > On 8 Jun 2017, at 13:43, Vladyslav Zakhozhai <v.zakhozhai at gmail.com>
>> wrote:
>> >
>> > Giovanni, thank you for your reply.
>> >
>> > Yes of course, I've read it. And the main question is - what is
>> appropriate way to inform freeswitch with "not found". And once more why
>> freeswitch treats 404 response as an error?
>> >
>> > E.g. freeswitch requests configuration for non-existent user. Here we
>> have two options:
>> >   1. Freeswitch replies with XML not found and HTTP 200. Freeswitch
>> understands it pretty correct and in logs I can see that freeswitch warns
>> me it was not able to find user in some domain. This is ok and correct.
>> >
>> >  2. Freeswitch replies with XML not found and HTTP 404. Freeswitch
>> reports an error. It do not wait HTTP 404 status code.
>> >
>> > And once more about p.2 from freeswitch's log:
>> >
>> > "[ERR] mod_xml_curl.c:315 Received HTTP error 404 trying to fetch ....."
>> >
>> >
>> > 2017-06-08 12:18 GMT+03:00 Giovanni Maruzzelli <gmaruzz at gmail.com>:
>> > """If it receives a valid response from your web application, then it
>> will load the configuration just like it would if you had put it into the
>> FreeSWITCH Configuration File. If it receives an invalid or 404 not found
>> response, then it will attempt to look for the file on disk instead."""
>> >
>> > """https://freeswitch.org/confluence/display/FREESWITCH/mod_
>> xml_curl#mod_xml_curl-Section:notfound"""
>> >
>> > https://freeswitch.org/confluence/display/FREESWITCH/mod_xml_curl
>> >
>> >
>> > On 8 June 2017 at 11:07, Vladyslav Zakhozhai <v.zakhozhai at gmail.com>
>> wrote:
>> > Hi,
>> >
>> > I am curious what HTTP status xml curl server must provide when no
>> result is found for request with body
>> >
>> > <?xml version="1.0" encoding="UTF-8" standalone="no"?>
>> > <document type="freeswitch/xml">
>> >   <section name="result">
>> >     <result status="not found" />
>> >   </section>
>> > </document>
>> >
>> >
>> > I mean HTTP 200 or HTTP 404.
>> >
>> >
>> > My problem is the following:
>> >
>> > 1. When freeswitch gets 404 for directory request it reports error:
>> >
>> > [ERR] mod_xml_curl.c:315 Received HTTP error 404 trying to fetch
>> >
>> >
>> > 2. When freeswitch gets 200 with status "not found" on dialplan request
>> failover to file xml config does not occure.
>> >
>> > Thank you in advance.
>> >
>> > --
>> > С уважением,
>> > Владислав Захожай
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.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
>> http://www.freeswitch.org
>>
>
>
>
> --
> С уважением,
> Владислав Захожай
>
>


-- 
С уважением,
Владислав Захожай
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170608/a346d91c/attachment.html>

From anthony.minessale at gmail.com  Thu Jun  8 16:16:12 2017
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 08 Jun 2017 16:16:12 +0000
Subject: [Freeswitch-users] Re-invite Glare issue - FS ending call
In-Reply-To: <CAEwTsAigBQTbJu8zcnX_sNkLe20MTj2V-9CKBb7E4tC_y-e4ag@mail.gmail.com>
References: <CANHADwNmo1FV6GO6xfhBhK8SfwGDZVbOXhzmBPr8JaGng71k7A@mail.gmail.com>
 <72A53E3A-47FC-498E-AF90-2B39448DE52A@jerris.com>
 <CANHADwPZtVGd0c5n7TvWcdJMJt9SZKxPQde7fHFBk-Xon4o3UQ@mail.gmail.com>
 <CANHADwNyoAX+nSwRADCXeLMxJx_+T=-=4bJxNbOUVyTPoo7WTQ@mail.gmail.com>
 <CAEwTsAigBQTbJu8zcnX_sNkLe20MTj2V-9CKBb7E4tC_y-e4ag@mail.gmail.com>
Message-ID: <CAKbxfG83om++EBxkvpaN-E0MssjfaXMLULXn1XUdCOgRsq_xiQ@mail.gmail.com>

As always dont report issues via email.
https://freeswitch.org/jira


On Thu, Jun 8, 2017 at 10:21 AM Michael Jerris <mike at jerris.com> wrote:

> the 491 is in response to their re-invite, in which we are the UAS.  For
> our re-invite we send we are UAC.
>
> On Thu, Jun 8, 2017 at 5:41 AM Jason Bedward <mase2hot at gmail.com> wrote:
>
>> Looking again as this according to RFC 3261 Freeswitch is doing this
>> wrong?
>>
>> For the B leg my FS is the UAC, and if so when the invite comes in from
>> the B leg it shouldn't respond 491 as that should only be done from the
>> UAS.
>>
>> Unless I'm wrong with this?
>>
>> On Thu, Jun 8, 2017 at 8:14 AM, Jason Bedward <mase2hot at gmail.com> wrote:
>>
>>> My provider is sending the reinvite on the A leg FS the sends this on to
>>> the B leg. At same time the B leg provider sends a reinvite.
>>>
>>> I dont believe FS is sending the reinvite on its own accord. Its only
>>> because of the A leg and that we are using bypass media. So I dont think
>>> session timers will make a difference.
>>>
>>> FYI A leg and B leg provider are the same company although different
>>> servers.
>>>
>>> On 7 Jun 2017 16:57, "Michael Jerris" <mike at jerris.com> wrote:
>>>
>>>> why are they both sending a re-invite at the same time?  This is called
>>>> glare, we seem to be handling it properly, the provider seems not to be.
>>>> The easiest fix here is probably to figure out why we are both sending
>>>> re-invite at the same time, maybe session timers and the provider is broken
>>>> and trying to send it when we said we would.
>>>>
>>>> On Jun 7, 2017, at 11:49 AM, Jason Bedward <mase2hot at gmail.com> wrote:
>>>>
>>>> Hi,
>>>>
>>>> I have an issue with re-invites on some calls. I'm using 1.6.15 and
>>>> these calls are using bypass media. I have Kamailio as inbound SBC but FS
>>>> connecting directly to my provider for B leg.
>>>>
>>>>
>>>>    - After 5 minutes plus on some calls my provider sends re-invite on
>>>>    A leg
>>>>    - FS then sends this re-invite to the B leg
>>>>    - At the same time the B leg sends a Re-invite
>>>>    - FS replys 491
>>>>    - B leg provider replys 100, then 500 (with retry in the 500)
>>>>    - FS send ACK and then BYE
>>>>
>>>> Not sure what setting I need to change or can change infact to either
>>>> retry the invite in accordance with the 500 retry request. Or something
>>>> else to stop the call ending...
>>>>
>>>> <image.png>
>>>>
>>>> Thanks
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.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
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.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
>>>> http://www.freeswitch.org
>>>>
>>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.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
>> http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org

-- 
Anthony Minessale II       ♬ @anthmfs  ♬ @FreeSWITCH  ♬

☞ http://freeswitch.org/  ☞ http://cluecon.com/  ☞
http://twitter.com/FreeSWITCH
☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+
<http://freeswitch.org/g+>*

ClueCon Weekly Development Call
☎ sip:888 at conference.freeswitch.org  ☎ +19193869900

https://www.youtube.com/watch?v=9XXgW34t40s
https://www.youtube.com/watch?v=NLaDpGQuZDA
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170608/9e65bf02/attachment-0001.html>

From krice at freeswitch.org  Thu Jun  8 17:03:39 2017
From: krice at freeswitch.org (Ken Rice)
Date: Thu, 8 Jun 2017 12:03:39 -0500
Subject: [Freeswitch-users] Freeswitch Dependencies
In-Reply-To: <CAPeGcy8gMcdbc3YkEBnHxe0mgYgS+xWTt-PmXdGGtxY=oPjpGA@mail.gmail.com>
References: <CAPeGcy8gMcdbc3YkEBnHxe0mgYgS+xWTt-PmXdGGtxY=oPjpGA@mail.gmail.com>
Message-ID: <49cb01d2e079$2dc6beb0$89543c10$@freeswitch.org>

To partially address some of your questions.

 

1.	The reason we recommend specific distros as it ends up being a known quantity from a supportability stand point. This means that we test builds on those distros with what they provide as versions for system supplied packages. We highly recommend Debian8 for the least pain as that is where primary development and bug fixes occur and from there they are ported to address any differences in what other systems support. Use of distros in general are encouraged for this reason. The one off support for a custom rolled system is beyond the capabilities of the project to effectively support with our small team.
2.	As far as libressl vs openssl. I don’t know of anyone actually testing this to see if it will work or not. We specifically test against versions of OpenSSL. The later versions of OpenSSL are required due not only to the security concerns you are trying to address with LibreSSL, but due to newer encryption technologies being included in the later versions of OpenSSL that may or may not be included in LibreSSL. (again this comes down to supporting what is available and most widely used on the supported distributions.)
3.	As far as mongo-c-driver-devel, you will find that there are several dependancies like that may or may not be required for you to build FreeSWITCH depending on which modules available that you may want to build. While some of the deps are required to build the core (such as sqlite et al), however others are only required if you plan on building a module that requires them such as mod_mongo. Other examples are mod_flite, mod_cepstral, mod_hiredis, mod_osp and mod_ladspa. (mod_ladspa itself is an example of a module that only works on Linux unless someone gotten around to porting this linux specific interface over to Windows and possible some of the BSDs)

Now that being said you can review the 

 

 

From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Richard Melville
Sent: Thursday, June 8, 2017 5:34 AM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Freeswitch Dependencies

 

I haven't used or built Freeswitch yet, but I'm getting closer to that point.  I will be building from source.  Having looked at the documentation I can see that Debian is preferred, and the use of distros in general.  However, I don't use distros, but rather build my own systems from scratch.

I've looked at the Centos page ( https://freeswitch.org/confluence/display/FREESWITCH/CentOS+7+and+RHEL+7 ) and under "Building from source" there appears to be a list of dependencies.  I already have most of those dependencies installed (other than the codecs) but I have three questions which maybe somebody can answer.

The first is: can openssl be substituted with libressl, which I use?

The second is: "mongo-c-driver-devel" suggests that mongodb is a dependency of Freeswitch.  I've seen no mention of mongodb anywhere in either the book, or the documentation generally, so why is this listed as a dependency?

The third is: there is no mention of package version numbers anywhere, so how can I find if there are any issues with particular versions?

 

Any help gratefully received.

-- 

Richard Melville

Systems Architect

cellularity.co.uk <http://cellularity.co.uk> 

stellarsystem.wordpress.com <http://stellarsystem.wordpress.com> 

+44 20 33 555 305

+44 7957 836330

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170608/a9dd7d14/attachment.html>

From brian at freeswitch.org  Thu Jun  8 20:23:58 2017
From: brian at freeswitch.org (Brian West)
Date: Thu, 8 Jun 2017 15:23:58 -0500
Subject: [Freeswitch-users] weird dtmf forwarding issue
In-Reply-To: <f823cdd8-a354-3698-d8c5-484b1b4cc8b4@xbipin.com>
References: <f823cdd8-a354-3698-d8c5-484b1b4cc8b4@xbipin.com>
Message-ID: <CAEJMVkDDQX=Ni3AOQxQOo=joGoAeYroxioVP=cu9w6Y2m7kLPA@mail.gmail.com>

Its a little unclear, you say 'detect', so I have to assume you're running
a dtmf detector.  You're running on x86 vs arm, so I can only assume its
possibly that differences in architecture are at play here.

/b


On Thu, Jun 8, 2017 at 10:41 AM, Bipin Patel <bipin at xbipin.com> wrote:

> hi,
>
> i have 2 setups running FS with the exact same profile and dialplan, one
> running on windows and the other on a raspberry pi, both receive calls from
> registered clients and forward to a remote gateway, the problem is when
> client sends dtmf, the windows FS forwards to gateway without issues but on
> the rpi most of the times fs isnt able to detect dtmf or at times detects
> but doesnt or partially sends to gateway.
> i have been banging my head with this from a few days but no idea whats
> going wrong, the client sending to rpi FS i even made him send call to
> gateway directly and then that works so this rules out client issue, no
> idea whats wrong with FS, both are on the latest build
>
> im using g711u throughout
>
> --
> Regards,
> Bipin
>
>
> ------------------------------
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org
>



-- 

*Brian West*
brian at freeswitch.org

*Twitter: @FreeSWITCH , @briankwest*

http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

Book a phone call (CST) <https://freeswitch.com/appointment>

Allison prompts for FreeSWITCH:

*https://www.gofundme.com/allison-prompts-for-freeswitch*
<https://www.gofundme.com/allison-prompts-for-freeswitch>

Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
/r/freeswitch <https://www.reddit.com/r/freeswitch>

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*Skype:*briankwest
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170608/ff45feb1/attachment.html>

From bipin at xbipin.com  Thu Jun  8 21:10:31 2017
From: bipin at xbipin.com (Bipin Patel)
Date: Fri, 09 Jun 2017 01:10:31 +0400
Subject: [Freeswitch-users] weird dtmf forwarding issue
In-Reply-To: <CAEJMVkDDQX=Ni3AOQxQOo=joGoAeYroxioVP=cu9w6Y2m7kLPA@mail.gmail.com>
References: <f823cdd8-a354-3698-d8c5-484b1b4cc8b4@xbipin.com>
 <CAEJMVkDDQX=Ni3AOQxQOo=joGoAeYroxioVP=cu9w6Y2m7kLPA@mail.gmail.com>
Message-ID: <15c898b5d58.2779.b07ebdf329620b8089087c7205b03f01@xbipin.com>

Detect as in when fs receives a DTMF it prints in cli, when it doesn't it 
didn't detect a DTMF even though client is sending it. First I tried 
sending inband and used start DTMF but that wasn't too good so moved to the 
rfc one and then finally ended up using info. I made some progress on this, 
wanted to know can fs receive DTMF as rfc or info and send info or rfc to 
gateway, something like a intercept and forward as a new method? When I set 
DTMF to rfc in profile and client sends in info then nothing goes to 
gateway, if client sends info and I set info in profile then it works. In 
summary I want to accept info and rfc and want to send to gateway in 
whatever mode they support rather than force client to send in a particular 
format.

Also is there a way to make DTMF timings a little relaxed, I have another 
client that uses fs as PBX and calls come to it using a pstn to VoIP 
gateway and due to the exchange under going hardware upgrades the DTMF 
timings are a little out or the trailing sound is a bit broken and most of 
the times the pstn gateway doesn't detect so I switched it to inband and if 
I can tune fs to accept a little messed up DTMF then it would make it work.

I had a similar issue in a different country when the exchange was moving 
to etsi fsk for caller ID and it took months for the company to figure out 
the issue and sort it out


On June 9, 2017 12:27:04 AM Brian West <brian at freeswitch.org> wrote:

> Its a little unclear, you say 'detect', so I have to assume you're running
> a dtmf detector.  You're running on x86 vs arm, so I can only assume its
> possibly that differences in architecture are at play here.
>
> /b
>
>
> On Thu, Jun 8, 2017 at 10:41 AM, Bipin Patel <bipin at xbipin.com> wrote:
>
>> hi,
>>
>> i have 2 setups running FS with the exact same profile and dialplan, one
>> running on windows and the other on a raspberry pi, both receive calls from
>> registered clients and forward to a remote gateway, the problem is when
>> client sends dtmf, the windows FS forwards to gateway without issues but on
>> the rpi most of the times fs isnt able to detect dtmf or at times detects
>> but doesnt or partially sends to gateway.
>> i have been banging my head with this from a few days but no idea whats
>> going wrong, the client sending to rpi FS i even made him send call to
>> gateway directly and then that works so this rules out client issue, no
>> idea whats wrong with FS, both are on the latest build
>>
>> im using g711u throughout
>>
>> --
>> Regards,
>> Bipin
>>
>>
>> ------------------------------
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.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
>> http://www.freeswitch.org
>>
>
>
>
> --
>
> *Brian West*
> brian at freeswitch.org
>
> *Twitter: @FreeSWITCH , @briankwest*
>
> http://www.freeswitchbook.com
> http://www.freeswitchcookbook.com
>
> Book a phone call (CST) <https://freeswitch.com/appointment>
>
> Allison prompts for FreeSWITCH:
>
> *https://www.gofundme.com/allison-prompts-for-freeswitch*
> <https://www.gofundme.com/allison-prompts-for-freeswitch>
>
> Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
> /r/freeswitch <https://www.reddit.com/r/freeswitch>
>
> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
> *Skype:*briankwest
>
>
>
> ----------
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170609/fffd94f3/attachment-0001.html>

From krice at freeswitch.org  Thu Jun  8 21:34:07 2017
From: krice at freeswitch.org (Ken Rice)
Date: Thu, 8 Jun 2017 16:34:07 -0500
Subject: [Freeswitch-users] Freeswitch Dependencies
In-Reply-To: <CAPeGcy8gMcdbc3YkEBnHxe0mgYgS+xWTt-PmXdGGtxY=oPjpGA@mail.gmail.com>
References: <CAPeGcy8gMcdbc3YkEBnHxe0mgYgS+xWTt-PmXdGGtxY=oPjpGA@mail.gmail.com>
Message-ID: <4bdb01d2e09e$f60fa240$e22ee6c0$@freeswitch.org>

And as usual I got interrupted and clicked send before finishing a thought… (sorry about that but it helps demonstrate our time constraints)

 

But following up my earlier comments 

 

 

Now that being said you can review the freeswitch.spec file in the root of the source tree or the Debian packaging files in the debian dir also in the source tree to get a list of build deps and if min versions are required what those are.

 

Things without version numbering in those files are that way due to the current versions in the appropriate Distros being sufficient and Distros typically do not change APIs once theyhave released a  version (or on minor versions upgrades)_

 

 

 

From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Richard Melville
Sent: Thursday, June 8, 2017 5:34 AM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Freeswitch Dependencies

 

I haven't used or built Freeswitch yet, but I'm getting closer to that point.  I will be building from source.  Having looked at the documentation I can see that Debian is preferred, and the use of distros in general.  However, I don't use distros, but rather build my own systems from scratch.

I've looked at the Centos page ( https://freeswitch.org/confluence/display/FREESWITCH/CentOS+7+and+RHEL+7 ) and under "Building from source" there appears to be a list of dependencies.  I already have most of those dependencies installed (other than the codecs) but I have three questions which maybe somebody can answer.

The first is: can openssl be substituted with libressl, which I use?

The second is: "mongo-c-driver-devel" suggests that mongodb is a dependency of Freeswitch.  I've seen no mention of mongodb anywhere in either the book, or the documentation generally, so why is this listed as a dependency?

The third is: there is no mention of package version numbers anywhere, so how can I find if there are any issues with particular versions?

 

Any help gratefully received.

-- 

Richard Melville

Systems Architect

cellularity.co.uk <http://cellularity.co.uk> 

stellarsystem.wordpress.com <http://stellarsystem.wordpress.com> 

+44 20 33 555 305

+44 7957 836330

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170608/c2c9509d/attachment.html>

From david.villasmil.work at gmail.com  Fri Jun  9 05:48:21 2017
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Fri, 9 Jun 2017 01:48:21 -0400
Subject: [Freeswitch-users] Removing userpart of contact
Message-ID: <CAFGRPVo2jMJc==9SpeOPZjhgbxARMAwgQfiquuYUd-UOR7UeHw@mail.gmail.com>

Hello guys,

I ran into a situation where I need the contact to be like:

<sip:1.2.3.4:5060>

Meaning I need to remove the username, i've trying doing this but FS adds
the user as "mod_sofia"... is it possible to do this?


Thanks and Regards,

David Villasmil
email: david.villasmil.work at gmail.com
phone: +34669448337
ᐧ
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170609/b265d56b/attachment.html>

From hardikitpl at gmail.com  Fri Jun  9 10:08:39 2017
From: hardikitpl at gmail.com (Hardik Patel)
Date: Fri, 9 Jun 2017 15:38:39 +0530
Subject: [Freeswitch-users] Fax receive issue with t30 codec
In-Reply-To: <CAEJMVkDtY8VeG+14cB42wHee5fjgzsXE5trA31dtp=F4rxoeRg@mail.gmail.com>
References: <CAHdn4vjxr5vVoO7Oc1P0u-xg6dJNYBhjxU=y5iz6VP5_JEcUjA@mail.gmail.com>
 <CAEJMVkDtY8VeG+14cB42wHee5fjgzsXE5trA31dtp=F4rxoeRg@mail.gmail.com>
Message-ID: <CAHdn4vitAzqN2D9F0qSF5u8T+1D_PXQRX9NSaMKUauttVCrg-w@mail.gmail.com>

Hi Brian,

We have update Freeswitch version but still fax is not working using t30.

We are getting below time out related error.

2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:496
==============================================================================
2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:508 Fax processing not
successful - result (3) Timed out waiting for the first message.
2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:513 Remote station id:
2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:514 Local station id:
SpanDSP Fax Ident
2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:515 Pages transferred:
0
2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:517 Total fax pages:
0
2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:518 Image resolution:
0x0
2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:519 Transfer Rate:
14400
2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:521 ECM status
off
2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:522 remote country:
2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:523 remote vendor:
2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:524 remote model:
2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:526
==============================================================================


On Tue, Jun 6, 2017 at 7:30 PM, Brian West <brian at freeswitch.org> wrote:

> You'll have to use 1.6.17 if you ever want any faxing to work in all test
> cases.
>
> /b
>
> On Tue, Jun 6, 2017 at 6:11 AM, Hardik Patel <hardikitpl at gmail.com> wrote:
>
>> Hello,
>>
>> I am using opensips as entry point using dispatcher. opensips(
>> 127.0.0.1), i am routing call to freeswitch server (127.0.0.3).
>>
>> Now I am trying to receive fax, my issue is when i try to send fax in
>> softphone(Zoiper) from the log i am seeing that it is sending fax using t30
>> codec. and i am not receiving the fax at destination, is it because of
>> codec, should it only work with t38 codec? if that is the issue than how am
>> i be able to send the fax using t38 from zoiper?
>>
>> Here i am attaching the fs log with loglevel 9 and sip trace is also
>> enabled.
>>
>> 127.0.0.2  => carrier/provider IP
>> 123456789 => Fax number
>> test at gamil.com => Email Address
>> 127.0.0.4 =>UI IP
>>
>> Pastebin link:https://pastebin.freeswitch.org/view/9ec52715
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.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
>> http://www.freeswitch.org
>>
>
>
>
> --
>
> *Brian West*
> brian at freeswitch.org
>
> *Twitter: @FreeSWITCH , @briankwest*
>
> http://www.freeswitchbook.com
> http://www.freeswitchcookbook.com
>
> Book a phone call (CST) <https://freeswitch.com/appointment>
>
> Allison prompts for FreeSWITCH:
>
> *https://www.gofundme.com/allison-prompts-for-freeswitch*
> <https://www.gofundme.com/allison-prompts-for-freeswitch>
>
> Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
> /r/freeswitch <https://www.reddit.com/r/freeswitch>
>
> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
> *Skype:*briankwest
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org
>



-- 
Hardik Patel
iNextrix Technologies Pvt Ltd
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170609/6b8ee691/attachment-0001.html>

From marcel.haldemann at convercom.ch  Fri Jun  9 10:30:51 2017
From: marcel.haldemann at convercom.ch (Marcel Haldemann)
Date: Fri, 9 Jun 2017 10:30:51 +0000
Subject: [Freeswitch-users] Removing userpart of contact
In-Reply-To: <CAFGRPVo2jMJc==9SpeOPZjhgbxARMAwgQfiquuYUd-UOR7UeHw@mail.gmail.com>
References: <CAFGRPVo2jMJc==9SpeOPZjhgbxARMAwgQfiquuYUd-UOR7UeHw@mail.gmail.com>
Message-ID: <HE1PR0602MB284222734F78B52EBF17E447EFCE0@HE1PR0602MB2842.eurprd06.prod.outlook.com>

U should be able to change it via:

sip_contact_uri

                             <action application="export" data="sip_contact_uri=1.2.3.4:5060<http://1.2.3.4:5060>" />

Not sure wheter u need to add sip: or not.

From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of David Villasmil
Sent: Friday, June 9, 2017 7:48 AM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: [Freeswitch-users] Removing userpart of contact

Hello guys,

I ran into a situation where I need the contact to be like:

<sip:1.2.3.4:5060<http://1.2.3.4:5060>>

Meaning I need to remove the username, i've trying doing this but FS adds the user as "mod_sofia"... is it possible to do this?


Thanks and Regards,

David Villasmil
email: david.villasmil.work at gmail.com<mailto:david.villasmil.work at gmail.com>
phone: +34669448337
[https://mailfoogae.appspot.com/t?sender=aZGF2aWQudmlsbGFzbWlsLndvcmtAZ21haWwuY29t&type=zerocontent&guid=4dfa2470-b916-44df-b5c7-3822b6d98d25]ᐧ
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170609/973f2a88/attachment.html>

From asilva at wirelessmundi.com  Fri Jun  9 14:53:57 2017
From: asilva at wirelessmundi.com (Antonio Silva)
Date: Fri, 9 Jun 2017 16:53:57 +0200
Subject: [Freeswitch-users] Open Source SIP\WebRTC clients compatible
 with FreeSWITCH
In-Reply-To: <BN6PR07MB355689F93B56219553B97C31BEC80@BN6PR07MB3556.namprd07.prod.outlook.com>
References: <6FD2F8B5BB72834E9939AEDF9FB802A901E860B047@mbx-01.sysconfig.co.uk>
 <10c301d236a7$1ea0f8f0$5be2ead0$@freeswitch.org>
 <CALR0W69hJEpLqwgnq2FE0hBi6E3BHOOWEE=TANcjFQ8SEMRBoA@mail.gmail.com>
 <10cd01d236a8$425136b0$c6f3a410$@freeswitch.org>
 <CAPNTnnr9FjeR4HGPzkUaU_mG6b326gZ_-RJk4Z=PX1C5XSco-g@mail.gmail.com>
 <BN6PR07MB355689F93B56219553B97C31BEC80@BN6PR07MB3556.namprd07.prod.outlook.com>
Message-ID: <6311b75e-abb2-ad9f-8629-e7d564a379e0@wirelessmundi.com>


meanwhile if build app from scratch we can use this:

https://github.com/ISBX/apprtc-ios


Saludos / Regards / Cumprimentos,
António silva

On 06/07/2017 11:50 PM, Mundkowsky, Robert wrote:
>
> Cool, not if they just support getUserMedia!
>
> Robert Mundkowsky
>
> *From:*FreeSWITCH-users 
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of 
> *Guillermo Ruiz Camauer
> *Sent:* Wednesday, June 7, 2017 4:18 PM
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Subject:* Re: [Freeswitch-users] Open Source SIP\WebRTC clients 
> compatible with FreeSWITCH
>
> Apple announces WebRTC support in iOS11 / Safari: 
> https://apple.slashdot.org/story/17/06/07/1958242/apple-announces-support-for-webrtc-in-safari-11
>
> On Fri, Nov 4, 2016 at 11:32 AM, Ken Rice <krice at freeswitch.org 
> <mailto:krice at freeswitch.org>> wrote:
>
>     No one supports Native WebRTC on iOS at this time except for
>     people using their own private SDKs that they are not allowing to
>     get out there…
>
>     Apple does not have webRTC in webkit (Safari) or iOS at this time.
>     Chrome on iOS is not even really Chrome, its just a wrapper around
>     the WebKIT APIs and is effectively just safari with a few extra
>     functions and built to look like chrome.
>
>     *From:*freeswitch-users-bounces at lists.freeswitch.org
>     <mailto:freeswitch-users-bounces at lists.freeswitch.org>
>     [mailto:freeswitch-users-bounces at lists.freeswitch.org
>     <mailto:freeswitch-users-bounces at lists.freeswitch.org>] *On Behalf
>     Of *Chandramouli P
>     *Sent:* Friday, November 4, 2016 9:29 AM
>     *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org
>     <mailto:freeswitch-users at lists.freeswitch.org>>
>     *Subject:* Re: [Freeswitch-users] Open Source SIP\WebRTC clients
>     compatible with FreeSWITCH
>
>     Hello Ken,
>
>     We worked with Google native WebRTC on Firefox, Chrome, and Opera
>     browsers on Windows OS. I recently noticed that Google has added
>     the WebRTC support for Android, and iOS platforms (webrtc.org
>     <http://webrtc.org>). Now, we are planning to develop video
>     calling module using Google native WebRTC on these new platforms.
>     Can anybody give me the information about my below queries:
>
>     1) Does Google native WebRTC supports Apple iOS platform (native
>     mobile app)?
>
>     2) Does Google native WebRTC supports Apple OS X platform?
>
>     3) Is it possible to develop video calling module using native
>     WebRTC on Safari, and Chrome browsers on Apple OS X platform?
>
>     4) Does Google native WebRTC supports Android platform (native
>     mobile app)?
>
>     5) If it supports, I could not find any documentation for Apple
>     iOS, Apple OS X, and Android platforms specifically. Could you
>     please send some referral links?
>
>     6) I could not able to find the referral examples also for Apple
>     iOS, Apple OS X, and Android platforms specifically. Could you
>     please send some referral links?
>
>     Please do needful.
>
>     Thank you,
>
>     Chandramouli.
>
>     On Fri, Nov 4, 2016 at 7:54 PM, Ken Rice <krice at freeswitch.org
>     <mailto:krice at freeswitch.org>> wrote:
>
>         Verto Works on pretty much any platform that has native webrtc
>         support now... unfortunately things like iOS and don’t have
>         native iOS support yet…
>
>         If you are looking to build something you might contact
>         consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>         and see if you can work with the FSS Team to develop something
>
>         *From:* freeswitch-users-bounces at lists.freeswitch.org
>         <mailto:freeswitch-users-bounces at lists.freeswitch.org>
>         [mailto:freeswitch-users-bounces at lists.freeswitch.org
>         <mailto:freeswitch-users-bounces at lists.freeswitch.org>] *On
>         Behalf Of *Shaun Stokes
>         *Sent:* Friday, November 4, 2016 8:40 AM
>         *To:* 'FreeSWITCH Users Help'
>         <freeswitch-users at lists.freeswitch.org
>         <mailto:freeswitch-users at lists.freeswitch.org>>
>         *Subject:* [Freeswitch-users] Open Source SIP\WebRTC clients
>         compatible with FreeSWITCH
>
>         Hi All,
>
>         Does anyone have any recommendations on a good open source
>         SIP\WebRTC client which works on multiple platforms (Windows,
>         Mac, Linux, Mobiles) to provide presence, voice, video,
>         instant messaging, screen sharing and file sharing? This must
>         be capable of integrating with FreeSWITCH for voice and video
>         (presence via FreeSWITCH would be an advantage).
>
>         Many Thanks,
>
>         Shaun
>
>         	
>
>         Shaun Stokes - Infrastructure Analyst
>
>
>         	
>
>         T :
>
>         	
>
>         01453 700713
>
>         E :
>
>         	
>
>         shaun.stokes at itec-support.co.uk
>         <mailto:shaun.stokes at itec-support.co.uk>
>
>         W :
>
>         	
>
>         www.itec-support.co.uk <http://www.itec-support.co.uk>
>
>         Registered Address :- ITEC Support, Suite 2 Prospect House,
>         Bath Road, Stroud, Gloucestershire GL5 3QF
>         Company No. 06908001
>
>
>         CONFIDENTIALITY NOTICE
>         This communication and the information it contains are
>         intended for the person or organisation to which it is
>         addressed. Its contents are confidential and may be protected
>         in law. Unauthorised use, copying or disclosure of any of it
>         may be unlawful. If you are not the intended recipient, please
>         contact us immediately.
>         The contents of any attachments in this e-mail may contain
>         software viruses, which could damage your own computer system.
>         While ITEC Support has taken every reasonable precaution to
>         minimise this risk, we cannot accept liability for any damage
>         which you sustain as a result of software viruses. You should
>         carry out your own virus checking procedure before opening any
>         attachment.
>
>
>         ______________________________________________________________________
>         This message has been checked for all known viruses by
>         MessageLabs Virus Scanning Service.
>         ______________________________________________________________________
>
>
>         _________________________________________________________________________
>         Professional FreeSWITCH Consulting Services:
>         consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>         http://www.freeswitchsolutions.com
>
>         Official FreeSWITCH Sites
>         http://www.freeswitch.org
>         http://confluence.freeswitch.org
>         http://www.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
>         http://www.freeswitch.org
>
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://confluence.freeswitch.org
>     http://www.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
>     http://www.freeswitch.org
>
>
>
> -- 
>
> Guillermo Ruiz Camauer
>
>
> ------------------------------------------------------------------------
>
> This e-mail and any files transmitted with it may contain privileged 
> or confidential information. It is solely for use by the individual 
> for whom it is intended, even if addressed incorrectly. If you 
> received this e-mail in error, please notify the sender; do not 
> disclose, copy, distribute, or take any action in reliance on the 
> contents of this information; and delete it from your system. Any 
> other use of this e-mail is prohibited.
>
>
> Thank you for your compliance.
>
> ------------------------------------------------------------------------
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170609/642272c6/attachment-0001.html>

From eastour at 163.com  Fri Jun  9 07:02:58 2017
From: eastour at 163.com (chenyzhi)
Date: Fri, 9 Jun 2017 15:02:58 +0800 (CST)
Subject: [Freeswitch-users] When calling out , freeswitch doesn't get DTMF ?
In-Reply-To: <15c898b5d58.2779.b07ebdf329620b8089087c7205b03f01@xbipin.com>
References: <f823cdd8-a354-3698-d8c5-484b1b4cc8b4@xbipin.com>
 <CAEJMVkDDQX=Ni3AOQxQOo=joGoAeYroxioVP=cu9w6Y2m7kLPA@mail.gmail.com>
 <15c898b5d58.2779.b07ebdf329620b8089087c7205b03f01@xbipin.com>
Message-ID: <310cf71c.8fc3.15c8ba9c4f6.Coremail.eastour@163.com>

Hi


call 5000 from x-lite , you can hear the IVR voice and if you press dtmf keys , freeswitch can receive the dtmf keys .


but , if you enter the command  "originate user/1001 5000"  , x-lite will ring ,answer it ,you can hear the IVR voice , press some keys ,the freeswitch can NOT receive any dtmf ,


why?
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170609/466ebccd/attachment.html>

From manpower13.cse at gmail.com  Fri Jun  9 15:41:04 2017
From: manpower13.cse at gmail.com (Murugan Pandian)
Date: Fri, 9 Jun 2017 21:11:04 +0530
Subject: [Freeswitch-users] Open Source SIP\WebRTC clients compatible
	with FreeSWITCH
In-Reply-To: <6311b75e-abb2-ad9f-8629-e7d564a379e0@wirelessmundi.com>
References: <6FD2F8B5BB72834E9939AEDF9FB802A901E860B047@mbx-01.sysconfig.co.uk>
 <10c301d236a7$1ea0f8f0$5be2ead0$@freeswitch.org>
 <CALR0W69hJEpLqwgnq2FE0hBi6E3BHOOWEE=TANcjFQ8SEMRBoA@mail.gmail.com>
 <10cd01d236a8$425136b0$c6f3a410$@freeswitch.org>
 <CAPNTnnr9FjeR4HGPzkUaU_mG6b326gZ_-RJk4Z=PX1C5XSco-g@mail.gmail.com>
 <BN6PR07MB355689F93B56219553B97C31BEC80@BN6PR07MB3556.namprd07.prod.outlook.com>
 <6311b75e-abb2-ad9f-8629-e7d564a379e0@wirelessmundi.com>
Message-ID: <CAMRuXGg4WRMQo3K1a8xPxUPtLsuF=oCG4=MOx00macg5bvg8+A@mail.gmail.com>

HI ,

You can try this React-native JSSIP,It  support WebRTC i just tested this
,It's working fine with Android
https://github.com/telecmi/react-native-chub

On Fri, Jun 9, 2017 at 8:23 PM, Antonio Silva <asilva at wirelessmundi.com>
wrote:

>
> meanwhile if build app from scratch we can use this:
>
> https://github.com/ISBX/apprtc-ios
>
>
> Saludos / Regards / Cumprimentos,
> António silva
>
> On 06/07/2017 11:50 PM, Mundkowsky, Robert wrote:
>
> Cool, not if they just support getUserMedia!
>
>
>
> Robert Mundkowsky
>
>
>
> *From:* FreeSWITCH-users [mailto:freeswitch-users-
> bounces at lists.freeswitch.org
> <freeswitch-users-bounces at lists.freeswitch.org>] *On Behalf Of *Guillermo
> Ruiz Camauer
> *Sent:* Wednesday, June 7, 2017 4:18 PM
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> <freeswitch-users at lists.freeswitch.org>
> *Subject:* Re: [Freeswitch-users] Open Source SIP\WebRTC clients
> compatible with FreeSWITCH
>
>
>
> Apple announces WebRTC support in iOS11 / Safari:
> https://apple.slashdot.org/story/17/06/07/1958242/apple-
> announces-support-for-webrtc-in-safari-11
>
>
>
>
>
>
>
> On Fri, Nov 4, 2016 at 11:32 AM, Ken Rice <krice at freeswitch.org> wrote:
>
> No one supports Native WebRTC on iOS at this time except for people using
> their own private SDKs that they are not allowing to get out there…
>
>
>
> Apple does not have webRTC in webkit (Safari) or iOS at this time. Chrome
> on iOS is not even really Chrome, its just a wrapper around the WebKIT APIs
> and is effectively just safari with a few extra functions and built to look
> like chrome.
>
>
>
>
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Chandramouli
> P
> *Sent:* Friday, November 4, 2016 9:29 AM
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Subject:* Re: [Freeswitch-users] Open Source SIP\WebRTC clients
> compatible with FreeSWITCH
>
>
>
> Hello Ken,
>
>
>
> We worked with Google native WebRTC on Firefox, Chrome, and Opera browsers
> on Windows OS. I recently noticed that Google has added the WebRTC support
> for Android, and iOS platforms (webrtc.org). Now, we are planning to
> develop video calling module using Google native WebRTC on these new
> platforms. Can anybody give me the information about my below queries:
>
>
>
> 1) Does Google native WebRTC supports Apple iOS platform (native mobile
> app)?
>
> 2) Does Google native WebRTC supports Apple OS X platform?
>
> 3) Is it possible to develop video calling module using native WebRTC on
> Safari, and Chrome browsers on Apple OS X platform?
>
> 4) Does Google native WebRTC supports Android platform (native mobile app)?
>
> 5) If it supports, I could not find any documentation for Apple iOS, Apple
> OS X, and Android platforms specifically. Could you please send some
> referral links?
>
> 6) I could not able to find the referral examples also for Apple iOS,
> Apple OS X, and Android platforms specifically. Could you please send some
> referral links?
>
>
>
> Please do needful.
>
>
>
> Thank you,
>
> Chandramouli.
>
>
>
>
>
> On Fri, Nov 4, 2016 at 7:54 PM, Ken Rice <krice at freeswitch.org> wrote:
>
> Verto Works on pretty much any platform that has native webrtc support
> now... unfortunately things like iOS and don’t have native iOS support yet…
>
>
>
> If you are looking to build something you might contact
> consulting at freeswitch.org and see if you can work with the FSS Team to
> develop something
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Shaun Stokes
> *Sent:* Friday, November 4, 2016 8:40 AM
> *To:* 'FreeSWITCH Users Help' <freeswitch-users at lists.freeswitch.org>
> *Subject:* [Freeswitch-users] Open Source SIP\WebRTC clients compatible
> with FreeSWITCH
>
>
>
> Hi All,
>
>
>
> Does anyone have any recommendations on a good open source SIP\WebRTC
> client which works on multiple platforms (Windows, Mac, Linux, Mobiles) to
> provide presence, voice, video, instant messaging, screen sharing and file
> sharing? This must be capable of integrating with FreeSWITCH for voice and
> video (presence via FreeSWITCH would be an advantage).
>
>
>
> Many Thanks,
>
> Shaun
>
> Shaun Stokes - Infrastructure Analyst
>
>
> T :
>
> 01453 700713
>
> E :
>
> shaun.stokes at itec-support.co.uk
>
> W :
>
> www.itec-support.co.uk
>
> Registered Address :- ITEC Support, Suite 2 Prospect House, Bath Road,
> Stroud, Gloucestershire GL5 3QF
> Company No. 06908001
>
>
> CONFIDENTIALITY NOTICE
> This communication and the information it contains are intended for the
> person or organisation to which it is addressed. Its contents are
> confidential and may be protected in law. Unauthorised use, copying or
> disclosure of any of it may be unlawful. If you are not the intended
> recipient, please contact us immediately.
> The contents of any attachments in this e-mail may contain software
> viruses, which could damage your own computer system. While ITEC Support
> has taken every reasonable precaution to minimise this risk, we cannot
> accept liability for any damage which you sustain as a result of software
> viruses. You should carry out your own virus checking procedure before
> opening any attachment.
>
>
> ______________________________________________________________________
> This message has been checked for all known viruses by MessageLabs Virus
> Scanning Service.
> ______________________________________________________________________
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org
>
>
>
>
>
> --
>
> Guillermo Ruiz Camauer
>
> ------------------------------
>
> This e-mail and any files transmitted with it may contain privileged or
> confidential information. It is solely for use by the individual for whom
> it is intended, even if addressed incorrectly. If you received this e-mail
> in error, please notify the sender; do not disclose, copy, distribute, or
> take any action in reliance on the contents of this information; and delete
> it from your system. Any other use of this e-mail is prohibited.
>
> Thank you for your compliance.
> ------------------------------
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>
> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.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-usershttp://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170609/20912ae6/attachment-0001.html>

From dig1234 at gmail.com  Fri Jun  9 15:50:48 2017
From: dig1234 at gmail.com (Daniel Greenwald)
Date: Fri, 9 Jun 2017 11:50:48 -0400
Subject: [Freeswitch-users] RPORT still being sent in TCP calls
Message-ID: <CAOsHgtszqP2ph+LLadnHWmm6cMmU4r6=Og4smfydScoVkMh0wQ@mail.gmail.com>

We have noticed that FS is sending RPORT in TCP calls to a gateway. It was
reported as fixed in this bug:
https://freeswitch.org/jira/browse/FS-6612

We are running:
1.5.15b+git~20150512T053645Z~9eb887af47~64bit

I am not sure why RPORT is still being sent. Is this there a config
parameter which needs to be set to suppress the RPORT? Or was this change
reverted in later versions for some reason. Provider is telling us we
should not be sending RPORT in TCP...

Any info would be greatly appreciated.

Thanks!
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170609/fda4f839/attachment.html>

From krice at freeswitch.org  Fri Jun  9 16:06:55 2017
From: krice at freeswitch.org (Ken Rice)
Date: Fri, 9 Jun 2017 11:06:55 -0500
Subject: [Freeswitch-users] Open Source SIP\WebRTC clients
	compatible	with FreeSWITCH
In-Reply-To: <CAPNTnnr9FjeR4HGPzkUaU_mG6b326gZ_-RJk4Z=PX1C5XSco-g@mail.gmail.com>
References: <6FD2F8B5BB72834E9939AEDF9FB802A901E860B047@mbx-01.sysconfig.co.uk>
 <10c301d236a7$1ea0f8f0$5be2ead0$@freeswitch.org>
 <CALR0W69hJEpLqwgnq2FE0hBi6E3BHOOWEE=TANcjFQ8SEMRBoA@mail.gmail.com>
 <10cd01d236a8$425136b0$c6f3a410$@freeswitch.org>
 <CAPNTnnr9FjeR4HGPzkUaU_mG6b326gZ_-RJk4Z=PX1C5XSco-g@mail.gmail.com>
Message-ID: <4fc801d2e13a$6b052de0$410f89a0$@freeswitch.org>

Yes they just announced this and we’re quite excited about it. I’m personally waiting to see what this means not just for Safari but also for iOS apps themselves.

 

Rest assured the FreeSWITCH team will be testing with Safari.

 

From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Guillermo Ruiz Camauer
Sent: Wednesday, June 7, 2017 3:18 PM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Open Source SIP\WebRTC clients compatible with FreeSWITCH

 

Apple announces WebRTC support in iOS11 / Safari:  https://apple.slashdot.org/story/17/06/07/1958242/apple-announces-support-for-webrtc-in-safari-11

 

 

 

On Fri, Nov 4, 2016 at 11:32 AM, Ken Rice <krice at freeswitch.org <mailto:krice at freeswitch.org> > wrote:

No one supports Native WebRTC on iOS at this time except for people using their own private SDKs that they are not allowing to get out there…

 

Apple does not have webRTC in webkit (Safari) or iOS at this time. Chrome on iOS is not even really Chrome, its just a wrapper around the WebKIT APIs and is effectively just safari with a few extra functions and built to look like chrome.

 

 

 

From: freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org>  [mailto:freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org> ] On Behalf Of Chandramouli P
Sent: Friday, November 4, 2016 9:29 AM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org> >
Subject: Re: [Freeswitch-users] Open Source SIP\WebRTC clients compatible with FreeSWITCH

 

Hello Ken,

 

We worked with Google native WebRTC on Firefox, Chrome, and Opera browsers on Windows OS. I recently noticed that Google has added the WebRTC support for Android, and iOS platforms (webrtc.org <http://webrtc.org> ). Now, we are planning to develop video calling module using Google native WebRTC on these new platforms. Can anybody give me the information about my below queries:

 

1) Does Google native WebRTC supports Apple iOS platform (native mobile app)?

2) Does Google native WebRTC supports Apple OS X platform?

3) Is it possible to develop video calling module using native WebRTC on Safari, and Chrome browsers on Apple OS X platform?

4) Does Google native WebRTC supports Android platform (native mobile app)?

5) If it supports, I could not find any documentation for Apple iOS, Apple OS X, and Android platforms specifically. Could you please send some referral links?

6) I could not able to find the referral examples also for Apple iOS, Apple OS X, and Android platforms specifically. Could you please send some referral links?

 

Please do needful. 

 

Thank you,

Chandramouli.

 

 

On Fri, Nov 4, 2016 at 7:54 PM, Ken Rice <krice at freeswitch.org <mailto:krice at freeswitch.org> > wrote:

Verto Works on pretty much any platform that has native webrtc support now... unfortunately things like iOS and don’t have native iOS support yet…

 

If you are looking to build something you might contact consulting at freeswitch.org <mailto:consulting at freeswitch.org>  and see if you can work with the FSS Team to develop something

 

From: freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org>  [mailto:freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org> ] On Behalf Of Shaun Stokes
Sent: Friday, November 4, 2016 8:40 AM
To: 'FreeSWITCH Users Help' <freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org> >
Subject: [Freeswitch-users] Open Source SIP\WebRTC clients compatible with FreeSWITCH

 

Hi All,

 

Does anyone have any recommendations on a good open source SIP\WebRTC client which works on multiple platforms (Windows, Mac, Linux, Mobiles) to provide presence, voice, video, instant messaging, screen sharing and file sharing? This must be capable of integrating with FreeSWITCH for voice and video (presence via FreeSWITCH would be an advantage).

 

Many Thanks,

Shaun


  <http://www.itec-support.co.uk/wp-content/uploads/2016/07/email_logo.jpg> 

Shaun Stokes - Infrastructure Analyst

		

T : 

01453 700713


E : 

shaun.stokes at itec-support.co.uk <mailto:shaun.stokes at itec-support.co.uk> 


W : 

www.itec-support.co.uk <http://www.itec-support.co.uk> 

Registered Address :- ITEC Support, Suite 2 Prospect House, Bath Road, Stroud, Gloucestershire GL5 3QF 
Company No. 06908001 


CONFIDENTIALITY NOTICE 
This communication and the information it contains are intended for the person or organisation to which it is addressed. Its contents are confidential and may be protected in law. Unauthorised use, copying or disclosure of any of it may be unlawful. If you are not the intended recipient, please contact us immediately. 
The contents of any attachments in this e-mail may contain software viruses, which could damage your own computer system. While ITEC Support has taken every reasonable precaution to minimise this risk, we cannot accept liability for any damage which you sustain as a result of software viruses. You should carry out your own virus checking procedure before opening any attachment. 


______________________________________________________________________
This message has been checked for all known viruses by MessageLabs Virus Scanning Service.
______________________________________________________________________


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org <mailto:consulting at freeswitch.org> 
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.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
http://www.freeswitch.org

 


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org <mailto:consulting at freeswitch.org> 
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.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
http://www.freeswitch.org





 

-- 

Guillermo Ruiz Camauer

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170609/5c589e6e/attachment-0001.html>

From richardm at cellularity.co.uk  Fri Jun  9 17:23:44 2017
From: richardm at cellularity.co.uk (Richard Melville)
Date: Fri, 9 Jun 2017 18:23:44 +0100
Subject: [Freeswitch-users] Freeswitch Dependencies
Message-ID: <CAPeGcy8GXEM0jnezTF5BEBxcHAYi1D_5CbcYAufovcys2jK2wg@mail.gmail.com>

On 8 June 2017 at 17:01, Michael Jerris <mike at jerris.com> wrote:

>
> On Jun 8, 2017, at 6:33 AM, Richard Melville <richardm at cellularity.co.uk>
> wrote:
>
> I haven't used or built Freeswitch yet, but I'm getting closer to that
> point.  I will be building from source.  Having looked at the documentation
> I can see that Debian is preferred, and the use of distros in general.
> However, I don't use distros, but rather build my own systems from scratch.
>
> I've looked at the Centos page ( https://freeswitch.org/conflue
> nce/display/FREESWITCH/CentOS+7+and+RHEL+7 ) and under "Building from
> source" there appears to be a list of dependencies.  I already have most of
> those dependencies installed (other than the codecs) but I have three
> questions which maybe somebody can answer.
>
> The first is: can openssl be substituted with libressl, which I use?
>
>
> Not sure, depends if libressl has the required pieces we need for
> dtls-srtp and all the required ciphers required by the browsers for webrtc.
>

Thanks, I suppose the answer is to give it a try.

>
>
> The second is: "mongo-c-driver-devel" suggests that mongodb is a
> dependency of Freeswitch.  I've seen no mention of mongodb anywhere in
> either the book, or the documentation generally, so why is this listed as a
> dependency?
>
>
> Its a module, if you don’t want that module, its not needed.
>

That's what I thought; thanks for the heads up.

>
>
> The third is: there is no mention of package version numbers anywhere, so
> how can I find if there are any issues with particular versions?
>
>
> We don’t test a vast array of different package versions, we do testing
> based on the ones for the distros we package for.  The versions in Debian 8
> are well tested, other versions are much less well tested or not tested at
> all.  As for other libs, use the ones in our stash project for dep libs
> when not otherwise available.  Creating extensive documentation for
> building on your own distro would be far more work than even adding support
> for a new distro, and we don’t have any plans to create that.  I’m happy to
> respond to some specific questions, but there are limits to the amount of
> time that it makes sense for us to spend on issues like this for a single
> person.
>

Again, I suppose the answer is to just try the build with my versions (most
of which are fairly recent -- maybe too recent, but I'll see) and if there
are any problems with a particular package then refer back to Debian 8.

>
>
> Any help gratefully received.
>
> Thanks for your help, Michael.

-- 
Richard Melville
Systems Architect
cellularity.co.uk
stellarsystem.wordpress.com
+44 20 33 555 305
+44 7957 836330
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170609/cccaa1ff/attachment.html>

From mike at jerris.com  Fri Jun  9 18:01:54 2017
From: mike at jerris.com (Michael Jerris)
Date: Fri, 9 Jun 2017 14:01:54 -0400
Subject: [Freeswitch-users] Freeswitch Dependencies
In-Reply-To: <CAPeGcy8GXEM0jnezTF5BEBxcHAYi1D_5CbcYAufovcys2jK2wg@mail.gmail.com>
References: <CAPeGcy8GXEM0jnezTF5BEBxcHAYi1D_5CbcYAufovcys2jK2wg@mail.gmail.com>
Message-ID: <7A1E6C8D-009F-45B2-8F02-D3946EC64DA9@jerris.com>


> On Jun 9, 2017, at 1:23 PM, Richard Melville <richardm at cellularity.co.uk> wrote:
> 
> On 8 June 2017 at 17:01, Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>> wrote:
> 
>> On Jun 8, 2017, at 6:33 AM, Richard Melville <richardm at cellularity.co.uk <mailto:richardm at cellularity.co.uk>> wrote:
>> 
>> I haven't used or built Freeswitch yet, but I'm getting closer to that point.  I will be building from source.  Having looked at the documentation I can see that Debian is preferred, and the use of distros in general.  However, I don't use distros, but rather build my own systems from scratch.
>> 
>> I've looked at the Centos page ( https://freeswitch.org/confluence/display/FREESWITCH/CentOS+7+and+RHEL+7 <https://freeswitch.org/confluence/display/FREESWITCH/CentOS+7+and+RHEL+7> ) and under "Building from source" there appears to be a list of dependencies.  I already have most of those dependencies installed (other than the codecs) but I have three questions which maybe somebody can answer.
>> 
>> The first is: can openssl be substituted with libressl, which I use?
> 
> Not sure, depends if libressl has the required pieces we need for dtls-srtp and all the required ciphers required by the browsers for webrtc.
> 
> Thanks, I suppose the answer is to give it a try. 
> 
>> 
>> The second is: "mongo-c-driver-devel" suggests that mongodb is a dependency of Freeswitch.  I've seen no mention of mongodb anywhere in either the book, or the documentation generally, so why is this listed as a dependency?
> 
> Its a module, if you don’t want that module, its not needed.
> 
> That's what I thought; thanks for the heads up. 
> 
>> 
>> The third is: there is no mention of package version numbers anywhere, so how can I find if there are any issues with particular versions?
> 
> We don’t test a vast array of different package versions, we do testing based on the ones for the distros we package for.  The versions in Debian 8 are well tested, other versions are much less well tested or not tested at all.  As for other libs, use the ones in our stash project for dep libs when not otherwise available.  Creating extensive documentation for building on your own distro would be far more work than even adding support for a new distro, and we don’t have any plans to create that.  I’m happy to respond to some specific questions, but there are limits to the amount of time that it makes sense for us to spend on issues like this for a single person.
> 
> Again, I suppose the answer is to just try the build with my versions (most of which are fairly recent -- maybe too recent, but I'll see) and if there are any problems with a particular package then refer back to Debian 8.  

No problem. The easiest way to see the module dep chain would be to look at this file:

https://freeswitch.org/stash/projects/FS/repos/freeswitch/browse/debian/control-modules <https://freeswitch.org/stash/projects/FS/repos/freeswitch/browse/debian/control-modules>

> 
>> 
>> Any help gratefully received.
>> 
> 
> Thanks for your help, Michael.
> 
> -- 
> Richard Melville
> Systems Architect
> cellularity.co.uk <http://cellularity.co.uk/>
> stellarsystem.wordpress.com <http://stellarsystem.wordpress.com/>
> +44 20 33 555 305
> +44 7957 836330
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170609/d29d4baf/attachment.html>

From richardm at cellularity.co.uk  Fri Jun  9 18:22:31 2017
From: richardm at cellularity.co.uk (Richard Melville)
Date: Fri, 9 Jun 2017 19:22:31 +0100
Subject: [Freeswitch-users] Freeswitch Dependencies
Message-ID: <CAPeGcy_5WQiBUooDp-5c3SiDapaLdMY8rQ91Jzbg15J-8cezRA@mail.gmail.com>

On 8 June 2017 at 22:34, Ken Rice <krice at freeswitch.org> wrote:

>
> And as usual I got interrupted and clicked send before finishing a
> thought… (sorry about that but it helps demonstrate our time constraints)
>
>
>
> But following up my earlier comments
>
>
>
>
>
> Now that being said you can review the freeswitch.spec file in the root of
> the source tree or the Debian packaging files in the debian dir also in the
> source tree to get a list of build deps and if min versions are required
> what those are.
>
>
>
> Things without version numbering in those files are that way due to the
> current versions in the appropriate Distros being sufficient and Distros
> typically do not change APIs once theyhave released a  version (or on minor
> versions upgrades)_
>
> Thanks Ken.  In reply to this, and your earlier email, that's really
useful info.  I'm not looking for one-off support, but just to be pointed
in roughly the right direction.  Regarding LibreSSL I guess that it's just
a case of trying it out.

Thanks again.
-- 
Richard Melville
Systems Architect
cellularity.co.uk
stellarsystem.wordpress.com
+44 20 33 555 305
+44 7957 836330
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170609/cb2a6ea8/attachment-0001.html>

From richardm at cellularity.co.uk  Fri Jun  9 18:27:59 2017
From: richardm at cellularity.co.uk (Richard Melville)
Date: Fri, 9 Jun 2017 19:27:59 +0100
Subject: [Freeswitch-users] Freeswitch Dependencies
Message-ID: <CAPeGcy8jet4gqn7ebYMRhMAuLS1tQwSv5ep7o2PNxqydvn0t-A@mail.gmail.com>

On 9 June 2017 at 19:01, Michael Jerris <mike at jerris.com> wrote:

>
> On Jun 9, 2017, at 1:23 PM, Richard Melville <richardm at cellularity.co.uk>
> wrote:
>
> On 8 June 2017 at 17:01, Michael Jerris <mike at jerris.com> wrote:
>
>>
>> On Jun 8, 2017, at 6:33 AM, Richard Melville <richardm at cellularity.co.uk>
>> wrote:
>>
>> I haven't used or built Freeswitch yet, but I'm getting closer to that
>> point.  I will be building from source.  Having looked at the documentation
>> I can see that Debian is preferred, and the use of distros in general.
>> However, I don't use distros, but rather build my own systems from scratch.
>>
>> I've looked at the Centos page ( https://freeswitch.org/conflue
>> nce/display/FREESWITCH/CentOS+7+and+RHEL+7 ) and under "Building from
>> source" there appears to be a list of dependencies.  I already have most of
>> those dependencies installed (other than the codecs) but I have three
>> questions which maybe somebody can answer.
>>
>> The first is: can openssl be substituted with libressl, which I use?
>>
>>
>> Not sure, depends if libressl has the required pieces we need for
>> dtls-srtp and all the required ciphers required by the browsers for webrtc.
>>
>
> Thanks, I suppose the answer is to give it a try.
>
>>
>>
>> The second is: "mongo-c-driver-devel" suggests that mongodb is a
>> dependency of Freeswitch.  I've seen no mention of mongodb anywhere in
>> either the book, or the documentation generally, so why is this listed as a
>> dependency?
>>
>>
>> Its a module, if you don’t want that module, its not needed.
>>
>
> That's what I thought; thanks for the heads up.
>
>>
>>
>> The third is: there is no mention of package version numbers anywhere, so
>> how can I find if there are any issues with particular versions?
>>
>>
>> We don’t test a vast array of different package versions, we do testing
>> based on the ones for the distros we package for.  The versions in Debian 8
>> are well tested, other versions are much less well tested or not tested at
>> all.  As for other libs, use the ones in our stash project for dep libs
>> when not otherwise available.  Creating extensive documentation for
>> building on your own distro would be far more work than even adding support
>> for a new distro, and we don’t have any plans to create that.  I’m happy to
>> respond to some specific questions, but there are limits to the amount of
>> time that it makes sense for us to spend on issues like this for a single
>> person.
>>
>
> Again, I suppose the answer is to just try the build with my versions
> (most of which are fairly recent -- maybe too recent, but I'll see) and if
> there are any problems with a particular package then refer back to Debian
> 8.
>
>
> No problem. The easiest way to see the module dep chain would be to look
> at this file:
>
> https://freeswitch.org/stash/projects/FS/repos/freeswitch/
> browse/debian/control-modules
>
> Excellent, thanks for that.

-- 
Richard Melville
Systems Architect
cellularity.co.uk
stellarsystem.wordpress.com
+44 20 33 555 305
+44 7957 836330
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170609/0a10055b/attachment.html>

From anthony.minessale at gmail.com  Fri Jun  9 18:56:47 2017
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 9 Jun 2017 13:56:47 -0500
Subject: [Freeswitch-users] RPORT still being sent in TCP calls
In-Reply-To: <CAOsHgtszqP2ph+LLadnHWmm6cMmU4r6=Og4smfydScoVkMh0wQ@mail.gmail.com>
References: <CAOsHgtszqP2ph+LLadnHWmm6cMmU4r6=Og4smfydScoVkMh0wQ@mail.gmail.com>
Message-ID: <CAKbxfG8n7ycf_vgkFE73=mDX_anW2z284onh89vdSB2mZO=ykw@mail.gmail.com>

Odd number releases are not stable releases they are dev releases.
The version you quoted is from may 2015 2 years ago.

The latest stable release is 1.6.17 and latest build release changes every
day.



On Fri, Jun 9, 2017 at 10:50 AM, Daniel Greenwald <dig1234 at gmail.com> wrote:

> We have noticed that FS is sending RPORT in TCP calls to a gateway. It was
> reported as fixed in this bug:
> https://freeswitch.org/jira/browse/FS-6612
>
> We are running:
> 1.5.15b+git~20150512T053645Z~9eb887af47~64bit
>
> I am not sure why RPORT is still being sent. Is this there a config
> parameter which needs to be set to suppress the RPORT? Or was this change
> reverted in later versions for some reason. Provider is telling us we
> should not be sending RPORT in TCP...
>
> Any info would be greatly appreciated.
>
> Thanks!
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org
>



-- 
Anthony Minessale II       ♬ @anthmfs  ♬ @FreeSWITCH  ♬

☞ http://freeswitch.org/  ☞ http://cluecon.com/  ☞
http://twitter.com/FreeSWITCH
☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+
<http://freeswitch.org/g+>*

ClueCon Weekly Development Call
☎ sip:888 at conference.freeswitch.org  ☎ +19193869900

https://www.youtube.com/watch?v=9XXgW34t40s
https://www.youtube.com/watch?v=NLaDpGQuZDA
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170609/90533ac6/attachment.html>

From tculjaga at gmail.com  Fri Jun  9 20:42:24 2017
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Fri, 9 Jun 2017 22:42:24 +0200
Subject: [Freeswitch-users] Fax receive issue with t30 codec
In-Reply-To: <CAEJMVkDtY8VeG+14cB42wHee5fjgzsXE5trA31dtp=F4rxoeRg@mail.gmail.com>
References: <CAHdn4vjxr5vVoO7Oc1P0u-xg6dJNYBhjxU=y5iz6VP5_JEcUjA@mail.gmail.com>
 <CAEJMVkDtY8VeG+14cB42wHee5fjgzsXE5trA31dtp=F4rxoeRg@mail.gmail.com>
Message-ID: <CAJoKKdVBBBbZSmtT20f6ANgDJnpXjV+7a1==ivsdQ5qcaN3BqA@mail.gmail.com>

hi, whats new on faxing in 1.6.17  ?

T.

On 6 June 2017 at 16:00, Brian West <brian at freeswitch.org> wrote:

> You'll have to use 1.6.17 if you ever want any faxing to work in all test
> cases.
>
> /b
>
> On Tue, Jun 6, 2017 at 6:11 AM, Hardik Patel <hardikitpl at gmail.com> wrote:
>
>> Hello,
>>
>> I am using opensips as entry point using dispatcher. opensips(
>> 127.0.0.1), i am routing call to freeswitch server (127.0.0.3).
>>
>> Now I am trying to receive fax, my issue is when i try to send fax in
>> softphone(Zoiper) from the log i am seeing that it is sending fax using t30
>> codec. and i am not receiving the fax at destination, is it because of
>> codec, should it only work with t38 codec? if that is the issue than how am
>> i be able to send the fax using t38 from zoiper?
>>
>> Here i am attaching the fs log with loglevel 9 and sip trace is also
>> enabled.
>>
>> 127.0.0.2  => carrier/provider IP
>> 123456789 => Fax number
>> test at gamil.com => Email Address
>> 127.0.0.4 =>UI IP
>>
>> Pastebin link:https://pastebin.freeswitch.org/view/9ec52715
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.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
>> http://www.freeswitch.org
>>
>
>
>
> --
>
> *Brian West*
> brian at freeswitch.org
>
> *Twitter: @FreeSWITCH , @briankwest*
>
> http://www.freeswitchbook.com
> http://www.freeswitchcookbook.com
>
> Book a phone call (CST) <https://freeswitch.com/appointment>
>
> Allison prompts for FreeSWITCH:
>
> *https://www.gofundme.com/allison-prompts-for-freeswitch*
> <https://www.gofundme.com/allison-prompts-for-freeswitch>
>
> Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
> /r/freeswitch <https://www.reddit.com/r/freeswitch>
>
> *T:*+19184209001 <+1%20918-420-9001> | *F:*+19184209002
> <+1%20918-420-9002> | *M:*+1918424WEST (9378)
> *Skype:*briankwest
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170609/1a2591f0/attachment-0001.html>

From brian at freeswitch.org  Fri Jun  9 21:12:51 2017
From: brian at freeswitch.org (Brian West)
Date: Fri, 9 Jun 2017 16:12:51 -0500
Subject: [Freeswitch-users] Fax receive issue with t30 codec
In-Reply-To: <CAJoKKdVBBBbZSmtT20f6ANgDJnpXjV+7a1==ivsdQ5qcaN3BqA@mail.gmail.com>
References: <CAHdn4vjxr5vVoO7Oc1P0u-xg6dJNYBhjxU=y5iz6VP5_JEcUjA@mail.gmail.com>
 <CAEJMVkDtY8VeG+14cB42wHee5fjgzsXE5trA31dtp=F4rxoeRg@mail.gmail.com>
 <CAJoKKdVBBBbZSmtT20f6ANgDJnpXjV+7a1==ivsdQ5qcaN3BqA@mail.gmail.com>
Message-ID: <CAEJMVkBFD6b=8porYC9Vm-si5Qy3OsDCNKQrmOBDWmm4A-qTEA@mail.gmail.com>

Your log snip doesn't really help, I know without a single doubt faxing
works fine.  So what are you doing and how are you doing it?

On Fri, Jun 9, 2017 at 3:42 PM, Tihomir Culjaga <tculjaga at gmail.com> wrote:

> hi, whats new on faxing in 1.6.17  ?
>
> T.
>
> On 6 June 2017 at 16:00, Brian West <brian at freeswitch.org> wrote:
>
>> You'll have to use 1.6.17 if you ever want any faxing to work in all test
>> cases.
>>
>> /b
>>
>> On Tue, Jun 6, 2017 at 6:11 AM, Hardik Patel <hardikitpl at gmail.com>
>> wrote:
>>
>>> Hello,
>>>
>>> I am using opensips as entry point using dispatcher. opensips(
>>> 127.0.0.1), i am routing call to freeswitch server (127.0.0.3).
>>>
>>> Now I am trying to receive fax, my issue is when i try to send fax in
>>> softphone(Zoiper) from the log i am seeing that it is sending fax using t30
>>> codec. and i am not receiving the fax at destination, is it because of
>>> codec, should it only work with t38 codec? if that is the issue than how am
>>> i be able to send the fax using t38 from zoiper?
>>>
>>> Here i am attaching the fs log with loglevel 9 and sip trace is also
>>> enabled.
>>>
>>> 127.0.0.2  => carrier/provider IP
>>> 123456789 => Fax number
>>> test at gamil.com => Email Address
>>> 127.0.0.4 =>UI IP
>>>
>>> Pastebin link:https://pastebin.freeswitch.org/view/9ec52715
>>>
>>> ____________________________________________________________
>>> _____________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.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
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>>
>> *Brian West*
>> brian at freeswitch.org
>>
>> *Twitter: @FreeSWITCH , @briankwest*
>>
>> http://www.freeswitchbook.com
>> http://www.freeswitchcookbook.com
>>
>> Book a phone call (CST) <https://freeswitch.com/appointment>
>>
>> Allison prompts for FreeSWITCH:
>>
>> *https://www.gofundme.com/allison-prompts-for-freeswitch*
>> <https://www.gofundme.com/allison-prompts-for-freeswitch>
>>
>> Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
>> /r/freeswitch <https://www.reddit.com/r/freeswitch>
>>
>> *T:*+19184209001 <+1%20918-420-9001> | *F:*+19184209002
>> <+1%20918-420-9002> | *M:*+1918424WEST (9378)
>> *Skype:*briankwest
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.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
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org
>



-- 

*Brian West*
brian at freeswitch.org

*Twitter: @FreeSWITCH , @briankwest*

http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

Book a phone call (CST) <https://freeswitch.com/appointment>

Allison prompts for FreeSWITCH:

*https://www.gofundme.com/allison-prompts-for-freeswitch*
<https://www.gofundme.com/allison-prompts-for-freeswitch>

Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
/r/freeswitch <https://www.reddit.com/r/freeswitch>

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*Skype:*briankwest
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170609/885dcd91/attachment.html>

From chad at apartmentlines.com  Fri Jun  9 22:18:38 2017
From: chad at apartmentlines.com (Chad Phillips)
Date: Fri, 9 Jun 2017 15:18:38 -0700
Subject: [Freeswitch-users] Open Source SIP\WebRTC clients compatible
	with FreeSWITCH
In-Reply-To: <4fc801d2e13a$6b052de0$410f89a0$@freeswitch.org>
References: <6FD2F8B5BB72834E9939AEDF9FB802A901E860B047@mbx-01.sysconfig.co.uk>
 <10c301d236a7$1ea0f8f0$5be2ead0$@freeswitch.org>
 <CALR0W69hJEpLqwgnq2FE0hBi6E3BHOOWEE=TANcjFQ8SEMRBoA@mail.gmail.com>
 <10cd01d236a8$425136b0$c6f3a410$@freeswitch.org>
 <CAPNTnnr9FjeR4HGPzkUaU_mG6b326gZ_-RJk4Z=PX1C5XSco-g@mail.gmail.com>
 <4fc801d2e13a$6b052de0$410f89a0$@freeswitch.org>
Message-ID: <CAJ0sghTLL6xjdbx9ALLN3J_Oty=jrXxEyvCgyKMP6DB7s_2AFw@mail.gmail.com>

Looks like the Janus devs got Safari working pretty easily:
https://twitter.com/elminiero/status/873178028683255808



On Fri, Jun 9, 2017 at 9:06 AM, Ken Rice <krice at freeswitch.org> wrote:

> Yes they just announced this and we’re quite excited about it. I’m
> personally waiting to see what this means not just for Safari but also for
> iOS apps themselves.
>
>
>
> Rest assured the FreeSWITCH team will be testing with Safari.
>
>
>
> *From:* FreeSWITCH-users [mailto:freeswitch-users-
> bounces at lists.freeswitch.org] *On Behalf Of *Guillermo Ruiz Camauer
> *Sent:* Wednesday, June 7, 2017 3:18 PM
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Subject:* Re: [Freeswitch-users] Open Source SIP\WebRTC clients
> compatible with FreeSWITCH
>
>
>
> Apple announces WebRTC support in iOS11 / Safari:
> https://apple.slashdot.org/story/17/06/07/1958242/apple-
> announces-support-for-webrtc-in-safari-11
>
>
>
>
>
>
>
> On Fri, Nov 4, 2016 at 11:32 AM, Ken Rice <krice at freeswitch.org> wrote:
>
> No one supports Native WebRTC on iOS at this time except for people using
> their own private SDKs that they are not allowing to get out there…
>
>
>
> Apple does not have webRTC in webkit (Safari) or iOS at this time. Chrome
> on iOS is not even really Chrome, its just a wrapper around the WebKIT APIs
> and is effectively just safari with a few extra functions and built to look
> like chrome.
>
>
>
>
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Chandramouli
> P
> *Sent:* Friday, November 4, 2016 9:29 AM
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Subject:* Re: [Freeswitch-users] Open Source SIP\WebRTC clients
> compatible with FreeSWITCH
>
>
>
> Hello Ken,
>
>
>
> We worked with Google native WebRTC on Firefox, Chrome, and Opera browsers
> on Windows OS. I recently noticed that Google has added the WebRTC support
> for Android, and iOS platforms (webrtc.org). Now, we are planning to
> develop video calling module using Google native WebRTC on these new
> platforms. Can anybody give me the information about my below queries:
>
>
>
> 1) Does Google native WebRTC supports Apple iOS platform (native mobile
> app)?
>
> 2) Does Google native WebRTC supports Apple OS X platform?
>
> 3) Is it possible to develop video calling module using native WebRTC on
> Safari, and Chrome browsers on Apple OS X platform?
>
> 4) Does Google native WebRTC supports Android platform (native mobile app)?
>
> 5) If it supports, I could not find any documentation for Apple iOS, Apple
> OS X, and Android platforms specifically. Could you please send some
> referral links?
>
> 6) I could not able to find the referral examples also for Apple iOS,
> Apple OS X, and Android platforms specifically. Could you please send some
> referral links?
>
>
>
> Please do needful.
>
>
>
> Thank you,
>
> Chandramouli.
>
>
>
>
>
> On Fri, Nov 4, 2016 at 7:54 PM, Ken Rice <krice at freeswitch.org> wrote:
>
> Verto Works on pretty much any platform that has native webrtc support
> now... unfortunately things like iOS and don’t have native iOS support yet…
>
>
>
> If you are looking to build something you might contact
> consulting at freeswitch.org and see if you can work with the FSS Team to
> develop something
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Shaun Stokes
> *Sent:* Friday, November 4, 2016 8:40 AM
> *To:* 'FreeSWITCH Users Help' <freeswitch-users at lists.freeswitch.org>
> *Subject:* [Freeswitch-users] Open Source SIP\WebRTC clients compatible
> with FreeSWITCH
>
>
>
> Hi All,
>
>
>
> Does anyone have any recommendations on a good open source SIP\WebRTC
> client which works on multiple platforms (Windows, Mac, Linux, Mobiles) to
> provide presence, voice, video, instant messaging, screen sharing and file
> sharing? This must be capable of integrating with FreeSWITCH for voice and
> video (presence via FreeSWITCH would be an advantage).
>
>
>
> Many Thanks,
>
> Shaun
>
> Shaun Stokes - Infrastructure Analyst
>
> T :
>
> 01453 700713
>
> E :
>
> shaun.stokes at itec-support.co.uk
>
> W :
>
> www.itec-support.co.uk
>
> Registered Address :- ITEC Support, Suite 2 Prospect House, Bath Road,
> Stroud, Gloucestershire GL5 3QF
> Company No. 06908001
>
>
> CONFIDENTIALITY NOTICE
> This communication and the information it contains are intended for the
> person or organisation to which it is addressed. Its contents are
> confidential and may be protected in law. Unauthorised use, copying or
> disclosure of any of it may be unlawful. If you are not the intended
> recipient, please contact us immediately.
> The contents of any attachments in this e-mail may contain software
> viruses, which could damage your own computer system. While ITEC Support
> has taken every reasonable precaution to minimise this risk, we cannot
> accept liability for any damage which you sustain as a result of software
> viruses. You should carry out your own virus checking procedure before
> opening any attachment.
>
>
> ______________________________________________________________________
> This message has been checked for all known viruses by MessageLabs Virus
> Scanning Service.
> ______________________________________________________________________
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org
>
>
>
>
>
> --
>
> Guillermo Ruiz Camauer
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170609/edd5d906/attachment-0001.html>

From david.villasmil at gmail.com  Fri Jun  9 22:48:18 2017
From: david.villasmil at gmail.com (David Villasmil Govea)
Date: Fri, 09 Jun 2017 22:48:18 +0000
Subject: [Freeswitch-users] Removing userpart of contact
In-Reply-To: <HE1PR0602MB284222734F78B52EBF17E447EFCE0@HE1PR0602MB2842.eurprd06.prod.outlook.com>
References: <CAFGRPVo2jMJc==9SpeOPZjhgbxARMAwgQfiquuYUd-UOR7UeHw@mail.gmail.com>
 <HE1PR0602MB284222734F78B52EBF17E447EFCE0@HE1PR0602MB2842.eurprd06.prod.outlook.com>
Message-ID: <CAPTi0gqiSHHB3eg-HC76n-CQsdxb0+32HWHX5TpUHwO0e4F==g@mail.gmail.com>

Hello,

thanks for your help, but i had already tried that, and it doesn't do what
i need, the contact still is like "mod_sofia at 1.2.3.4"

I was looking at the source, and it seems to me, if the contact begins with
"@" which i assume is added somewhere i couldn't find, then fs adds the
"mod_sofia"...

Any thoughts?

Regards,

David


On Fri, Jun 9, 2017 at 6:31 AM Marcel Haldemann <
marcel.haldemann at convercom.ch> wrote:

> U should be able to change it via:
>
>
>
> sip_contact_uri
>
>
>
>                              <action application="export" data="
> sip_contact_uri=1.2.3.4:5060" />
>
>
>
> Not sure wheter u need to add sip: or not.
>
>
>
> *From:* FreeSWITCH-users [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *David
> Villasmil
> *Sent:* Friday, June 9, 2017 7:48 AM
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Subject:* [Freeswitch-users] Removing userpart of contact
>
>
>
> Hello guys,
>
>
>
> I ran into a situation where I need the contact to be like:
>
> <sip:1.2.3.4:5060>
>
>
>
> Meaning I need to remove the username, i've trying doing this but FS adds
> the user as "mod_sofia"... is it possible to do this?
>
>
>
>
>
> Thanks and Regards,
>
>
>
> David Villasmil
>
> email: david.villasmil.work at gmail.com
>
> phone: +34669448337
>
> ᐧ
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170609/17466b56/attachment.html>

From david.villasmil.work at gmail.com  Fri Jun  9 22:49:34 2017
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Fri, 09 Jun 2017 22:49:34 +0000
Subject: [Freeswitch-users] When calling out ,
	freeswitch doesn't get DTMF ?
In-Reply-To: <310cf71c.8fc3.15c8ba9c4f6.Coremail.eastour@163.com>
References: <f823cdd8-a354-3698-d8c5-484b1b4cc8b4@xbipin.com>
 <CAEJMVkDDQX=Ni3AOQxQOo=joGoAeYroxioVP=cu9w6Y2m7kLPA@mail.gmail.com>
 <15c898b5d58.2779.b07ebdf329620b8089087c7205b03f01@xbipin.com>
 <310cf71c.8fc3.15c8ba9c4f6.Coremail.eastour@163.com>
Message-ID: <CAFGRPVqFA+tpaOQkAGGCR7s5gY0z0TbdArWFdkfnsaaG4Z9Dgw@mail.gmail.com>

Have you looked at the log? Bump the logging up and see what shows up...
what you're seeing is very weird

David
On Fri, Jun 9, 2017 at 11:18 AM chenyzhi <eastour at 163.com> wrote:

> Hi
>
> call 5000 from x-lite , you can hear the IVR voice and if you press dtmf
> keys , freeswitch can receive the dtmf keys .
>
> but , if you enter the command  "originate user/1001 5000"  , x-lite will
> ring ,answer it ,you can hear the IVR voice , press some keys ,the
> freeswitch can NOT receive any dtmf ,
>
> why?
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170609/92820881/attachment.html>

From hardikitpl at gmail.com  Sat Jun 10 04:16:51 2017
From: hardikitpl at gmail.com (Hardik Patel)
Date: Sat, 10 Jun 2017 09:46:51 +0530
Subject: [Freeswitch-users] Fax receive issue with t30 codec
In-Reply-To: <CAEJMVkBFD6b=8porYC9Vm-si5Qy3OsDCNKQrmOBDWmm4A-qTEA@mail.gmail.com>
References: <CAHdn4vjxr5vVoO7Oc1P0u-xg6dJNYBhjxU=y5iz6VP5_JEcUjA@mail.gmail.com>
 <CAEJMVkDtY8VeG+14cB42wHee5fjgzsXE5trA31dtp=F4rxoeRg@mail.gmail.com>
 <CAJoKKdVBBBbZSmtT20f6ANgDJnpXjV+7a1==ivsdQ5qcaN3BqA@mail.gmail.com>
 <CAEJMVkBFD6b=8porYC9Vm-si5Qy3OsDCNKQrmOBDWmm4A-qTEA@mail.gmail.com>
Message-ID: <CAHdn4vgRHZ5zNFLQYhDkUfrMGDNzqnEwQdQ3coORHfqQmOMOQA@mail.gmail.com>

Hi Brian,

Thanks for the support.

We are testing receive fax functionality using real fax machine and here i
have listed the model which we are using to send fax.

the models of real fax machines that we have used are group 3 CCITT / ITU,
they are the following:
1 ) konica minolta bizhub-c220
2 ) HP Officejet 4500
3 ) HP Officejet G85


>From above list one of our machine is sending fax without T38 support and
we got the failure with same error which have posted on bug but if we use
T38 support then it works fine for us.



*ERROR :*
2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:508 Fax processing not
successful - result (3) Timed out waiting for the first message.
2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:513 Remote station id:
2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:514 Local station id:
SpanDSP Fax Ident
2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:515 Pages transferred:
0
2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:517 Total fax pages:
0
2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:518 Image resolution:
0x0
2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:519 Transfer Rate:
14400
2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:521 ECM status
off
2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:522 remote country:
2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:523 remote vendor:
2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:524 remote model:
2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:526
==============================

On Sat, Jun 10, 2017 at 2:42 AM, Brian West <brian at freeswitch.org> wrote:

> Your log snip doesn't really help, I know without a single doubt faxing
> works fine.  So what are you doing and how are you doing it?
>
> On Fri, Jun 9, 2017 at 3:42 PM, Tihomir Culjaga <tculjaga at gmail.com>
> wrote:
>
>> hi, whats new on faxing in 1.6.17  ?
>>
>> T.
>>
>> On 6 June 2017 at 16:00, Brian West <brian at freeswitch.org> wrote:
>>
>>> You'll have to use 1.6.17 if you ever want any faxing to work in all
>>> test cases.
>>>
>>> /b
>>>
>>> On Tue, Jun 6, 2017 at 6:11 AM, Hardik Patel <hardikitpl at gmail.com>
>>> wrote:
>>>
>>>> Hello,
>>>>
>>>> I am using opensips as entry point using dispatcher. opensips(
>>>> 127.0.0.1), i am routing call to freeswitch server (127.0.0.3).
>>>>
>>>> Now I am trying to receive fax, my issue is when i try to send fax in
>>>> softphone(Zoiper) from the log i am seeing that it is sending fax using t30
>>>> codec. and i am not receiving the fax at destination, is it because of
>>>> codec, should it only work with t38 codec? if that is the issue than how am
>>>> i be able to send the fax using t38 from zoiper?
>>>>
>>>> Here i am attaching the fs log with loglevel 9 and sip trace is also
>>>> enabled.
>>>>
>>>> 127.0.0.2  => carrier/provider IP
>>>> 123456789 => Fax number
>>>> test at gamil.com => Email Address
>>>> 127.0.0.4 =>UI IP
>>>>
>>>> Pastebin link:https://pastebin.freeswitch.org/view/9ec52715
>>>>
>>>> ____________________________________________________________
>>>> _____________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.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/free
>>>> switch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> --
>>>
>>> *Brian West*
>>> brian at freeswitch.org
>>>
>>> *Twitter: @FreeSWITCH , @briankwest*
>>>
>>> http://www.freeswitchbook.com
>>> http://www.freeswitchcookbook.com
>>>
>>> Book a phone call (CST) <https://freeswitch.com/appointment>
>>>
>>> Allison prompts for FreeSWITCH:
>>>
>>> *https://www.gofundme.com/allison-prompts-for-freeswitch*
>>> <https://www.gofundme.com/allison-prompts-for-freeswitch>
>>>
>>> Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
>>> /r/freeswitch <https://www.reddit.com/r/freeswitch>
>>>
>>> *T:*+19184209001 <+1%20918-420-9001> | *F:*+19184209002
>>> <+1%20918-420-9002> | *M:*+1918424WEST (9378)
>>> *Skype:*briankwest
>>>
>>> ____________________________________________________________
>>> _____________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.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
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.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
>> http://www.freeswitch.org
>>
>
>
>
> --
>
> *Brian West*
> brian at freeswitch.org
>
> *Twitter: @FreeSWITCH , @briankwest*
>
> http://www.freeswitchbook.com
> http://www.freeswitchcookbook.com
>
> Book a phone call (CST) <https://freeswitch.com/appointment>
>
> Allison prompts for FreeSWITCH:
>
> *https://www.gofundme.com/allison-prompts-for-freeswitch*
> <https://www.gofundme.com/allison-prompts-for-freeswitch>
>
> Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
> /r/freeswitch <https://www.reddit.com/r/freeswitch>
>
> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
> *Skype:*briankwest
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org
>



-- 
Hardik Patel
iNextrix Technologies Pvt Ltd
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170610/34f2e65a/attachment-0001.html>

From john at industromatic.com  Fri Jun  9 23:33:26 2017
From: john at industromatic.com (John Griessen)
Date: Fri, 9 Jun 2017 18:33:26 -0500
Subject: [Freeswitch-users] latency and performance
Message-ID: <44eb01b6-b9d6-fad6-fd40-43447167b9aa@industromatic.com>

Say your office is in Austin.  Can you run a freeswitch server in a Dallas datacenter with a ping time of 15ms and get useful 
performance using a voip phone in Austin?  I've read that running in a VM ruins/randomizes some of the timing the kernel needs
to keep voice packets coming well...  Is there a way to measure if that is happening too much?

-- 
John Griessen


From caleb at bclife.biz  Sat Jun 10 04:14:07 2017
From: caleb at bclife.biz (Caleb Bartholomew)
Date: Fri, 9 Jun 2017 22:14:07 -0600
Subject: [Freeswitch-users] Multichannel Transcdription
Message-ID: <6E3BD1FE-212C-4005-9A4B-8C19DE18758C@bclife.biz>

Hi all,

I am currently trying to find a solution inside of Freeswitch that will allow me to either record each participant of a conference room separately so it can later be processed for transcription.  I’ve looked into a lot of different methods that might be able to do this but I’m not sure any of them are quite what I need.   I stumbled across mod_vlc which does hint at this ability but I’m not sure what exactly a “raw” conference stream is or its format.  Any help is greatly appreciated.

Thanks,
Caleb
-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 842 bytes
Desc: Message signed with OpenPGP
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170609/fee9fa67/attachment.sig>

From rbetancor at gmail.com  Sat Jun 10 10:30:43 2017
From: rbetancor at gmail.com (=?UTF-8?Q?Ra=C3=BAl_Alexis_Betancor_Santana?=)
Date: Sat, 10 Jun 2017 11:30:43 +0100
Subject: [Freeswitch-users] latency and performance
In-Reply-To: <44eb01b6-b9d6-fad6-fd40-43447167b9aa@industromatic.com>
References: <44eb01b6-b9d6-fad6-fd40-43447167b9aa@industromatic.com>
Message-ID: <CAKsfQBhNyxi-_D+7+mCNsoUUwX+J1RbOWNeXYBwE5SvZy=PUXg@mail.gmail.com>

You could have your vpbx on Kracovia and your phones on Patagonia, 250ms of
RTT and still run your ip voice service smooth.

2017-06-10 0:33 GMT+01:00 John Griessen <john at industromatic.com>:

> Say your office is in Austin.  Can you run a freeswitch server in a Dallas
> datacenter with a ping time of 15ms and get useful performance using a voip
> phone in Austin?  I've read that running in a VM ruins/randomizes some of
> the timing the kernel needs
> to keep voice packets coming well...  Is there a way to measure if that is
> happening too much?
>
> --
> John Griessen
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170610/daeceb19/attachment.html>

From david.villasmil.work at gmail.com  Sat Jun 10 20:40:44 2017
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Sat, 10 Jun 2017 20:40:44 +0000
Subject: [Freeswitch-users] latency and performance
In-Reply-To: <CAKsfQBhNyxi-_D+7+mCNsoUUwX+J1RbOWNeXYBwE5SvZy=PUXg@mail.gmail.com>
References: <44eb01b6-b9d6-fad6-fd40-43447167b9aa@industromatic.com>
 <CAKsfQBhNyxi-_D+7+mCNsoUUwX+J1RbOWNeXYBwE5SvZy=PUXg@mail.gmail.com>
Message-ID: <CAFGRPVreXZgtWPdE08FW5vA3duXwwH+j9q6GfzkqueXyLPtT-Q@mail.gmail.com>

Nothing like testing it yourself. It should be ok as long as you don't put
heavy load on it.
On Sat, Jun 10, 2017 at 6:31 AM Raúl Alexis Betancor Santana <
rbetancor at gmail.com> wrote:

> You could have your vpbx on Kracovia and your phones on Patagonia, 250ms
> of RTT and still run your ip voice service smooth.
>
> 2017-06-10 0:33 GMT+01:00 John Griessen <john at industromatic.com>:
>
>> Say your office is in Austin.  Can you run a freeswitch server in a
>> Dallas datacenter with a ping time of 15ms and get useful performance using
>> a voip phone in Austin?  I've read that running in a VM ruins/randomizes
>> some of the timing the kernel needs
>> to keep voice packets coming well...  Is there a way to measure if that
>> is happening too much?
>>
>> --
>> John Griessen
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.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
>> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170610/8beb56cb/attachment.html>

From david.villasmil.work at gmail.com  Sun Jun 11 01:39:04 2017
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Sun, 11 Jun 2017 01:39:04 +0000
Subject: [Freeswitch-users] Removing userpart of contact
In-Reply-To: <CAPTi0gqiSHHB3eg-HC76n-CQsdxb0+32HWHX5TpUHwO0e4F==g@mail.gmail.com>
References: <CAFGRPVo2jMJc==9SpeOPZjhgbxARMAwgQfiquuYUd-UOR7UeHw@mail.gmail.com>
 <HE1PR0602MB284222734F78B52EBF17E447EFCE0@HE1PR0602MB2842.eurprd06.prod.outlook.com>
 <CAPTi0gqiSHHB3eg-HC76n-CQsdxb0+32HWHX5TpUHwO0e4F==g@mail.gmail.com>
Message-ID: <CAFGRPVqweQCv1ng1zQntNhb3P=ezDqNwbQJut5E-sRSCBZYrRg@mail.gmail.com>

Any thoughts on this, guys?
On Fri, Jun 9, 2017 at 6:49 PM David Villasmil Govea <
david.villasmil at gmail.com> wrote:

> Hello,
>
> thanks for your help, but i had already tried that, and it doesn't do what
> i need, the contact still is like "mod_sofia at 1.2.3.4"
>
> I was looking at the source, and it seems to me, if the contact begins
> with "@" which i assume is added somewhere i couldn't find, then fs adds
> the "mod_sofia"...
>
> Any thoughts?
>
> Regards,
>
> David
>
>
> On Fri, Jun 9, 2017 at 6:31 AM Marcel Haldemann <
> marcel.haldemann at convercom.ch> wrote:
>
>> U should be able to change it via:
>>
>>
>>
>> sip_contact_uri
>>
>>
>>
>>                              <action application="export" data="
>> sip_contact_uri=1.2.3.4:5060" />
>>
>>
>>
>> Not sure wheter u need to add sip: or not.
>>
>>
>>
>> *From:* FreeSWITCH-users [mailto:
>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *David
>> Villasmil
>> *Sent:* Friday, June 9, 2017 7:48 AM
>> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> *Subject:* [Freeswitch-users] Removing userpart of contact
>>
>>
>>
>> Hello guys,
>>
>>
>>
>> I ran into a situation where I need the contact to be like:
>>
>> <sip:1.2.3.4:5060>
>>
>>
>>
>> Meaning I need to remove the username, i've trying doing this but FS adds
>> the user as "mod_sofia"... is it possible to do this?
>>
>>
>>
>>
>>
>> Thanks and Regards,
>>
>>
>>
>> David Villasmil
>>
>> email: david.villasmil.work at gmail.com
>>
>> phone: +34669448337
>>
>> ᐧ
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.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
>> http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170611/f62eb262/attachment.html>

From ksrigo at gmail.com  Sun Jun 11 10:21:03 2017
From: ksrigo at gmail.com (Srigo Kana)
Date: Sun, 11 Jun 2017 12:21:03 +0200
Subject: [Freeswitch-users] Removing userpart of contact
In-Reply-To: <CAFGRPVqweQCv1ng1zQntNhb3P=ezDqNwbQJut5E-sRSCBZYrRg@mail.gmail.com>
References: <CAFGRPVo2jMJc==9SpeOPZjhgbxARMAwgQfiquuYUd-UOR7UeHw@mail.gmail.com>
 <HE1PR0602MB284222734F78B52EBF17E447EFCE0@HE1PR0602MB2842.eurprd06.prod.outlook.com>
 <CAPTi0gqiSHHB3eg-HC76n-CQsdxb0+32HWHX5TpUHwO0e4F==g@mail.gmail.com>
 <CAFGRPVqweQCv1ng1zQntNhb3P=ezDqNwbQJut5E-sRSCBZYrRg@mail.gmail.com>
Message-ID: <8DF8A1B6-5E7F-412E-9CB4-70F80578A2EC@gmail.com>

Hi,

Cld u post ur dialplan where you are doing your bridge?

Srigo

Sent from my iPhone

> On 11 Jun 2017, at 03:39, David Villasmil <david.villasmil.work at gmail.com> wrote:
> 
> Any thoughts on this, guys?
>> On Fri, Jun 9, 2017 at 6:49 PM David Villasmil Govea <david.villasmil at gmail.com> wrote:
>> Hello,
>> 
>> thanks for your help, but i had already tried that, and it doesn't do what i need, the contact still is like "mod_sofia at 1.2.3.4"
>> 
>> I was looking at the source, and it seems to me, if the contact begins with "@" which i assume is added somewhere i couldn't find, then fs adds the "mod_sofia"...
>> 
>> Any thoughts?
>> 
>> Regards,
>> 
>> David
>> 
>> 
>>> On Fri, Jun 9, 2017 at 6:31 AM Marcel Haldemann <marcel.haldemann at convercom.ch> wrote:
>>> U should be able to change it via:
>>> 
>>>  
>>> 
>>> sip_contact_uri
>>> 
>>>  
>>> 
>>>                              <action application="export" data="sip_contact_uri=1.2.3.4:5060" />
>>> 
>>>  
>>> 
>>> Not sure wheter u need to add sip: or not.
>>> 
>>>  
>>> 
>>> From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of David Villasmil
>>> Sent: Friday, June 9, 2017 7:48 AM
>>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>> Subject: [Freeswitch-users] Removing userpart of contact
>>> 
>>>  
>>> 
>>> Hello guys,
>>> 
>>>  
>>> 
>>> I ran into a situation where I need the contact to be like:
>>> 
>>> <sip:1.2.3.4:5060>
>>> 
>>>  
>>> 
>>> Meaning I need to remove the username, i've trying doing this but FS adds the user as "mod_sofia"... is it possible to do this?
>>> 
>>>  
>>> 
>>>  
>>> 
>>> Thanks and Regards,
>>> 
>>>  
>>> 
>>> David Villasmil
>>> 
>>> email: david.villasmil.work at gmail.com
>>> 
>>> phone: +34669448337
>>> 
>>> ᐧ
>>> 
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.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
>>> http://www.freeswitch.org
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.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
>> http://www.freeswitch.org
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170611/a33f7daa/attachment-0001.html>

From ssinyagin at gmail.com  Sun Jun 11 14:14:42 2017
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Sun, 11 Jun 2017 16:14:42 +0200
Subject: [Freeswitch-users] Collection of notification sounds
Message-ID: <CACOYK=iTRN2psQan7=YJnd010Gbi+Qd5SSw5F1LtDHs==bwBYw@mail.gmail.com>

here's a new collection of notification sounds for your telephony projects:
https://github.com/voxserv/dzwin

enjoy. There are files already converted for various bitrates. You can
download the whole lot from "Releases" tab.


From gmaruzz at gmail.com  Sun Jun 11 14:54:32 2017
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Sun, 11 Jun 2017 16:54:32 +0200
Subject: [Freeswitch-users] Collection of notification sounds
In-Reply-To: <CACOYK=iTRN2psQan7=YJnd010Gbi+Qd5SSw5F1LtDHs==bwBYw@mail.gmail.com>
References: <CACOYK=iTRN2psQan7=YJnd010Gbi+Qd5SSw5F1LtDHs==bwBYw@mail.gmail.com>
Message-ID: <CALXCt0pwfMN+As8Tx3JVSM38fDLf+XhPQDNnw3Wy7RvcAWC7Aw@mail.gmail.com>

Very nice, thanks!



On 11 June 2017 at 16:14, Stanislav Sinyagin <ssinyagin at gmail.com> wrote:

> here's a new collection of notification sounds for your telephony projects:
> https://github.com/voxserv/dzwin
>
> enjoy. There are files already converted for various bitrates. You can
> download the whole lot from "Releases" tab.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org




-- 

Sincerely,

Giovanni Maruzzelli
OpenTelecom.IT
cell: +39 347 266 56 18
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170611/c33f8b13/attachment.html>

From david.villasmil at gmail.com  Sun Jun 11 16:08:10 2017
From: david.villasmil at gmail.com (David Villasmil Govea)
Date: Sun, 11 Jun 2017 12:08:10 -0400
Subject: [Freeswitch-users] Removing userpart of contact
In-Reply-To: <8DF8A1B6-5E7F-412E-9CB4-70F80578A2EC@gmail.com>
References: <CAFGRPVo2jMJc==9SpeOPZjhgbxARMAwgQfiquuYUd-UOR7UeHw@mail.gmail.com>
 <HE1PR0602MB284222734F78B52EBF17E447EFCE0@HE1PR0602MB2842.eurprd06.prod.outlook.com>
 <CAPTi0gqiSHHB3eg-HC76n-CQsdxb0+32HWHX5TpUHwO0e4F==g@mail.gmail.com>
 <CAFGRPVqweQCv1ng1zQntNhb3P=ezDqNwbQJut5E-sRSCBZYrRg@mail.gmail.com>
 <8DF8A1B6-5E7F-412E-9CB4-70F80578A2EC@gmail.com>
Message-ID: <CAPTi0gqnxXRRhz4MSOQxKRMBsjMf3FzUHSv-hG5m71T718Q7Rw@mail.gmail.com>

hello, here's the dialplan:

        <action application="export"
data="sip-force-contact=$${local_ip_v4}:$${external_sip_port}"/>
        <action application="bridge"
data="sip_h_aleg-callid=${variable_sip_call_id},sip
nolocal:rtp_secure_media=true,nolocal:sdp_secure_savp_only=true,absolute_codec_string=^^:OPUS:PCMA:PCMU}sofia/external/${destination_number}@9.8.7.6:5060;transport=tcp"
/>

I've tried several ways, but it seems everytime I set the contact to be
"sip:1.2.3.4:5060", fs adds a "mod_sofia" as the user part...

Thanks for any help

David
ᐧ

2017-06-11 6:21 GMT-04:00 Srigo Kana <ksrigo at gmail.com>:

> Hi,
>
> Cld u post ur dialplan where you are doing your bridge?
>
> Srigo
>
> Sent from my iPhone
>
> On 11 Jun 2017, at 03:39, David Villasmil <david.villasmil.work at gmail.com>
> wrote:
>
> Any thoughts on this, guys?
> On Fri, Jun 9, 2017 at 6:49 PM David Villasmil Govea <
> david.villasmil at gmail.com> wrote:
>
>> Hello,
>>
>> thanks for your help, but i had already tried that, and it doesn't do
>> what i need, the contact still is like "mod_sofia at 1.2.3.4"
>>
>> I was looking at the source, and it seems to me, if the contact begins
>> with "@" which i assume is added somewhere i couldn't find, then fs adds
>> the "mod_sofia"...
>>
>> Any thoughts?
>>
>> Regards,
>>
>> David
>>
>>
>> On Fri, Jun 9, 2017 at 6:31 AM Marcel Haldemann <
>> marcel.haldemann at convercom.ch> wrote:
>>
>>> U should be able to change it via:
>>>
>>>
>>>
>>> sip_contact_uri
>>>
>>>
>>>
>>>                              <action application="export" data="
>>> sip_contact_uri=1.2.3.4:5060" />
>>>
>>>
>>>
>>> Not sure wheter u need to add sip: or not.
>>>
>>>
>>>
>>> *From:* FreeSWITCH-users [mailto:freeswitch-users-
>>> bounces at lists.freeswitch.org] *On Behalf Of *David Villasmil
>>> *Sent:* Friday, June 9, 2017 7:48 AM
>>> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>> *Subject:* [Freeswitch-users] Removing userpart of contact
>>>
>>>
>>>
>>> Hello guys,
>>>
>>>
>>>
>>> I ran into a situation where I need the contact to be like:
>>>
>>> <sip:1.2.3.4:5060>
>>>
>>>
>>>
>>> Meaning I need to remove the username, i've trying doing this but FS
>>> adds the user as "mod_sofia"... is it possible to do this?
>>>
>>>
>>>
>>>
>>>
>>> Thanks and Regards,
>>>
>>>
>>>
>>> David Villasmil
>>>
>>> email: david.villasmil.work at gmail.com
>>>
>>> phone: +34669448337
>>>
>>> ᐧ
>>> ____________________________________________________________
>>> _____________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.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
>>> http://www.freeswitch.org
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.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
>> http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org
>



-- 
DVG

-- 
Imagination is more important than knowledge
Albert Einstein
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170611/a2e5e144/attachment-0001.html>

From david.villasmil.work at gmail.com  Sun Jun 11 16:43:18 2017
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Sun, 11 Jun 2017 16:43:18 +0000
Subject: [Freeswitch-users] Collection of notification sounds
In-Reply-To: <CALXCt0pwfMN+As8Tx3JVSM38fDLf+XhPQDNnw3Wy7RvcAWC7Aw@mail.gmail.com>
References: <CACOYK=iTRN2psQan7=YJnd010Gbi+Qd5SSw5F1LtDHs==bwBYw@mail.gmail.com>
 <CALXCt0pwfMN+As8Tx3JVSM38fDLf+XhPQDNnw3Wy7RvcAWC7Aw@mail.gmail.com>
Message-ID: <CAFGRPVo2n9vcKgJ9-eSabQX0DUanu9r5uMfUomUtpMtAWt_iew@mail.gmail.com>

Thanks!
On Sun, Jun 11, 2017 at 10:56 AM Giovanni Maruzzelli <gmaruzz at gmail.com>
wrote:

> Very nice, thanks!
>
>
>
> On 11 June 2017 at 16:14, Stanislav Sinyagin <ssinyagin at gmail.com> wrote:
>
>> here's a new collection of notification sounds for your telephony
>> projects:
>> https://github.com/voxserv/dzwin
>>
>> enjoy. There are files already converted for various bitrates. You can
>> download the whole lot from "Releases" tab.
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.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
>> http://www.freeswitch.org
>
>
>
>
> --
>
> Sincerely,
>
> Giovanni Maruzzelli
> OpenTelecom.IT
> cell: +39 347 266 56 18
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170611/1b34ccc3/attachment.html>

From eastour at 163.com  Mon Jun 12 00:32:53 2017
From: eastour at 163.com (chenyzhi)
Date: Mon, 12 Jun 2017 08:32:53 +0800 (CST)
Subject: [Freeswitch-users] When calling out ,
 freeswitch doesn't get DTMF ?
In-Reply-To: <CAFGRPVqFA+tpaOQkAGGCR7s5gY0z0TbdArWFdkfnsaaG4Z9Dgw@mail.gmail.com>
References: <f823cdd8-a354-3698-d8c5-484b1b4cc8b4@xbipin.com>
 <CAEJMVkDDQX=Ni3AOQxQOo=joGoAeYroxioVP=cu9w6Y2m7kLPA@mail.gmail.com>
 <15c898b5d58.2779.b07ebdf329620b8089087c7205b03f01@xbipin.com>
 <310cf71c.8fc3.15c8ba9c4f6.Coremail.eastour@163.com>
 <CAFGRPVqFA+tpaOQkAGGCR7s5gY0z0TbdArWFdkfnsaaG4Z9Dgw@mail.gmail.com>
Message-ID: <2a18422f.ef8.15c99b7b9bb.Coremail.eastour@163.com>

I have read the logs ,but I didn't find any difference.


Please make a test to see if this happens in your box.






At 2017-06-10 06:49:34, "David Villasmil" <david.villasmil.work at gmail.com> wrote:
Have you looked at the log? Bump the logging up and see what shows up... what you're seeing is very weird

David

On Fri, Jun 9, 2017 at 11:18 AM chenyzhi <eastour at 163.com> wrote:

Hi


call 5000 from x-lite , you can hear the IVR voice and if you press dtmf keys , freeswitch can receive the dtmf keys .


but , if you enter the command  "originate user/1001 5000"  , x-lite will ring ,answer it ,you can hear the IVR voice , press some keys ,the freeswitch can NOT receive any dtmf ,


why?




 

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.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
http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170612/dcf5facc/attachment.html>

From gmaruzz at gmail.com  Mon Jun 12 09:37:49 2017
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Mon, 12 Jun 2017 11:37:49 +0200
Subject: [Freeswitch-users] When calling out ,
	freeswitch doesn't get DTMF ?
In-Reply-To: <2a18422f.ef8.15c99b7b9bb.Coremail.eastour@163.com>
References: <f823cdd8-a354-3698-d8c5-484b1b4cc8b4@xbipin.com>
 <CAEJMVkDDQX=Ni3AOQxQOo=joGoAeYroxioVP=cu9w6Y2m7kLPA@mail.gmail.com>
 <15c898b5d58.2779.b07ebdf329620b8089087c7205b03f01@xbipin.com>
 <310cf71c.8fc3.15c8ba9c4f6.Coremail.eastour@163.com>
 <CAFGRPVqFA+tpaOQkAGGCR7s5gY0z0TbdArWFdkfnsaaG4Z9Dgw@mail.gmail.com>
 <2a18422f.ef8.15c99b7b9bb.Coremail.eastour@163.com>
Message-ID: <CALXCt0pKKEJoSeXXZ=tn04nifEZvQCRSFCDKL8jSu+pfOgvRNQ@mail.gmail.com>

you need to check the DTMF type, you probably are using the wrong one
(info-inband-rfc2833), and for some reason they are not negotiated

On 12 June 2017 at 02:32, chenyzhi <eastour at 163.com> wrote:

> I have read the logs ,but I didn't find any difference.
>
> Please make a test to see if this happens in your box.
>
>
>
>
>
> At 2017-06-10 06:49:34, "David Villasmil" <david.villasmil.work at gmail.com>
> wrote:
>
> Have you looked at the log? Bump the logging up and see what shows up...
> what you're seeing is very weird
>
> David
> On Fri, Jun 9, 2017 at 11:18 AM chenyzhi <eastour at 163.com> wrote:
>
>> Hi
>>
>> call 5000 from x-lite , you can hear the IVR voice and if you press dtmf
>> keys , freeswitch can receive the dtmf keys .
>>
>> but , if you enter the command  "originate user/1001 5000"  , x-lite will
>> ring ,answer it ,you can hear the IVR voice , press some keys ,the
>> freeswitch can NOT receive any dtmf ,
>>
>> why?
>>
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.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
>> http://www.freeswitch.org
>
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org
>



-- 

Sincerely,

Giovanni Maruzzelli
OpenTelecom.IT
cell: +39 347 266 56 18
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170612/d0d8ffce/attachment.html>

From vbvbrj at gmail.com  Mon Jun 12 12:22:24 2017
From: vbvbrj at gmail.com (Mimiko)
Date: Mon, 12 Jun 2017 15:22:24 +0300
Subject: [Freeswitch-users] callcenter member recall same agent.
Message-ID: <0f5df8c3-1e59-0a85-82b1-08757a33673c@gmail.com>

Hello.

I have a question.

Is there a parameter that a member, after talking to some agent, leaving callcenter and then call back will be connected specifically to same agent, 
but be in same queue? If the agent is no more linked to the queue, then member will be connected to any other agent like normally.
If agent is still in system and is talking - then member will wait this agent to be free.

-- 
Mimiko desu.


From eastour at 163.com  Mon Jun 12 13:16:08 2017
From: eastour at 163.com (chenyzhi)
Date: Mon, 12 Jun 2017 21:16:08 +0800 (CST)
Subject: [Freeswitch-users] When calling out ,
 freeswitch doesn't get DTMF ?
In-Reply-To: <CALXCt0pKKEJoSeXXZ=tn04nifEZvQCRSFCDKL8jSu+pfOgvRNQ@mail.gmail.com>
References: <f823cdd8-a354-3698-d8c5-484b1b4cc8b4@xbipin.com>
 <CAEJMVkDDQX=Ni3AOQxQOo=joGoAeYroxioVP=cu9w6Y2m7kLPA@mail.gmail.com>
 <15c898b5d58.2779.b07ebdf329620b8089087c7205b03f01@xbipin.com>
 <310cf71c.8fc3.15c8ba9c4f6.Coremail.eastour@163.com>
 <CAFGRPVqFA+tpaOQkAGGCR7s5gY0z0TbdArWFdkfnsaaG4Z9Dgw@mail.gmail.com>
 <2a18422f.ef8.15c99b7b9bb.Coremail.eastour@163.com>
 <CALXCt0pKKEJoSeXXZ=tn04nifEZvQCRSFCDKL8jSu+pfOgvRNQ@mail.gmail.com>
Message-ID: <6c11dd0f.c2c1.15c9c727dc7.Coremail.eastour@163.com>

I don't think It's  the DTMF type , because  when I dial 5000 from x-lite (which has registered to freeswitch as 1001) ,I can hear the voice and if I press any dtmf on x-lite,  freeswitch can recieve the dtmfs.  This means that the DTMF type is correct ,otherwise freeswitch coudn't have received the dtmfs;


 when I enter the command  "originate user/1001 5000"  at the freeswitch console , my xlite will ring ,and I answered ,I can hear the voice ,I press some dtmf,but freeswitch can NOT receive any dtmf.  really weird.






At 2017-06-12 17:37:49, "Giovanni Maruzzelli" <gmaruzz at gmail.com> wrote:

you need to check the DTMF type, you probably are using the wrong one (info-inband-rfc2833), and for some reason they are not negotiated



On 12 June 2017 at 02:32, chenyzhi <eastour at 163.com> wrote:

I have read the logs ,but I didn't find any difference.


Please make a test to see if this happens in your box.






At 2017-06-10 06:49:34, "David Villasmil" <david.villasmil.work at gmail.com> wrote:
Have you looked at the log? Bump the logging up and see what shows up... what you're seeing is very weird

David

On Fri, Jun 9, 2017 at 11:18 AM chenyzhi <eastour at 163.com> wrote:

Hi


call 5000 from x-lite , you can hear the IVR voice and if you press dtmf keys , freeswitch can receive the dtmf keys .


but , if you enter the command  "originate user/1001 5000"  , x-lite will ring ,answer it ,you can hear the IVR voice , press some keys ,the freeswitch can NOT receive any dtmf ,


why?




 

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.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
http://www.freeswitch.org




 


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.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
http://www.freeswitch.org




--


Sincerely,

Giovanni Maruzzelli
OpenTelecom.IT
cell: +39 347 266 56 18
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170612/a3656ba3/attachment.html>

From italo at freeswitch.org  Mon Jun 12 15:12:31 2017
From: italo at freeswitch.org (=?UTF-8?B?w410YWxvIFJvc3Np?=)
Date: Mon, 12 Jun 2017 12:12:31 -0300
Subject: [Freeswitch-users] callcenter member recall same agent.
In-Reply-To: <0f5df8c3-1e59-0a85-82b1-08757a33673c@gmail.com>
References: <0f5df8c3-1e59-0a85-82b1-08757a33673c@gmail.com>
Message-ID: <CABMX_w8-T5SYqvmXR=e6+oesdwjTScRymRNvU63HKu4YHwi8yg@mail.gmail.com>

There's no such feature yet...

On Mon, Jun 12, 2017 at 9:22 AM, Mimiko <vbvbrj at gmail.com> wrote:

> Hello.
>
> I have a question.
>
> Is there a parameter that a member, after talking to some agent, leaving
> callcenter and then call back will be connected specifically to same agent,
> but be in same queue? If the agent is no more linked to the queue, then
> member will be connected to any other agent like normally.
> If agent is still in system and is talking - then member will wait this
> agent to be free.
>
> --
> Mimiko desu.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org




-- 
Ítalo Rossi
italo at freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170612/6498a73d/attachment.html>

From luis.daniel.lucio at gmail.com  Mon Jun 12 14:57:17 2017
From: luis.daniel.lucio at gmail.com (Luis Daniel Lucio Quiroz)
Date: Mon, 12 Jun 2017 10:57:17 -0400
Subject: [Freeswitch-users] When calling out ,
	freeswitch doesn't get DTMF ?
In-Reply-To: <6c11dd0f.c2c1.15c9c727dc7.Coremail.eastour@163.com>
References: <f823cdd8-a354-3698-d8c5-484b1b4cc8b4@xbipin.com>
 <CAEJMVkDDQX=Ni3AOQxQOo=joGoAeYroxioVP=cu9w6Y2m7kLPA@mail.gmail.com>
 <15c898b5d58.2779.b07ebdf329620b8089087c7205b03f01@xbipin.com>
 <310cf71c.8fc3.15c8ba9c4f6.Coremail.eastour@163.com>
 <CAFGRPVqFA+tpaOQkAGGCR7s5gY0z0TbdArWFdkfnsaaG4Z9Dgw@mail.gmail.com>
 <2a18422f.ef8.15c99b7b9bb.Coremail.eastour@163.com>
 <CALXCt0pKKEJoSeXXZ=tn04nifEZvQCRSFCDKL8jSu+pfOgvRNQ@mail.gmail.com>
 <6c11dd0f.c2c1.15c9c727dc7.Coremail.eastour@163.com>
Message-ID: <CAFLo2QwwwE6xeXBN9DkM1VEecEJpwnDMmQ7iT9EvSyx9RqE9mg@mail.gmail.com>

check if you have transcoding, and if you do, check that dftm type-codec on
leg b are compatible.

--
Luis Daniel Lucio Quiroz
CISSP, CISM, CISA
Linux, VoIP and much more fun
www.okay.com.mx

Need LCR? Check out LCR for FusionPBX with FreeSWITCH
Need Billing? Check out Billing for FusionPBX with FreeSWITCH

On Mon, Jun 12, 2017 at 9:16 AM, chenyzhi <eastour at 163.com> wrote:

> I don't think It's  the DTMF type , because  when I dial 5000 from x-lite
> (which has registered to freeswitch as 1001) ,I can hear the voice and if I
> press any dtmf on x-lite,  freeswitch can recieve the dtmfs.  This means
> that the DTMF type is correct ,otherwise freeswitch coudn't have received
> the dtmfs;
>
>  when I enter the command  "originate user/1001 5000"  at the freeswitch
> console , my xlite will ring ,and I answered ,I can hear the voice ,I press
> some dtmf,but freeswitch can NOT receive any dtmf.  really weird.
>
>
>
>
>
> At 2017-06-12 17:37:49, "Giovanni Maruzzelli" <gmaruzz at gmail.com> wrote:
>
> you need to check the DTMF type, you probably are using the wrong one
> (info-inband-rfc2833), and for some reason they are not negotiated
>
> On 12 June 2017 at 02:32, chenyzhi <eastour at 163.com> wrote:
>
>> I have read the logs ,but I didn't find any difference.
>>
>> Please make a test to see if this happens in your box.
>>
>>
>>
>>
>>
>> At 2017-06-10 06:49:34, "David Villasmil" <david.villasmil.work at gmail.com>
>> wrote:
>>
>> Have you looked at the log? Bump the logging up and see what shows up...
>> what you're seeing is very weird
>>
>> David
>> On Fri, Jun 9, 2017 at 11:18 AM chenyzhi <eastour at 163.com> wrote:
>>
>>> Hi
>>>
>>> call 5000 from x-lite , you can hear the IVR voice and if you press dtmf
>>> keys , freeswitch can receive the dtmf keys .
>>>
>>> but , if you enter the command  "originate user/1001 5000"  , x-lite
>>> will ring ,answer it ,you can hear the IVR voice , press some keys ,the
>>> freeswitch can NOT receive any dtmf ,
>>>
>>> why?
>>>
>>>
>>>
>>>
>>> ____________________________________________________________
>>> _____________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.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
>>> http://www.freeswitch.org
>>
>>
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.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
>> http://www.freeswitch.org
>>
>
>
>
> --
>
> Sincerely,
>
> Giovanni Maruzzelli
> OpenTelecom.IT
> cell: +39 347 266 56 18
>
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170612/f526d9cc/attachment-0001.html>

From igorolhovskiy at gmail.com  Mon Jun 12 17:12:36 2017
From: igorolhovskiy at gmail.com (Igor Olhovskiy)
Date: Mon, 12 Jun 2017 20:12:36 +0300
Subject: [Freeswitch-users] User CIDR and ACL
Message-ID: <75fee20d-577d-41d5-bbdf-77628144691e@Spark>

Hi!

Can you pls help me with little misunderstanding.
According to https://freeswitch.org/confluence/display/FREESWITCH/ACL#ACL-Users
I can create ACL for a user and also specify a CIDR in <user> tag.
Like
<user id="1000" cidr="12.34.56.78/32,20.0.0.0/8»>
So, what is difference between cidr parameter and «auth-acl» parameter, and can I specify, that users can register only from specified network range?

Like I want my user 1001 register only from 10.0.20.0/24 and user 1002 register only from 10.0.30.0/24, but they must register with username and pass.

Regards, Igor
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170612/7e572254/attachment.html>

From dig1234 at gmail.com  Mon Jun 12 20:32:06 2017
From: dig1234 at gmail.com (Daniel Greenwald)
Date: Mon, 12 Jun 2017 16:32:06 -0400
Subject: [Freeswitch-users] RPORT still being sent in TCP calls
In-Reply-To: <CAKbxfG8n7ycf_vgkFE73=mDX_anW2z284onh89vdSB2mZO=ykw@mail.gmail.com>
References: <CAOsHgtszqP2ph+LLadnHWmm6cMmU4r6=Og4smfydScoVkMh0wQ@mail.gmail.com>
 <CAKbxfG8n7ycf_vgkFE73=mDX_anW2z284onh89vdSB2mZO=ykw@mail.gmail.com>
Message-ID: <CAOsHgtt0X+-iLDCqEpu4wc8obgjp1kBe_9zAoprJi+7DU2SZ1g@mail.gmail.com>

We have upgraded to latest STABLE from Ubuntu packages but we are still
seeing rport in TCP calls:



INVITE sip:xxxxxxx at sip.freeswitch.com;transport=tcp SIP/2.0

Via: SIP/2.0/TCP x.x.x.x;rport;branch=z9hG4bKZDKF5N8QcFZcH

Max-Forwards: 70

From: "user" <sip:xxxxxx at sip.freeswitch.com>;tag=Z1pQm54mB22De

To: <sip:xxxxxx at sip.freeswitch.com;transport=tcp>

Call-ID: c79656fc-d2d0-4446-90c3-060dabf82fd6

CSeq: 108312748 INVITE

Contact: <sip:user at x.x.x.x:5060;transport=tcp;gw=provider>

User-Agent: FreeSWITCH-mod_sofia/1.6.10-4-726448d~64bit

On Fri, Jun 9, 2017 at 2:56 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> Odd number releases are not stable releases they are dev releases.
> The version you quoted is from may 2015 2 years ago.
>
> The latest stable release is 1.6.17 and latest build release changes every
> day.
>
>
>
> On Fri, Jun 9, 2017 at 10:50 AM, Daniel Greenwald <dig1234 at gmail.com>
> wrote:
>
>> We have noticed that FS is sending RPORT in TCP calls to a gateway. It
>> was reported as fixed in this bug:
>> https://freeswitch.org/jira/browse/FS-6612
>>
>> We are running:
>> 1.5.15b+git~20150512T053645Z~9eb887af47~64bit
>>
>> I am not sure why RPORT is still being sent. Is this there a config
>> parameter which needs to be set to suppress the RPORT? Or was this change
>> reverted in later versions for some reason. Provider is telling us we
>> should not be sending RPORT in TCP...
>>
>> Any info would be greatly appreciated.
>>
>> Thanks!
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.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
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II       ♬ @anthmfs  ♬ @FreeSWITCH  ♬
>
> ☞ http://freeswitch.org/  ☞ http://cluecon.com/  ☞
> http://twitter.com/FreeSWITCH
> ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+
> <http://freeswitch.org/g+>*
>
> ClueCon Weekly Development Call
> ☎ sip:888 at conference.freeswitch.org  ☎ +19193869900 <(919)%20386-9900>
>
> https://www.youtube.com/watch?v=9XXgW34t40s
> https://www.youtube.com/watch?v=NLaDpGQuZDA
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170612/6cdbe3a1/attachment.html>

From mike at jerris.com  Mon Jun 12 20:42:42 2017
From: mike at jerris.com (Michael Jerris)
Date: Mon, 12 Jun 2017 16:42:42 -0400
Subject: [Freeswitch-users] RPORT still being sent in TCP calls
In-Reply-To: <CAOsHgtt0X+-iLDCqEpu4wc8obgjp1kBe_9zAoprJi+7DU2SZ1g@mail.gmail.com>
References: <CAOsHgtszqP2ph+LLadnHWmm6cMmU4r6=Og4smfydScoVkMh0wQ@mail.gmail.com>
 <CAKbxfG8n7ycf_vgkFE73=mDX_anW2z284onh89vdSB2mZO=ykw@mail.gmail.com>
 <CAOsHgtt0X+-iLDCqEpu4wc8obgjp1kBe_9zAoprJi+7DU2SZ1g@mail.gmail.com>
Message-ID: <8E832DB1-BEF7-4EF1-826F-ED57C966E794@jerris.com>

Change is still in there, confirmed.  The packages you are using are very old (but should also have that patch in it)… I see that you are supplying modified sip traces.  If you can reproduce this on master code, please create a Jira with configuration and full debug logs with sip trace (unmodified) attached.

Thanks
Mike

> On Jun 12, 2017, at 4:32 PM, Daniel Greenwald <dig1234 at gmail.com> wrote:
> 
> We have upgraded to latest STABLE from Ubuntu packages but we are still seeing rport in TCP calls:
> 
> 
> INVITE sip:xxxxxxx at sip.freeswitch.com <mailto:sip%3Axxxxxxx at sip.freeswitch.com>;transport=tcp SIP/2.0
> Via: SIP/2.0/TCP x.x.x.x;rport;branch=z9hG4bKZDKF5N8QcFZcH
> Max-Forwards: 70
> From: "user" <sip:xxxxxx at sip.freeswitch.com <mailto:sip%3Axxxxxx at sip.freeswitch.com>>;tag=Z1pQm54mB22De
> To: <sip:xxxxxx at sip.freeswitch.com <mailto:sip%3Axxxxxx at sip.freeswitch.com>;transport=tcp>
> Call-ID: c79656fc-d2d0-4446-90c3-060dabf82fd6
> CSeq: 108312748 INVITE
> Contact: <sip:user at x.x.x.x:5060;transport=tcp;gw=provider>
> User-Agent: FreeSWITCH-mod_sofia/1.6.10-4-726448d~64bit
> 
> On Fri, Jun 9, 2017 at 2:56 PM, Anthony Minessale <anthony.minessale at gmail.com <mailto:anthony.minessale at gmail.com>> wrote:
> Odd number releases are not stable releases they are dev releases.
> The version you quoted is from may 2015 2 years ago.
> 
> The latest stable release is 1.6.17 and latest build release changes every day.
> 
> 
> 
> On Fri, Jun 9, 2017 at 10:50 AM, Daniel Greenwald <dig1234 at gmail.com <mailto:dig1234 at gmail.com>> wrote:
> We have noticed that FS is sending RPORT in TCP calls to a gateway. It was reported as fixed in this bug:
> https://freeswitch.org/jira/browse/FS-6612 <https://freeswitch.org/jira/browse/FS-6612>
> 
> We are running:
> 1.5.15b+git~20150512T053645Z~9eb887af47~64bit
> 
> I am not sure why RPORT is still being sent. Is this there a config parameter which needs to be set to suppress the RPORT? Or was this change reverted in later versions for some reason. Provider is telling us we should not be sending RPORT in TCP...
> 
> Any info would be greatly appreciated. 
> 
> Thanks!
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> http://www.cluecon.com <http://www.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>
> http://www.freeswitch.org <http://www.freeswitch.org/>
> 
> 
> 
> -- 
> Anthony Minessale II       ♬ @anthmfs  ♬ @FreeSWITCH  ♬
> 
> ☞ http://freeswitch.org/ <http://freeswitch.org/>  ☞ http://cluecon.com/ <http://cluecon.com/>  ☞ http://twitter.com/FreeSWITCH <http://twitter.com/FreeSWITCH>
> ☞ irc.freenode.net <http://irc.freenode.net/> #freeswitch ☞ http://freeswitch.org/g+ <http://freeswitch.org/g+>
> 
> ClueCon Weekly Development Call 
> ☎ sip:888 at conference.freeswitch.org <mailto:sip%3A888 at conference.freeswitch.org>  ☎ +19193869900 <tel:(919)%20386-9900> 
> 
> https://www.youtube.com/watch?v=9XXgW34t40s <https://www.youtube.com/watch?v=9XXgW34t40s>
> https://www.youtube.com/watch?v=NLaDpGQuZDA <https://www.youtube.com/watch?v=NLaDpGQuZDA>
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> http://www.cluecon.com <http://www.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>
> http://www.freeswitch.org <http://www.freeswitch.org/>
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170612/7d1855d1/attachment-0001.html>

From prex5609 at gmail.com  Tue Jun 13 00:28:44 2017
From: prex5609 at gmail.com (Peter Rex)
Date: Mon, 12 Jun 2017 18:28:44 -0600
Subject: [Freeswitch-users] Debian Stretch
Message-ID: <CAKDN4FeqnOGn+45=5GXoNM8xqmks66ew8uM0GX4JiFEVrqBd+w@mail.gmail.com>

Stretch is the new stable on Saturday. I've looked through Confluence and
the mailing lists but I can't find anything relevant. I see interesting
possibilities at http://files.freeswitch.org/repo/deb, but I thought I
would ask the mailing list if there's a plan yet to add or move the
_production_ build to Stretch.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170612/742a188b/attachment.html>

From mike at jerris.com  Tue Jun 13 00:44:20 2017
From: mike at jerris.com (Michael Jerris)
Date: Mon, 12 Jun 2017 20:44:20 -0400
Subject: [Freeswitch-users] Debian Stretch
In-Reply-To: <CAKDN4FeqnOGn+45=5GXoNM8xqmks66ew8uM0GX4JiFEVrqBd+w@mail.gmail.com>
References: <CAKDN4FeqnOGn+45=5GXoNM8xqmks66ew8uM0GX4JiFEVrqBd+w@mail.gmail.com>
Message-ID: <CF850B37-D178-46C8-8AE6-6BB4CF6C4DF6@jerris.com>

Stretch won’t build yet.  I’ll have some patches over the next few weeks to fix that.  1.8 when released will likely target Stretch as its primary but still a bunch of testing to do.  The patches to fix build for stretch will go back into 1.6 branch, once they are complete and tested.


> On Jun 12, 2017, at 8:28 PM, Peter Rex <prex5609 at gmail.com> wrote:
> 
> Stretch is the new stable on Saturday. I've looked through Confluence and the mailing lists but I can't find anything relevant. I see interesting possibilities at http://files.freeswitch.org/repo/deb <http://files.freeswitch.org/repo/deb>, but I thought I would ask the mailing list if there's a plan yet to add or move the _production_ build to Stretch.

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170612/bf833c8f/attachment.html>

From prex5609 at gmail.com  Tue Jun 13 01:32:31 2017
From: prex5609 at gmail.com (Peter Rex)
Date: Mon, 12 Jun 2017 19:32:31 -0600
Subject: [Freeswitch-users] Debian Stretch
In-Reply-To: <CF850B37-D178-46C8-8AE6-6BB4CF6C4DF6@jerris.com>
References: <CAKDN4FeqnOGn+45=5GXoNM8xqmks66ew8uM0GX4JiFEVrqBd+w@mail.gmail.com>
 <CF850B37-D178-46C8-8AE6-6BB4CF6C4DF6@jerris.com>
Message-ID: <CAKDN4Ffs7g4cGAqr-YJrO9Cj=p9D4jV0fxrUnmgZ00hcV1bD0w@mail.gmail.com>

Thanks Michael. Hate to do this to you, but is there an estimate on 1.8
timeframe? Mailing list shows people were talking about configs and feature
requests in January, but can't see much else. Maybe I'm not looking in the
right place.

On Mon, Jun 12, 2017 at 6:44 PM, Michael Jerris <mike at jerris.com> wrote:

> Stretch won’t build yet.  I’ll have some patches over the next few weeks
> to fix that.  1.8 when released will likely target Stretch as its primary
> but still a bunch of testing to do.  The patches to fix build for stretch
> will go back into 1.6 branch, once they are complete and tested.
>
>
> On Jun 12, 2017, at 8:28 PM, Peter Rex <prex5609 at gmail.com> wrote:
>
> Stretch is the new stable on Saturday. I've looked through Confluence and
> the mailing lists but I can't find anything relevant. I see interesting
> possibilities at http://files.freeswitch.org/repo/deb, but I thought I
> would ask the mailing list if there's a plan yet to add or move the
> _production_ build to Stretch.
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170612/a5b37f92/attachment.html>

From mike at jerris.com  Tue Jun 13 02:14:03 2017
From: mike at jerris.com (Michael Jerris)
Date: Mon, 12 Jun 2017 22:14:03 -0400
Subject: [Freeswitch-users] Debian Stretch
In-Reply-To: <CAKDN4Ffs7g4cGAqr-YJrO9Cj=p9D4jV0fxrUnmgZ00hcV1bD0w@mail.gmail.com>
References: <CAKDN4FeqnOGn+45=5GXoNM8xqmks66ew8uM0GX4JiFEVrqBd+w@mail.gmail.com>
 <CF850B37-D178-46C8-8AE6-6BB4CF6C4DF6@jerris.com>
 <CAKDN4Ffs7g4cGAqr-YJrO9Cj=p9D4jV0fxrUnmgZ00hcV1bD0w@mail.gmail.com>
Message-ID: <733A7BE7-6E66-4BD8-8736-B6DDBFE695A4@jerris.com>

announcements will come out when we have real dates.

> On Jun 12, 2017, at 9:32 PM, Peter Rex <prex5609 at gmail.com> wrote:
> 
> Thanks Michael. Hate to do this to you, but is there an estimate on 1.8 timeframe? Mailing list shows people were talking about configs and feature requests in January, but can't see much else. Maybe I'm not looking in the right place.
> 
> On Mon, Jun 12, 2017 at 6:44 PM, Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>> wrote:
> Stretch won’t build yet.  I’ll have some patches over the next few weeks to fix that.  1.8 when released will likely target Stretch as its primary but still a bunch of testing to do.  The patches to fix build for stretch will go back into 1.6 branch, once they are complete and tested.
> 
> 
>> On Jun 12, 2017, at 8:28 PM, Peter Rex <prex5609 at gmail.com <mailto:prex5609 at gmail.com>> wrote:
>> 
>> Stretch is the new stable on Saturday. I've looked through Confluence and the mailing lists but I can't find anything relevant. I see interesting possibilities at http://files.freeswitch.org/repo/deb <http://files.freeswitch.org/repo/deb>, but I thought I would ask the mailing list if there's a plan yet to add or move the _production_ build to Stretch.

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170612/f4b7945f/attachment.html>

From vbvbrj at gmail.com  Tue Jun 13 06:05:21 2017
From: vbvbrj at gmail.com (Mimiko)
Date: Tue, 13 Jun 2017 09:05:21 +0300
Subject: [Freeswitch-users] callcenter member recall same agent.
In-Reply-To: <CABMX_w8-T5SYqvmXR=e6+oesdwjTScRymRNvU63HKu4YHwi8yg@mail.gmail.com>
References: <0f5df8c3-1e59-0a85-82b1-08757a33673c@gmail.com>
 <CABMX_w8-T5SYqvmXR=e6+oesdwjTScRymRNvU63HKu4YHwi8yg@mail.gmail.com>
Message-ID: <677a8fd6-a4f2-c009-b0cd-52576a5574c1@gmail.com>

Ok. There is no such feature.

Is it hard to implement if paying?

On 12.06.2017 18:12, Ítalo Rossi wrote:
> There's no such feature yet...
> 
> On Mon, Jun 12, 2017 at 9:22 AM, Mimiko <vbvbrj at gmail.com <mailto:vbvbrj at gmail.com>> wrote:
> 
>     Hello.
> 
>     I have a question.
> 
>     Is there a parameter that a member, after talking to some agent, leaving callcenter and then call back will be connected specifically to same
>     agent, but be in same queue? If the agent is no more linked to the queue, then member will be connected to any other agent like normally.
>     If agent is still in system and is talking - then member will wait this agent to be free.


-- 
Mimiko desu.


From kbdfck at gmail.com  Tue Jun 13 07:52:29 2017
From: kbdfck at gmail.com (Dmitry Sytchev)
Date: Tue, 13 Jun 2017 10:52:29 +0300
Subject: [Freeswitch-users] User CIDR and ACL
In-Reply-To: <75fee20d-577d-41d5-bbdf-77628144691e@Spark>
References: <75fee20d-577d-41d5-bbdf-77628144691e@Spark>
Message-ID: <CA+GmxbZztNqv0tuTRKVjBjg=ZqVA3zQFiQkVQk=oFcp=DjYGdg@mail.gmail.com>

With CIDR it is possible to bypass password auth for specified networks

2017-06-12 20:12 GMT+03:00 Igor Olhovskiy <igorolhovskiy at gmail.com>:
> Hi!
>
> Can you pls help me with little misunderstanding.
> According to
> https://freeswitch.org/confluence/display/FREESWITCH/ACL#ACL-Users
> I can create ACL for a user and also specify a CIDR in <user> tag.
> Like
> <user id="1000" cidr="12.34.56.78/32,20.0.0.0/8»>
> So, what is difference between cidr parameter and «auth-acl» parameter, and
> can I specify, that users can register only from specified network range?
>
> Like I want my user 1001 register only from 10.0.20.0/24 and user 1002
> register only from 10.0.30.0/24, but they must register with username and
> pass.
>
> Regards, Igor
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org



-- 
Best regards,

Dmitry Sytchev,
IT Engineer


From raman.chv at gmail.com  Tue Jun 13 08:35:26 2017
From: raman.chv at gmail.com (Ram)
Date: Tue, 13 Jun 2017 14:05:26 +0530
Subject: [Freeswitch-users] Record-routes in NOTIFY
Message-ID: <CAArNKuejWc2jm-YHesBwG35TpzmhJ45V9hB=-d=hK61NdhV1pQ@mail.gmail.com>

Hi,

Record routes in SUBSCRIBE is not honored in NOTIFY, In my case i am having
3 record routes in SUBSCRIBE, but only one i.e top record route is used for
NOTIFY is causing routing issue. I am using freeswitch version 1.6.17 for
testing.

Following is the trace for SUBSCRIBE and NOTIFY at freeswitch.

T 2017/06/13 08:20:14.868294 10.1.30.27:55503 -> 10.2.30.63:5060 [AP]
SUBSCRIBE sip:500 at 52.64.221.219:5060;transport=tcp SIP/2.0
Record-Route: <sip:10.1.30.27:5060;transport=tcp;ftag=46baee66;lr>
Record-Route: <sip:10.1.30.174;transport=tcp;r2=on;lr=on>
Record-Route: <sip:10.1.30.178:5061;transport=tls;r2=on;lr=on>
Via: SIP/2.0/TCP
10.1.30.199;branch=z9hG4bKc038.62a9c22ab84694b453503a45210a1392.0;i=1
Via: SIP/2.0/TCP
10.1.30.174;branch=z9hG4bKc038.8d466a0ae4f1821a3f0d08e6602cdadc.0;i=82
Via: SIP/2.0/TLS 10.1.30.146:51890
;received=10.1.30.146;rport=51890;branch=z9hG4bKd1633fda00007
From: "RamanTest"<sip:ramantest at freeconf.com>;tag=46baee66
To: "sip:500 at freeconf.com"<sip:500 at freeconf.com
;transport=tls>;tag=pv4B8Q9XUDtgD
Call-ID: 2d118609-1 at 10.1.30.180
CSeq: 1805684444 SUBSCRIBE
Max-Forwards: 69
Contact: "RamanTest"<sip:ramantest at 10.1.30.146:51890;transport=tls>
User-Agent: TestConference
Event: conference
Expires: 3600
Allow:
INVITE,ACK,BYE,CANCEL,REFER,NOTIFY,OPTIONS,PRACK,UPDATE,INFO,MESSAGE,SUBSCRIBE,PUBLISH
Allow-Events: refer, presence
Supported: replaces, timer, gruu, join
Date: Tue, 13 Jun 2017 08:24:00 GMT
Content-Length: 0


##
T 2017/06/13 08:20:14.873026 10.2.30.63:5060 -> 10.1.30.27:55503 [AP]
SIP/2.0 202 Accepted
Via: SIP/2.0/TCP
10.1.30.199;branch=z9hG4bKc038.62a9c22ab84694b453503a45210a1392.0;i=1;received=10.1.30.27;rport=55503
Via: SIP/2.0/TCP
10.1.30.174;branch=z9hG4bKc038.8d466a0ae4f1821a3f0d08e6602cdadc.0;i=82
Via: SIP/2.0/TLS 10.1.30.146:51890
;received=10.1.30.146;rport=51890;branch=z9hG4bKd1633fda00007
Record-Route: <sip:10.1.30.27:5060;transport=tcp;ftag=46baee66;lr>
Record-Route: <sip:10.1.30.174;transport=tcp;r2=on;lr=on>
Record-Route: <sip:10.1.30.178:5061;transport=tls;r2=on;lr=on>
From: "RamanTest" <sip:ramantest at freeconf.com>;tag=46baee66
To: "sip:500 at freeconf.com" <sip:500 at freeconf.com
;transport=tls>;tag=pv4B8Q9XUDtgD
Call-ID: 2d118609-1 at 10.1.30.180
CSeq: 1805684444 SUBSCRIBE
Contact: <sip:500 at 52.64.221.219:5060;transport=tcp>
Expires: 3600
User-Agent: FreeSWITCH-mod_sofia/1.6.17~64bit
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, presence, as-feature-event, dialog,
line-seize, call-info, sla, include-session-description, presence.winfo,
message-summary, refer
Subscription-State: active;expires=3600
Content-Length: 0


####
T 2017/06/13 08:20:15.173591 10.2.30.63:58879 -> 10.1.30.27:5060 [AP]
NOTIFY sip:ramantest at 10.1.30.146:51890;transport=tls SIP/2.0
Via: SIP/2.0/TCP 52.64.221.219;rport;branch=z9hG4bKvr9Kyp8Fe829g
Route: <sip:10.1.30.27:5060>;transport=tcp;ftag=46baee66;lr
Record-Route: <sip:10.1.30.27:5060>;transport=tcp;ftag=46baee66;lr
Max-Forwards: 70
From: "sip:500 at freeconf.com" <sip:500 at freeconf.com
;transport=tls>;tag=pv4B8Q9XUDtgD;tag=pv4B8Q9XUDtgD
To: "RamanTest" <sip:ramantest at freeconf.com>;tag=46baee66
Call-ID: 2d118609-1 at 10.1.30.180
CSeq: 705660701 NOTIFY
Contact: <sip:500 at 52.64.221.219:5060;transport=tcp>;isfocus
User-Agent: FreeSWITCH-mod_sofia/1.6.17~64bit
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, path, replaces
Event: conference
Allow-Events: talk, hold, conference, presence, as-feature-event, dialog,
line-seize, call-info, sla, include-session-description, presence.winfo,
message-summary, refer
Subscription-State: active;expires=3600
Content-Type: application/conference-info+xml
Content-Length: 1028

<?xml version="1.0"?>
<conference-info version="1" state="full"
xmlns="urn:ietf:params:xml:ns:conference-info" entity="sip:500 at freeconf.com
">
  <conference-description>
    <display-text>FreeSWITCH Conference</display-text>
    <conf-uris>
      <entry>
        <uri>sip:500 at freeconf.com</uri>
      </entry>
    </conf-uris>
  </conference-description>
  <conference-state>
    <user-count>1</user-count>
    <active>true</active>
  </conference-state>
  <users>
    <user state="full" entity="sip:ramantest at freeconf.com">
      <display-text>RamanTest</display-text>
      <endpoint entity="sip:ramantest at freeconf.com">
        <display-text>RamanTest</display-text>
        <status>connected</status>
        <joining-info>
          <when>2017-06-13T08:20:13+00:00</when>
        </joining-info>
        <media id="2a">
          <type>audio</type>
          <src-id>4048072604</src-id>
          <status>sendrecv</status>
        </media>
      </endpoint>
    </user>
  </users>
</conference-info>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170613/82a15c28/attachment.html>

From agubbe at gmail.com  Tue Jun 13 11:24:26 2017
From: agubbe at gmail.com (=?UTF-8?Q?Agust=C3=AD_Ubalde_Bellot?=)
Date: Tue, 13 Jun 2017 13:24:26 +0200
Subject: [Freeswitch-users] Freeswitch sslv3 support
Message-ID: <CAGCz3RBiBhE7ism+Wfr81HShk42TDht4c-Yp_DY0cp35OCNqLQ@mail.gmail.com>

Hi all,

Is there a FreeSWITCH update where sslv3 support is disabled?


Thanks,
Agustí
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170613/3fb9ee9b/attachment.html>

From sdpaek21 at gmail.com  Tue Jun 13 12:59:31 2017
From: sdpaek21 at gmail.com (Sp Pho)
Date: Tue, 13 Jun 2017 15:59:31 +0300
Subject: [Freeswitch-users] B-Leg Early Media - Normal Clearing vs
	Originator Cancelled
Message-ID: <CAL0o4X=G8iikibMhyaW4c71AJhvc6EPgNss1LpjXpiL=t-9BJw@mail.gmail.com>

Hi,

I'm creating a bridge to terminate call arriving inbound from a DID.

Everything works fine, however the termination point plays a recorded
message - which as rightfully so by the switch is seen as early media (with
pre-answer).

However, when the originator hangs up while the early media is playing, the
CDRs indicate NORMAL_CLEARING. If ignore early media is enabled, the media
does not play and ORIGINATOR_CANCELis the hang-up cause.

Is there any way to receive the early media and from the CDR know whether
or not the call was subsequently answered or not by the agent (i.e. if the
originator canceled the call during the early media or while on hold)?

Thanks in advance,
SP
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170613/0370a581/attachment.html>

From brian at freeswitch.org  Tue Jun 13 13:55:58 2017
From: brian at freeswitch.org (Brian West)
Date: Tue, 13 Jun 2017 08:55:58 -0500
Subject: [Freeswitch-users] Freeswitch sslv3 support
In-Reply-To: <CAGCz3RBiBhE7ism+Wfr81HShk42TDht4c-Yp_DY0cp35OCNqLQ@mail.gmail.com>
References: <CAGCz3RBiBhE7ism+Wfr81HShk42TDht4c-Yp_DY0cp35OCNqLQ@mail.gmail.com>
Message-ID: <CAEJMVkASnTR+COBdrChMvgNb2NQBd2HiwuOwCTybeCegn8bZnQ@mail.gmail.com>

You can already disable it via config.  Our vanilla config already ships
with only 'tlsv1,tlsv1.1,tlsv1.2' enabled.

/b


On Tue, Jun 13, 2017 at 6:24 AM, Agustí Ubalde Bellot <agubbe at gmail.com>
wrote:

> Hi all,
>
> Is there a FreeSWITCH update where sslv3 support is disabled?
>
>
> Thanks,
> Agustí
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org
>



-- 

*Brian West*
brian at freeswitch.org

*Twitter: @FreeSWITCH , @briankwest*

http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

Book a phone call (CST) <https://freeswitch.com/appointment>

Allison prompts for FreeSWITCH:

*https://www.gofundme.com/allison-prompts-for-freeswitch*
<https://www.gofundme.com/allison-prompts-for-freeswitch>

Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
/r/freeswitch <https://www.reddit.com/r/freeswitch>

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*Skype:*briankwest
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170613/acfc4a23/attachment.html>

From brian at freeswitch.org  Tue Jun 13 13:57:10 2017
From: brian at freeswitch.org (Brian West)
Date: Tue, 13 Jun 2017 08:57:10 -0500
Subject: [Freeswitch-users] Record-routes in NOTIFY
In-Reply-To: <CAArNKuejWc2jm-YHesBwG35TpzmhJ45V9hB=-d=hK61NdhV1pQ@mail.gmail.com>
References: <CAArNKuejWc2jm-YHesBwG35TpzmhJ45V9hB=-d=hK61NdhV1pQ@mail.gmail.com>
Message-ID: <CAEJMVkDCFmvyNX9Xb_zV=fOybZ8dr2apPf9QFE24TPSKiDOQPQ@mail.gmail.com>

Any bug reports belong on JIRA,
https://freeswitch.org/confluence/display/FREESWITCH/Reporting+Bugs+to+JIRA

Thanks,


On Tue, Jun 13, 2017 at 3:35 AM, Ram <raman.chv at gmail.com> wrote:

> Hi,
>
> Record routes in SUBSCRIBE is not honored in NOTIFY, In my case i am
> having 3 record routes in SUBSCRIBE, but only one i.e top record route is
> used for NOTIFY is causing routing issue. I am using freeswitch version
> 1.6.17 for testing.
>
> Following is the trace for SUBSCRIBE and NOTIFY at freeswitch.
>
> T 2017/06/13 08:20:14.868294 10.1.30.27:55503 -> 10.2.30.63:5060 [AP]
> SUBSCRIBE sip:500 at 52.64.221.219:5060;transport=tcp SIP/2.0
> Record-Route: <sip:10.1.30.27:5060;transport=tcp;ftag=46baee66;lr>
> Record-Route: <sip:10.1.30.174;transport=tcp;r2=on;lr=on>
> Record-Route: <sip:10.1.30.178:5061;transport=tls;r2=on;lr=on>
> Via: SIP/2.0/TCP 10.1.30.199;branch=z9hG4bKc038.
> 62a9c22ab84694b453503a45210a1392.0;i=1
> Via: SIP/2.0/TCP 10.1.30.174;branch=z9hG4bKc038.
> 8d466a0ae4f1821a3f0d08e6602cdadc.0;i=82
> Via: SIP/2.0/TLS 10.1.30.146:51890;received=10.
> 1.30.146;rport=51890;branch=z9hG4bKd1633fda00007
> From: "RamanTest"<sip:ramantest at freeconf.com>;tag=46baee66
> To: "sip:500 at freeconf.com"<sip:500 at freeconf.com;transport=
> tls>;tag=pv4B8Q9XUDtgD
> Call-ID: 2d118609-1 at 10.1.30.180
> CSeq: 1805684444 SUBSCRIBE
> Max-Forwards: 69
> Contact: "RamanTest"<sip:ramantest at 10.1.30.146:51890;transport=tls>
> User-Agent: TestConference
> Event: conference
> Expires: 3600
> Allow: INVITE,ACK,BYE,CANCEL,REFER,NOTIFY,OPTIONS,PRACK,UPDATE,
> INFO,MESSAGE,SUBSCRIBE,PUBLISH
> Allow-Events: refer, presence
> Supported: replaces, timer, gruu, join
> Date: Tue, 13 Jun 2017 08:24:00 GMT
> Content-Length: 0
>
>
> ##
> T 2017/06/13 08:20:14.873026 10.2.30.63:5060 -> 10.1.30.27:55503 [AP]
> SIP/2.0 202 Accepted
> Via: SIP/2.0/TCP 10.1.30.199;branch=z9hG4bKc038.
> 62a9c22ab84694b453503a45210a1392.0;i=1;received=10.1.30.27;rport=55503
> Via: SIP/2.0/TCP 10.1.30.174;branch=z9hG4bKc038.
> 8d466a0ae4f1821a3f0d08e6602cdadc.0;i=82
> Via: SIP/2.0/TLS 10.1.30.146:51890;received=10.
> 1.30.146;rport=51890;branch=z9hG4bKd1633fda00007
> Record-Route: <sip:10.1.30.27:5060;transport=tcp;ftag=46baee66;lr>
> Record-Route: <sip:10.1.30.174;transport=tcp;r2=on;lr=on>
> Record-Route: <sip:10.1.30.178:5061;transport=tls;r2=on;lr=on>
> From: "RamanTest" <sip:ramantest at freeconf.com>;tag=46baee66
> To: "sip:500 at freeconf.com" <sip:500 at freeconf.com;transport=tls>;tag=
> pv4B8Q9XUDtgD
> Call-ID: 2d118609-1 at 10.1.30.180
> CSeq: 1805684444 SUBSCRIBE
> Contact: <sip:500 at 52.64.221.219:5060;transport=tcp>
> Expires: 3600
> User-Agent: FreeSWITCH-mod_sofia/1.6.17~64bit
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
> REFER, NOTIFY, PUBLISH, SUBSCRIBE
> Supported: timer, path, replaces
> Allow-Events: talk, hold, conference, presence, as-feature-event, dialog,
> line-seize, call-info, sla, include-session-description, presence.winfo,
> message-summary, refer
> Subscription-State: active;expires=3600
> Content-Length: 0
>
>
> ####
> T 2017/06/13 08:20:15.173591 10.2.30.63:58879 -> 10.1.30.27:5060 [AP]
> NOTIFY sip:ramantest at 10.1.30.146:51890;transport=tls SIP/2.0
> Via: SIP/2.0/TCP 52.64.221.219;rport;branch=z9hG4bKvr9Kyp8Fe829g
> Route: <sip:10.1.30.27:5060>;transport=tcp;ftag=46baee66;lr
> Record-Route: <sip:10.1.30.27:5060>;transport=tcp;ftag=46baee66;lr
> Max-Forwards: 70
> From: "sip:500 at freeconf.com" <sip:500 at freeconf.com;transport=tls>;tag=
> pv4B8Q9XUDtgD;tag=pv4B8Q9XUDtgD
> To: "RamanTest" <sip:ramantest at freeconf.com>;tag=46baee66
> Call-ID: 2d118609-1 at 10.1.30.180
> CSeq: 705660701 NOTIFY
> Contact: <sip:500 at 52.64.221.219:5060;transport=tcp>;isfocus
> User-Agent: FreeSWITCH-mod_sofia/1.6.17~64bit
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
> REFER, NOTIFY, PUBLISH, SUBSCRIBE
> Supported: timer, path, replaces
> Event: conference
> Allow-Events: talk, hold, conference, presence, as-feature-event, dialog,
> line-seize, call-info, sla, include-session-description, presence.winfo,
> message-summary, refer
> Subscription-State: active;expires=3600
> Content-Type: application/conference-info+xml
> Content-Length: 1028
>
> <?xml version="1.0"?>
> <conference-info version="1" state="full" xmlns="urn:ietf:params:xml:ns:conference-info"
> entity="sip:500 at freeconf.com">
>   <conference-description>
>     <display-text>FreeSWITCH Conference</display-text>
>     <conf-uris>
>       <entry>
>         <uri>sip:500 at freeconf.com</uri>
>       </entry>
>     </conf-uris>
>   </conference-description>
>   <conference-state>
>     <user-count>1</user-count>
>     <active>true</active>
>   </conference-state>
>   <users>
>     <user state="full" entity="sip:ramantest at freeconf.com">
>       <display-text>RamanTest</display-text>
>       <endpoint entity="sip:ramantest at freeconf.com">
>         <display-text>RamanTest</display-text>
>         <status>connected</status>
>         <joining-info>
>           <when>2017-06-13T08:20:13+00:00</when>
>         </joining-info>
>         <media id="2a">
>           <type>audio</type>
>           <src-id>4048072604 <(404)%20807-2604></src-id>
>           <status>sendrecv</status>
>         </media>
>       </endpoint>
>     </user>
>   </users>
> </conference-info>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org
>



-- 

*Brian West*
brian at freeswitch.org

*Twitter: @FreeSWITCH , @briankwest*

http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

Book a phone call (CST) <https://freeswitch.com/appointment>

Allison prompts for FreeSWITCH:

*https://www.gofundme.com/allison-prompts-for-freeswitch*
<https://www.gofundme.com/allison-prompts-for-freeswitch>

Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
/r/freeswitch <https://www.reddit.com/r/freeswitch>

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*Skype:*briankwest
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170613/7ede93a0/attachment-0001.html>

From joel at gogii.net  Tue Jun 13 14:38:57 2017
From: joel at gogii.net (Joel Serrano)
Date: Tue, 13 Jun 2017 07:38:57 -0700
Subject: [Freeswitch-users] Multiple FreeSWITCH servers behind
	kamailio-websocket
In-Reply-To: <CAFArqsZjjVP63_OscW=qVOh_skaddndKuXzKaWJp2WgHg9KaEw@mail.gmail.com>
References: <CAFArqsZjjVP63_OscW=qVOh_skaddndKuXzKaWJp2WgHg9KaEw@mail.gmail.com>
Message-ID: <CAMtXxQkrCf0+iv9-5e0VtQFN-wuAXAPcbboSV4FfPwraAmFDFg@mail.gmail.com>

Hi Karsten,

Have you tried with regular Kamailio (w/ dispatcher+websocket+xhttp
modules)? I don't see why it wouldn't work...

Joel.


On Wed, Jun 7, 2017 at 4:47 AM, Karsten Horsmann <khorsmann at gmail.com>
wrote:

> Hello List,
>
>
> is there any howto about webrtc loadbalance in combination with kamailio
> and FreeSWITCH?
>
> I want to share one WSS address/endpoint to multiple FreeSWITCH backends.
> Or is there any other best practice?
>
> My callflow is mostly that my internal SIP Servers called my registered
> webrtc clients.
>
> Would be nice to get some input.
>
> --
> Kind Regards
> *Karsten Horsmann*
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170613/96d1cf0c/attachment.html>

From eastour at 163.com  Tue Jun 13 14:19:07 2017
From: eastour at 163.com (chenyzhi)
Date: Tue, 13 Jun 2017 22:19:07 +0800 (CST)
Subject: [Freeswitch-users] When calling out ,
 freeswitch doesn't get DTMF ?
In-Reply-To: <CAFLo2QwwwE6xeXBN9DkM1VEecEJpwnDMmQ7iT9EvSyx9RqE9mg@mail.gmail.com>
References: <f823cdd8-a354-3698-d8c5-484b1b4cc8b4@xbipin.com>
 <CAEJMVkDDQX=Ni3AOQxQOo=joGoAeYroxioVP=cu9w6Y2m7kLPA@mail.gmail.com>
 <15c898b5d58.2779.b07ebdf329620b8089087c7205b03f01@xbipin.com>
 <310cf71c.8fc3.15c8ba9c4f6.Coremail.eastour@163.com>
 <CAFGRPVqFA+tpaOQkAGGCR7s5gY0z0TbdArWFdkfnsaaG4Z9Dgw@mail.gmail.com>
 <2a18422f.ef8.15c99b7b9bb.Coremail.eastour@163.com>
 <CALXCt0pKKEJoSeXXZ=tn04nifEZvQCRSFCDKL8jSu+pfOgvRNQ@mail.gmail.com>
 <6c11dd0f.c2c1.15c9c727dc7.Coremail.eastour@163.com>
 <CAFLo2QwwwE6xeXBN9DkM1VEecEJpwnDMmQ7iT9EvSyx9RqE9mg@mail.gmail.com>
Message-ID: <2f48505d.d9c5.15ca1d28424.Coremail.eastour@163.com>

It's a one leg call .there is no b-leg.


please make a test on your freeswitch box .


just type the command "originate user/1001 5000" on the freeswitch console to see if your freeswitch instance can detect dtmf input.






At 2017-06-12 22:57:17, "Luis Daniel Lucio Quiroz" <luis.daniel.lucio at gmail.com> wrote:

check if you have transcoding, and if you do, check that dftm type-codec on leg b are compatible.


--
Luis Daniel Lucio Quiroz
CISSP, CISM, CISA
Linux, VoIP and much more fun
www.okay.com.mx

Need LCR? Check out LCR for FusionPBX with FreeSWITCH
Need Billing? Check out Billing for FusionPBX with FreeSWITCH


On Mon, Jun 12, 2017 at 9:16 AM, chenyzhi <eastour at 163.com> wrote:

I don't think It's  the DTMF type , because  when I dial 5000 from x-lite (which has registered to freeswitch as 1001) ,I can hear the voice and if I press any dtmf on x-lite,  freeswitch can recieve the dtmfs.  This means that the DTMF type is correct ,otherwise freeswitch coudn't have received the dtmfs;


 when I enter the command  "originate user/1001 5000"  at the freeswitch console , my xlite will ring ,and I answered ,I can hear the voice ,I press some dtmf,but freeswitch can NOT receive any dtmf.  really weird.






At 2017-06-12 17:37:49, "Giovanni Maruzzelli" <gmaruzz at gmail.com> wrote:

you need to check the DTMF type, you probably are using the wrong one (info-inband-rfc2833), and for some reason they are not negotiated



On 12 June 2017 at 02:32, chenyzhi <eastour at 163.com> wrote:

I have read the logs ,but I didn't find any difference.


Please make a test to see if this happens in your box.






At 2017-06-10 06:49:34, "David Villasmil" <david.villasmil.work at gmail.com> wrote:
Have you looked at the log? Bump the logging up and see what shows up... what you're seeing is very weird

David

On Fri, Jun 9, 2017 at 11:18 AM chenyzhi <eastour at 163.com> wrote:

Hi


call 5000 from x-lite , you can hear the IVR voice and if you press dtmf keys , freeswitch can receive the dtmf keys .


but , if you enter the command  "originate user/1001 5000"  , x-lite will ring ,answer it ,you can hear the IVR voice , press some keys ,the freeswitch can NOT receive any dtmf ,


why?




 

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.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
http://www.freeswitch.org




 


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.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
http://www.freeswitch.org




--


Sincerely,

Giovanni Maruzzelli
OpenTelecom.IT
cell: +39 347 266 56 18





 


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.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
http://www.freeswitch.org


-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170613/a42bde6c/attachment-0001.html>

From zzyroy at qq.com  Tue Jun 13 14:35:18 2017
From: zzyroy at qq.com (=?ISO-8859-1?B?enp5?=)
Date: Tue, 13 Jun 2017 22:35:18 +0800
Subject: [Freeswitch-users] How to use originate make A_leg to a queue
Message-ID: <tencent_423E77595F820C7F5B5862DB@qq.com>

Dear All,

I'm testing mod_callcenter now.

Is there any way to make A_leg to the call center queue first then bridge B_leg by dialplan?

Just like [ originate group/sales+A 8888 XML default ]

Thanks.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170613/b9631b76/attachment-0001.html>

From david.villasmil.work at gmail.com  Tue Jun 13 19:32:24 2017
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Tue, 13 Jun 2017 19:32:24 +0000
Subject: [Freeswitch-users] How to use originate make A_leg to a queue
In-Reply-To: <tencent_423E77595F820C7F5B5862DB@qq.com>
References: <tencent_423E77595F820C7F5B5862DB@qq.com>
Message-ID: <CAFGRPVprRMS8PQZLZPO9QBZgF3V3UpHjm+efL-FkX+sy9TQLAA@mail.gmail.com>

Originate blahnlah &callcenter(yourqueue)
Or something like that
On Tue, Jun 13, 2017 at 3:30 PM zzy <zzyroy at qq.com> wrote:

> Dear All,
>
> I'm testing mod_callcenter now.
>
> Is there any way to make A_leg to the call center queue first then bridge
> B_leg by dialplan?
>
> Just like [ originate group/sales+A 8888 XML default ]
>
> Thanks.
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170613/b3b92244/attachment.html>

From admin at tedssupply.com  Tue Jun 13 21:06:26 2017
From: admin at tedssupply.com (admin)
Date: Tue, 13 Jun 2017 17:06:26 -0400
Subject: [Freeswitch-users] When calling out ,
 freeswitch doesn't get DTMF ?
In-Reply-To: <2f48505d.d9c5.15ca1d28424.Coremail.eastour@163.com>
References: <f823cdd8-a354-3698-d8c5-484b1b4cc8b4@xbipin.com>
 <CAEJMVkDDQX=Ni3AOQxQOo=joGoAeYroxioVP=cu9w6Y2m7kLPA@mail.gmail.com>
 <15c898b5d58.2779.b07ebdf329620b8089087c7205b03f01@xbipin.com>
 <310cf71c.8fc3.15c8ba9c4f6.Coremail.eastour@163.com>
 <CAFGRPVqFA+tpaOQkAGGCR7s5gY0z0TbdArWFdkfnsaaG4Z9Dgw@mail.gmail.com>
 <2a18422f.ef8.15c99b7b9bb.Coremail.eastour@163.com>
 <CALXCt0pKKEJoSeXXZ=tn04nifEZvQCRSFCDKL8jSu+pfOgvRNQ@mail.gmail.com>
 <6c11dd0f.c2c1.15c9c727dc7.Coremail.eastour@163.com>
 <CAFLo2QwwwE6xeXBN9DkM1VEecEJpwnDMmQ7iT9EvSyx9RqE9mg@mail.gmail.com>
 <2f48505d.d9c5.15ca1d28424.Coremail.eastour@163.com>
Message-ID: <59401B92020000310000A34D@mail.tedssupply.com>

I am encountering the same issue.  I am using an ESL call to api originate, with little else except getting in and out of the port, and the caller DTMF is failing to reach the called number.  The called number auto answer attendant does not respond to DTMF, and a call to a test phone confirms no DTMF.  This is a new upgrade from 1.2 to 1.6 and I don't know if this was an issue in 1.2, but my users never complained before 1.6 upgrade.  Ideas?...



>>> chenyzhi  <eastour at 163.com> 06/13/17 3:32 PM >>>
It's a one leg call .there is no b-leg.

please make a test on your freeswitch box .

just type the command "originate user/1001 5000" on the freeswitch console to see if your freeswitch instance can detect dtmf input.







At 2017-06-12 22:57:17, "Luis Daniel Lucio Quiroz" <luis.daniel.lucio at gmail.com> wrote:
 check if you have transcoding, and if you do, check that dftm type-codec on leg b are compatible.

--
Luis Daniel Lucio Quiroz
CISSP, CISM, CISA
Linux, VoIP and much more fun
www.okay.com.mx

Need LCR? Check out LCR for FusionPBX with FreeSWITCH
Need Billing? Check out Billing for FusionPBX with FreeSWITCH



 
On Mon, Jun 12, 2017 at 9:16 AM, chenyzhi <eastour at 163.com> wrote:
I don't think It's  the DTMF type , because  when I dial 5000 from x-lite (which has registered to freeswitch as 1001) ,I can hear the voice and if I press any dtmf on x-lite,  freeswitch can recieve the dtmfs.  This means that the DTMF type is correct ,otherwise freeswitch coudn't have received the dtmfs;

 when I enter the command  "originate user/1001 5000"  at the freeswitch console , my xlite will ring ,and I answered ,I can hear the voice ,I press some dtmf,but freeswitch can NOT receive any dtmf.  really weird.







At 2017-06-12 17:37:49, "Giovanni Maruzzelli" <gmaruzz at gmail.com> wrote:
 you need to check the DTMF type, you probably are using the wrong one (info-inband-rfc2833), and for some reason they are not negotiated

On 12 June 2017 at 02:32, chenyzhi <eastour at 163.com> wrote:
I have read the logs ,but I didn't find any difference.

Please make a test to see if this happens in your box.







At 2017-06-10 06:49:34, "David Villasmil" <david.villasmil.work at gmail.com> wrote:
 Have you looked at the log? Bump the logging up and see what shows up... what you're seeing is very weird

David
On Fri, Jun 9, 2017 at 11:18 AM chenyzhi <eastour at 163.com> wrote:

Hi


 call 5000 from x-lite , you can hear the IVR voice and if you press dtmf keys , freeswitch can receive the dtmf keys .


but , if you enter the command  "originate user/1001 5000"  , x-lite will ring ,answer it ,you can hear the IVR voice , press some keys ,the freeswitch can NOT receive any dtmf ,


why?
 
 



_________________________________________________________________________
 Professional FreeSWITCH Consulting Services:
 consulting at freeswitch.org
 http://www.freeswitchsolutions.com
 
 Official FreeSWITCH Sites
 http://www.freeswitch.org
 http://confluence.freeswitch.org
 http://www.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
 http://www.freeswitch.org
 






_________________________________________________________________________
 Professional FreeSWITCH Consulting Services:
 consulting at freeswitch.org
 http://www.freeswitchsolutions.com
 
 Official FreeSWITCH Sites
 http://www.freeswitch.org
 http://confluence.freeswitch.org
 http://www.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
 http://www.freeswitch.org




-- 

Sincerely,

 Giovanni Maruzzelli
 OpenTelecom.IT
cell: +39 347 266 56 18



 

 






_________________________________________________________________________
 Professional FreeSWITCH Consulting Services:
 consulting at freeswitch.org
 http://www.freeswitchsolutions.com
 
 Official FreeSWITCH Sites
 http://www.freeswitch.org
 http://confluence.freeswitch.org
 http://www.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
 http://www.freeswitch.org



 






-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170613/d51e8d4f/attachment.html>

From gmaruzz at gmail.com  Wed Jun 14 07:53:17 2017
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Wed, 14 Jun 2017 09:53:17 +0200
Subject: [Freeswitch-users] When calling out ,
	freeswitch doesn't get DTMF ?
In-Reply-To: <59401B92020000310000A34D@mail.tedssupply.com>
References: <f823cdd8-a354-3698-d8c5-484b1b4cc8b4@xbipin.com>
 <CAEJMVkDDQX=Ni3AOQxQOo=joGoAeYroxioVP=cu9w6Y2m7kLPA@mail.gmail.com>
 <15c898b5d58.2779.b07ebdf329620b8089087c7205b03f01@xbipin.com>
 <310cf71c.8fc3.15c8ba9c4f6.Coremail.eastour@163.com>
 <CAFGRPVqFA+tpaOQkAGGCR7s5gY0z0TbdArWFdkfnsaaG4Z9Dgw@mail.gmail.com>
 <2a18422f.ef8.15c99b7b9bb.Coremail.eastour@163.com>
 <CALXCt0pKKEJoSeXXZ=tn04nifEZvQCRSFCDKL8jSu+pfOgvRNQ@mail.gmail.com>
 <6c11dd0f.c2c1.15c9c727dc7.Coremail.eastour@163.com>
 <CAFLo2QwwwE6xeXBN9DkM1VEecEJpwnDMmQ7iT9EvSyx9RqE9mg@mail.gmail.com>
 <2f48505d.d9c5.15ca1d28424.Coremail.eastour@163.com>
 <59401B92020000310000A34D@mail.tedssupply.com>
Message-ID: <CALXCt0qw4DCJ2Q0cF+kVEOpjCjp2T8eFgWbYx03sN=z5APBGGQ@mail.gmail.com>

Never heard such problems

Please pastebin your dialplan, your SIP profile, and the complete, since
beginning to end, unedited, debug output of console when receiving a call
which does not get DTMFs



sent from mobile
cell: +39 347 266 56 18
Giovanni Maruzzelli
OpenTelecom.IT

On Jun 13, 2017 23:07, "admin" <admin at tedssupply.com> wrote:

> I am encountering the same issue.  I am using an ESL call to api
> originate, with little else except getting in and out of the port, and the
> caller DTMF is failing to reach the called number.  The called number auto
> answer attendant does not respond to DTMF, and a call to a test phone
> confirms no DTMF.  This is a new upgrade from 1.2 to 1.6 and I don't know
> if this was an issue in 1.2, but my users never complained before 1.6
> upgrade.  Ideas?...
>
>
> >>> chenyzhi  <eastour at 163.com> 06/13/17 3:32 PM >>>
> It's a one leg call .there is no b-leg.
>
> please make a test on your freeswitch box .
>
> just type the command "originate user/1001 5000" on the freeswitch console
> to see if your freeswitch instance can detect dtmf input.
>
>
>
>
>
> At 2017-06-12 22:57:17, "Luis Daniel Lucio Quiroz" <
> luis.daniel.lucio at gmail.com> wrote:
>
> check if you have transcoding, and if you do, check that dftm type-codec
> on leg b are compatible.
>
> --
> Luis Daniel Lucio Quiroz
> CISSP, CISM, CISA
> Linux, VoIP and much more fun
> www.okay.com.mx
>
> Need LCR? Check out LCR for FusionPBX with FreeSWITCH
> Need Billing? Check out Billing for FusionPBX with FreeSWITCH
>
> On Mon, Jun 12, 2017 at 9:16 AM, chenyzhi <eastour at 163.com> wrote:
>
>> I don't think It's  the DTMF type , because  when I dial 5000 from x-lite
>> (which has registered to freeswitch as 1001) ,I can hear the voice and if I
>> press any dtmf on x-lite,  freeswitch can recieve the dtmfs.  This means
>> that the DTMF type is correct ,otherwise freeswitch coudn't have received
>> the dtmfs;
>>
>>  when I enter the command  "originate user/1001 5000"  at the freeswitch
>> console , my xlite will ring ,and I answered ,I can hear the voice ,I press
>> some dtmf,but freeswitch can NOT receive any dtmf.  really weird.
>>
>>
>>
>>
>>
>> At 2017-06-12 17:37:49, "Giovanni Maruzzelli" <gmaruzz at gmail.com> wrote:
>>
>> you need to check the DTMF type, you probably are using the wrong one
>> (info-inband-rfc2833), and for some reason they are not negotiated
>>
>> On 12 June 2017 at 02:32, chenyzhi <eastour at 163.com> wrote:
>>
>>> I have read the logs ,but I didn't find any difference.
>>>
>>> Please make a test to see if this happens in your box.
>>>
>>>
>>>
>>>
>>>
>>> At 2017-06-10 06:49:34, "David Villasmil" <david.villasmil.work at gmail.
>>> com> wrote:
>>>
>>> Have you looked at the log? Bump the logging up and see what shows up...
>>> what you're seeing is very weird
>>>
>>> David
>>> On Fri, Jun 9, 2017 at 11:18 AM chenyzhi <eastour at 163.com> wrote:
>>>
>>>> Hi
>>>>
>>>> call 5000 from x-lite , you can hear the IVR voice and if you press
>>>> dtmf keys , freeswitch can receive the dtmf keys .
>>>>
>>>> but , if you enter the command  "originate user/1001 5000"  , x-lite
>>>> will ring ,answer it ,you can hear the IVR voice , press some keys ,the
>>>> freeswitch can NOT receive any dtmf ,
>>>>
>>>> why?
>>>>
>>>>
>>>>
>>>>
>>>> ____________________________________________________________
>>>> _____________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.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
>>>> http://www.freeswitch.org
>>>
>>>
>>>
>>>
>>>
>>> ____________________________________________________________
>>> _____________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.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
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>>
>> Sincerely,
>>
>> Giovanni Maruzzelli
>> OpenTelecom.IT
>> cell: +39 347 266 56 18
>>
>>
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.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
>> http://www.freeswitch.org
>>
>
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170614/9e4cbaab/attachment-0001.html>

From miconda at gmail.com  Wed Jun 14 08:57:00 2017
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Wed, 14 Jun 2017 10:57:00 +0200
Subject: [Freeswitch-users] [SR-Users] Multiple FreeSWITCH servers
 behind kamailio-websocket
In-Reply-To: <CAMtXxQkrCf0+iv9-5e0VtQFN-wuAXAPcbboSV4FfPwraAmFDFg@mail.gmail.com>
References: <CAFArqsZjjVP63_OscW=qVOh_skaddndKuXzKaWJp2WgHg9KaEw@mail.gmail.com>
 <CAMtXxQkrCf0+iv9-5e0VtQFN-wuAXAPcbboSV4FfPwraAmFDFg@mail.gmail.com>
Message-ID: <407012cb-dc25-b237-17d9-69462307e05b@gmail.com>

Hello,

that combination of dispatcher+websocket+xhttp modules works just fine...

So to load balance the SIP signaling with Kamailio towards FreeSwitch,
just use the dispatcher module as usual. A sample config is available at:

  -
https://www.kamailio.org/docs/modules/stable/modules/dispatcher.html#dispatcher.ex.config

You need to add the support for websocket traffic via websocket module:

  -
https://www.kamailio.org/docs/modules/stable/modules/websocket.html#idp42826164

or extract from the tutorial linked in a previous email on this thread.

Cheers,
Daniel



On 13.06.17 16:38, Joel Serrano wrote:
> Hi Karsten, 
>
> Have you tried with regular Kamailio (w/ dispatcher+websocket+xhttp
> modules)? I don't see why it wouldn't work...
>
> Joel.
>
>
> On Wed, Jun 7, 2017 at 4:47 AM, Karsten Horsmann <khorsmann at gmail.com
> <mailto:khorsmann at gmail.com>> wrote:
>
>     Hello List,
>
>
>     is there any howto about webrtc loadbalance in combination with
>     kamailio and FreeSWITCH?
>
>     I want to share one WSS address/endpoint to multiple FreeSWITCH
>     backends.
>     Or is there any other best practice?
>
>     My callflow is mostly that my internal SIP Servers called my
>     registered webrtc clients.
>
>     Would be nice to get some input.
>
>     -- 
>     Kind Regards
>     *Karsten Horsmann*
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>     <http://www.freeswitchsolutions.com>
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://confluence.freeswitch.org <http://confluence.freeswitch.org>
>     http://www.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>
>     http://www.freeswitch.org
>
>
>
>
> _______________________________________________
> Kamailio (SER) - Users Mailing List
> sr-users at lists.kamailio.org
> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users



-- 
Daniel-Constantin Mierla
www.twitter.com/miconda -- www.linkedin.com/in/miconda
Kamailio Advanced Training - www.asipto.com
Kamailio World Conference - www.kamailioworld.com

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170614/2180a65a/attachment.html>

From nandy1925 at gmail.com  Wed Jun 14 09:43:30 2017
From: nandy1925 at gmail.com (Nandy Dagondon)
Date: Wed, 14 Jun 2017 09:43:30 +0000
Subject: [Freeswitch-users] Paging
In-Reply-To: <CANybnk5eZ3Yqn4-2gJz6hWaDXskgPOs804_v0CV3WgcCqHYB4A@mail.gmail.com>
References: <8AE4D7F5-1AE6-44F8-B85D-13C4FADC22CA@magicmail.mooo.com>
 <CANybnk5eZ3Yqn4-2gJz6hWaDXskgPOs804_v0CV3WgcCqHYB4A@mail.gmail.com>
Message-ID: <CAHKqMb7srMTQrm9Hz8ZNreKPZYuO7K11wJCL0_wD9ZJMyTtR=Q@mail.gmail.com>

I've done this using mod_portaudio with auto-answer from this example:
https://wiki.freeswitch.org/wiki/Mod_portaudio#PA_System_w.2F_Chime

/nandy

On Fri, May 19, 2017 at 12:04 PM, Lesley Pervis <lesley.pervis at gmail.com>
wrote:

> I've done this in Lua with outbound conference calls, but it was pretty
> complicated and you have to have endpoints that will auto-answer.
> https://freeswitch.org/confluence/display/FREESWITCH/
> Outbound+Conference+Calls
>
> On Wed, May 17, 2017 at 10:02 AM, Rick Jarvis <rick at magicmail.mooo.com>
> wrote:
>
>> Looking to set up paging (not multicast). What’s the best way of
>> achieving this? Specifically, I want to have the receiving handset(s)
>> answer muted for privacy reasons, so it’s literally like a PA system rather
>> than just auto answer…?
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.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
>> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170614/91179378/attachment.html>

From rick at magicmail.mooo.com  Wed Jun 14 11:03:00 2017
From: rick at magicmail.mooo.com (Rick Jarvis)
Date: Wed, 14 Jun 2017 12:03:00 +0100
Subject: [Freeswitch-users] Paging
In-Reply-To: <CAHKqMb7srMTQrm9Hz8ZNreKPZYuO7K11wJCL0_wD9ZJMyTtR=Q@mail.gmail.com>
References: <8AE4D7F5-1AE6-44F8-B85D-13C4FADC22CA@magicmail.mooo.com>
 <CANybnk5eZ3Yqn4-2gJz6hWaDXskgPOs804_v0CV3WgcCqHYB4A@mail.gmail.com>
 <CAHKqMb7srMTQrm9Hz8ZNreKPZYuO7K11wJCL0_wD9ZJMyTtR=Q@mail.gmail.com>
Message-ID: <5D4CCF18-DAA0-4F84-8203-4F69D66B251A@magicmail.mooo.com>

Thanks Nandy. We’ve used PA for stuff, but what we’re trying to accomplish here is paging through handsets. It’s kind of strange that there isn’t more support for this as it’s quite a standard feature even on the old analog PBX systems, Panasonic and so on.

> On 14 Jun 2017, at 10:43, Nandy Dagondon <nandy1925 at gmail.com> wrote:
> 
> I've done this using mod_portaudio with auto-answer from this example:
> https://wiki.freeswitch.org/wiki/Mod_portaudio#PA_System_w.2F_Chime <https://wiki.freeswitch.org/wiki/Mod_portaudio#PA_System_w.2F_Chime>
> 
> /nandy
> 
> On Fri, May 19, 2017 at 12:04 PM, Lesley Pervis <lesley.pervis at gmail.com <mailto:lesley.pervis at gmail.com>> wrote:
> I've done this in Lua with outbound conference calls, but it was pretty complicated and you have to have endpoints that will auto-answer. https://freeswitch.org/confluence/display/FREESWITCH/Outbound+Conference+Calls <https://freeswitch.org/confluence/display/FREESWITCH/Outbound+Conference+Calls>
> 
> On Wed, May 17, 2017 at 10:02 AM, Rick Jarvis <rick at magicmail.mooo.com <mailto:rick at magicmail.mooo.com>> wrote:
> Looking to set up paging (not multicast). What’s the best way of achieving this? Specifically, I want to have the receiving handset(s) answer muted for privacy reasons, so it’s literally like a PA system rather than just auto answer…?
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> http://www.cluecon.com <http://www.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>
> http://www.freeswitch.org <http://www.freeswitch.org/>
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> http://www.cluecon.com <http://www.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>
> http://www.freeswitch.org <http://www.freeswitch.org/>
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170614/ab71cc3d/attachment-0001.html>

From shaun.stokes at itec-support.co.uk  Wed Jun 14 11:11:00 2017
From: shaun.stokes at itec-support.co.uk (Shaun Stokes)
Date: Wed, 14 Jun 2017 11:11:00 +0000
Subject: [Freeswitch-users] Add long contact URI support for mod_callcenter
Message-ID: <6FD2F8B5BB72834E9939AEDF9FB802A901E8671F59@mbx-01.sysconfig.co.uk>

Currently we're unable to exceed the 255 character limit on the contact field for mod_callcenter which is required to support long contact URIs for agents in mod_callcenter.

FreeSWITCH and mod_callcenter fully support long contact URIs (up to 510 characters) as tested in our production environment over the last 6 months.

We've submitted a pull request to resolve this problem:
https://freeswitch.org/stash/projects/FS/repos/freeswitch/pull-requests/1165/diff

Any chance this can be committed?
[http://www.itec-support.co.uk/wp-content/uploads/2016/07/email_logo.jpg]
Shaun Stokes - Infrastructure Analyst

T :     01453 700713
E :     shaun.stokes at itec-support.co.uk
W :     www.itec-support.co.uk

Registered Address :- ITEC Support, Suite 2 Prospect House, Bath Road, Stroud, Gloucestershire GL5 3QF
Company No. 06908001

CONFIDENTIALITY NOTICE
This communication and the information it contains are intended for the person or organisation to which it is addressed. Its contents are confidential and may be protected in law. Unauthorised use, copying or disclosure of any of it may be unlawful. If you are not the intended recipient, please contact us immediately.
The contents of any attachments in this e-mail may contain software viruses, which could damage your own computer system. While ITEC Support has taken every reasonable precaution to minimise this risk, we cannot accept liability for any damage which you sustain as a result of software viruses. You should carry out your own virus checking procedure before opening any attachment.

______________________________________________________________________
This message has been checked for all known viruses by  MessageLabs Virus Scanning Service.
______________________________________________________________________

From hardikitpl at gmail.com  Wed Jun 14 11:52:23 2017
From: hardikitpl at gmail.com (Hardik Patel)
Date: Wed, 14 Jun 2017 17:22:23 +0530
Subject: [Freeswitch-users] Fax receive issue with t30 codec
In-Reply-To: <CAHdn4vgRHZ5zNFLQYhDkUfrMGDNzqnEwQdQ3coORHfqQmOMOQA@mail.gmail.com>
References: <CAHdn4vjxr5vVoO7Oc1P0u-xg6dJNYBhjxU=y5iz6VP5_JEcUjA@mail.gmail.com>
 <CAEJMVkDtY8VeG+14cB42wHee5fjgzsXE5trA31dtp=F4rxoeRg@mail.gmail.com>
 <CAJoKKdVBBBbZSmtT20f6ANgDJnpXjV+7a1==ivsdQ5qcaN3BqA@mail.gmail.com>
 <CAEJMVkBFD6b=8porYC9Vm-si5Qy3OsDCNKQrmOBDWmm4A-qTEA@mail.gmail.com>
 <CAHdn4vgRHZ5zNFLQYhDkUfrMGDNzqnEwQdQ3coORHfqQmOMOQA@mail.gmail.com>
Message-ID: <CAHdn4vhNkD9naQ9dOh5NLFEjgcemeyr6PqYU1dAsUMQGmMte9A@mail.gmail.com>

Hi Brian,

Is there any solution for that?

On Sat, Jun 10, 2017 at 9:46 AM, Hardik Patel <hardikitpl at gmail.com> wrote:

> Hi Brian,
>
> Thanks for the support.
>
> We are testing receive fax functionality using real fax machine and here i
> have listed the model which we are using to send fax.
>
> the models of real fax machines that we have used are group 3 CCITT / ITU,
> they are the following:
> 1 ) konica minolta bizhub-c220
> 2 ) HP Officejet 4500
> 3 ) HP Officejet G85
>
>
> From above list one of our machine is sending fax without T38 support and
> we got the failure with same error which have posted on bug but if we use
> T38 support then it works fine for us.
>
>
>
> *ERROR :*
> 2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:508 Fax processing
> not successful - result (3) Timed out waiting for the first message.
> 2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:513 Remote station
> id:
> 2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:514 Local station
> id:  SpanDSP Fax Ident
> 2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:515 Pages
> transferred: 0
> 2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:517 Total fax
> pages:   0
> 2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:518 Image
> resolution:  0x0
> 2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:519 Transfer
> Rate:     14400
> 2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:521 ECM
> status         off
> 2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:522 remote country:
> 2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:523 remote vendor:
> 2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:524 remote model:
> 2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:526
> ==============================
>
> On Sat, Jun 10, 2017 at 2:42 AM, Brian West <brian at freeswitch.org> wrote:
>
>> Your log snip doesn't really help, I know without a single doubt faxing
>> works fine.  So what are you doing and how are you doing it?
>>
>> On Fri, Jun 9, 2017 at 3:42 PM, Tihomir Culjaga <tculjaga at gmail.com>
>> wrote:
>>
>>> hi, whats new on faxing in 1.6.17  ?
>>>
>>> T.
>>>
>>> On 6 June 2017 at 16:00, Brian West <brian at freeswitch.org> wrote:
>>>
>>>> You'll have to use 1.6.17 if you ever want any faxing to work in all
>>>> test cases.
>>>>
>>>> /b
>>>>
>>>> On Tue, Jun 6, 2017 at 6:11 AM, Hardik Patel <hardikitpl at gmail.com>
>>>> wrote:
>>>>
>>>>> Hello,
>>>>>
>>>>> I am using opensips as entry point using dispatcher. opensips(
>>>>> 127.0.0.1), i am routing call to freeswitch server (127.0.0.3).
>>>>>
>>>>> Now I am trying to receive fax, my issue is when i try to send fax in
>>>>> softphone(Zoiper) from the log i am seeing that it is sending fax using t30
>>>>> codec. and i am not receiving the fax at destination, is it because of
>>>>> codec, should it only work with t38 codec? if that is the issue than how am
>>>>> i be able to send the fax using t38 from zoiper?
>>>>>
>>>>> Here i am attaching the fs log with loglevel 9 and sip trace is also
>>>>> enabled.
>>>>>
>>>>> 127.0.0.2  => carrier/provider IP
>>>>> 123456789 => Fax number
>>>>> test at gamil.com => Email Address
>>>>> 127.0.0.4 =>UI IP
>>>>>
>>>>> Pastebin link:https://pastebin.freeswitch.org/view/9ec52715
>>>>>
>>>>> ____________________________________________________________
>>>>> _____________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.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/free
>>>>> switch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>>
>>>> *Brian West*
>>>> brian at freeswitch.org
>>>>
>>>> *Twitter: @FreeSWITCH , @briankwest*
>>>>
>>>> http://www.freeswitchbook.com
>>>> http://www.freeswitchcookbook.com
>>>>
>>>> Book a phone call (CST) <https://freeswitch.com/appointment>
>>>>
>>>> Allison prompts for FreeSWITCH:
>>>>
>>>> *https://www.gofundme.com/allison-prompts-for-freeswitch*
>>>> <https://www.gofundme.com/allison-prompts-for-freeswitch>
>>>>
>>>> Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
>>>> /r/freeswitch <https://www.reddit.com/r/freeswitch>
>>>>
>>>> *T:*+19184209001 <+1%20918-420-9001> | *F:*+19184209002
>>>> <+1%20918-420-9002> | *M:*+1918424WEST (9378)
>>>> *Skype:*briankwest
>>>>
>>>> ____________________________________________________________
>>>> _____________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.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/free
>>>> switch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> ____________________________________________________________
>>> _____________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.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
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>>
>> *Brian West*
>> brian at freeswitch.org
>>
>> *Twitter: @FreeSWITCH , @briankwest*
>>
>> http://www.freeswitchbook.com
>> http://www.freeswitchcookbook.com
>>
>> Book a phone call (CST) <https://freeswitch.com/appointment>
>>
>> Allison prompts for FreeSWITCH:
>>
>> *https://www.gofundme.com/allison-prompts-for-freeswitch*
>> <https://www.gofundme.com/allison-prompts-for-freeswitch>
>>
>> Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
>> /r/freeswitch <https://www.reddit.com/r/freeswitch>
>>
>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
>> *Skype:*briankwest
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.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
>> http://www.freeswitch.org
>>
>
>
>
> --
> Hardik Patel
> iNextrix Technologies Pvt Ltd
>



-- 
Hardik Patel
iNextrix Technologies Pvt Ltd
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170614/df6fd808/attachment-0001.html>

From gmaruzz at gmail.com  Wed Jun 14 11:58:25 2017
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Wed, 14 Jun 2017 13:58:25 +0200
Subject: [Freeswitch-users] Fax receive issue with t30 codec
In-Reply-To: <CAHdn4vhNkD9naQ9dOh5NLFEjgcemeyr6PqYU1dAsUMQGmMte9A@mail.gmail.com>
References: <CAHdn4vjxr5vVoO7Oc1P0u-xg6dJNYBhjxU=y5iz6VP5_JEcUjA@mail.gmail.com>
 <CAEJMVkDtY8VeG+14cB42wHee5fjgzsXE5trA31dtp=F4rxoeRg@mail.gmail.com>
 <CAJoKKdVBBBbZSmtT20f6ANgDJnpXjV+7a1==ivsdQ5qcaN3BqA@mail.gmail.com>
 <CAEJMVkBFD6b=8porYC9Vm-si5Qy3OsDCNKQrmOBDWmm4A-qTEA@mail.gmail.com>
 <CAHdn4vgRHZ5zNFLQYhDkUfrMGDNzqnEwQdQ3coORHfqQmOMOQA@mail.gmail.com>
 <CAHdn4vhNkD9naQ9dOh5NLFEjgcemeyr6PqYU1dAsUMQGmMte9A@mail.gmail.com>
Message-ID: <CALXCt0pPS1MnPOFpvENhb0T646E7zUeTg3UzT+Kzs8Q=8tf-vA@mail.gmail.com>

try setting the transfer to 9600 (fax_disable_v17)

Also, you may find this page useful:

https://freeswitch.org/confluence/display/FREESWITCH/mod_spandsp#mod_spandsp-Fax



On 14 June 2017 at 13:52, Hardik Patel <hardikitpl at gmail.com> wrote:

> Hi Brian,
>
> Is there any solution for that?
>
> On Sat, Jun 10, 2017 at 9:46 AM, Hardik Patel <hardikitpl at gmail.com>
> wrote:
>
>> Hi Brian,
>>
>> Thanks for the support.
>>
>> We are testing receive fax functionality using real fax machine and here
>> i have listed the model which we are using to send fax.
>>
>> the models of real fax machines that we have used are group 3 CCITT /
>> ITU, they are the following:
>> 1 ) konica minolta bizhub-c220
>> 2 ) HP Officejet 4500
>> 3 ) HP Officejet G85
>>
>>
>> From above list one of our machine is sending fax without T38 support and
>> we got the failure with same error which have posted on bug but if we use
>> T38 support then it works fine for us.
>>
>>
>>
>> *ERROR :*
>> 2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:508 Fax processing
>> not successful - result (3) Timed out waiting for the first message.
>> 2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:513 Remote station
>> id:
>> 2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:514 Local station
>> id:  SpanDSP Fax Ident
>> 2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:515 Pages
>> transferred: 0
>> 2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:517 Total fax
>> pages:   0
>> 2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:518 Image
>> resolution:  0x0
>> 2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:519 Transfer
>> Rate:     14400
>> 2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:521 ECM
>> status         off
>> 2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:522 remote
>> country:
>> 2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:523 remote
>> vendor:
>> 2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:524 remote
>> model:
>> 2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:526
>> ==============================
>>
>> On Sat, Jun 10, 2017 at 2:42 AM, Brian West <brian at freeswitch.org> wrote:
>>
>>> Your log snip doesn't really help, I know without a single doubt faxing
>>> works fine.  So what are you doing and how are you doing it?
>>>
>>> On Fri, Jun 9, 2017 at 3:42 PM, Tihomir Culjaga <tculjaga at gmail.com>
>>> wrote:
>>>
>>>> hi, whats new on faxing in 1.6.17  ?
>>>>
>>>> T.
>>>>
>>>> On 6 June 2017 at 16:00, Brian West <brian at freeswitch.org> wrote:
>>>>
>>>>> You'll have to use 1.6.17 if you ever want any faxing to work in all
>>>>> test cases.
>>>>>
>>>>> /b
>>>>>
>>>>> On Tue, Jun 6, 2017 at 6:11 AM, Hardik Patel <hardikitpl at gmail.com>
>>>>> wrote:
>>>>>
>>>>>> Hello,
>>>>>>
>>>>>> I am using opensips as entry point using dispatcher. opensips(
>>>>>> 127.0.0.1), i am routing call to freeswitch server (127.0.0.3).
>>>>>>
>>>>>> Now I am trying to receive fax, my issue is when i try to send fax in
>>>>>> softphone(Zoiper) from the log i am seeing that it is sending fax using t30
>>>>>> codec. and i am not receiving the fax at destination, is it because of
>>>>>> codec, should it only work with t38 codec? if that is the issue than how am
>>>>>> i be able to send the fax using t38 from zoiper?
>>>>>>
>>>>>> Here i am attaching the fs log with loglevel 9 and sip trace is also
>>>>>> enabled.
>>>>>>
>>>>>> 127.0.0.2  => carrier/provider IP
>>>>>> 123456789 => Fax number
>>>>>> test at gamil.com => Email Address
>>>>>> 127.0.0.4 =>UI IP
>>>>>>
>>>>>> Pastebin link:https://pastebin.freeswitch.org/view/9ec52715
>>>>>>
>>>>>> ____________________________________________________________
>>>>>> _____________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://confluence.freeswitch.org
>>>>>> http://www.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/free
>>>>>> switch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>>
>>>>> --
>>>>>
>>>>> *Brian West*
>>>>> brian at freeswitch.org
>>>>>
>>>>> *Twitter: @FreeSWITCH , @briankwest*
>>>>>
>>>>> http://www.freeswitchbook.com
>>>>> http://www.freeswitchcookbook.com
>>>>>
>>>>> Book a phone call (CST) <https://freeswitch.com/appointment>
>>>>>
>>>>> Allison prompts for FreeSWITCH:
>>>>>
>>>>> *https://www.gofundme.com/allison-prompts-for-freeswitch*
>>>>> <https://www.gofundme.com/allison-prompts-for-freeswitch>
>>>>>
>>>>> Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
>>>>> /r/freeswitch <https://www.reddit.com/r/freeswitch>
>>>>>
>>>>> *T:*+19184209001 <+1%20918-420-9001> | *F:*+19184209002
>>>>> <+1%20918-420-9002> | *M:*+1918424WEST (9378)
>>>>> *Skype:*briankwest
>>>>>
>>>>> ____________________________________________________________
>>>>> _____________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.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/free
>>>>> switch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>> ____________________________________________________________
>>>> _____________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.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/free
>>>> switch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> --
>>>
>>> *Brian West*
>>> brian at freeswitch.org
>>>
>>> *Twitter: @FreeSWITCH , @briankwest*
>>>
>>> http://www.freeswitchbook.com
>>> http://www.freeswitchcookbook.com
>>>
>>> Book a phone call (CST) <https://freeswitch.com/appointment>
>>>
>>> Allison prompts for FreeSWITCH:
>>>
>>> *https://www.gofundme.com/allison-prompts-for-freeswitch*
>>> <https://www.gofundme.com/allison-prompts-for-freeswitch>
>>>
>>> Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
>>> /r/freeswitch <https://www.reddit.com/r/freeswitch>
>>>
>>> *T:*+19184209001 <(918)%20420-9001> | *F:*+19184209002
>>> <(918)%20420-9002> | *M:*+1918424WEST (9378)
>>> *Skype:*briankwest
>>>
>>> ____________________________________________________________
>>> _____________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.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
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Hardik Patel
>> iNextrix Technologies Pvt Ltd
>>
>
>
>
> --
> Hardik Patel
> iNextrix Technologies Pvt Ltd
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org
>



-- 

Sincerely,

Giovanni Maruzzelli
OpenTelecom.IT
cell: +39 347 266 56 18
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170614/8e7b615a/attachment-0001.html>

From italo at freeswitch.org  Wed Jun 14 12:11:48 2017
From: italo at freeswitch.org (=?UTF-8?B?w410YWxvIFJvc3Np?=)
Date: Wed, 14 Jun 2017 09:11:48 -0300
Subject: [Freeswitch-users] Add long contact URI support for
	mod_callcenter
In-Reply-To: <6FD2F8B5BB72834E9939AEDF9FB802A901E8671F59@mbx-01.sysconfig.co.uk>
References: <6FD2F8B5BB72834E9939AEDF9FB802A901E8671F59@mbx-01.sysconfig.co.uk>
Message-ID: <CABMX_w_zBSHkmr0zX+5ZibjJn7pzowiJaEajS3rdnmEyQrYsEg@mail.gmail.com>

Sure, I'll merge it later today. Thank you.

On Wed, Jun 14, 2017 at 8:11 AM, Shaun Stokes <
shaun.stokes at itec-support.co.uk> wrote:

> Currently we're unable to exceed the 255 character limit on the contact
> field for mod_callcenter which is required to support long contact URIs for
> agents in mod_callcenter.
>
> FreeSWITCH and mod_callcenter fully support long contact URIs (up to 510
> characters) as tested in our production environment over the last 6 months.
>
> We've submitted a pull request to resolve this problem:
> https://freeswitch.org/stash/projects/FS/repos/freeswitch/
> pull-requests/1165/diff
>
> Any chance this can be committed?
> [http://www.itec-support.co.uk/wp-content/uploads/2016/07/email_logo.jpg]
> Shaun Stokes - Infrastructure Analyst
>
> T :     01453 700713
> E :     shaun.stokes at itec-support.co.uk
> W :     www.itec-support.co.uk
>
> Registered Address :- ITEC Support, Suite 2 Prospect House, Bath Road,
> Stroud, Gloucestershire GL5 3QF
> Company No. 06908001
>
> CONFIDENTIALITY NOTICE
> This communication and the information it contains are intended for the
> person or organisation to which it is addressed. Its contents are
> confidential and may be protected in law. Unauthorised use, copying or
> disclosure of any of it may be unlawful. If you are not the intended
> recipient, please contact us immediately.
> The contents of any attachments in this e-mail may contain software
> viruses, which could damage your own computer system. While ITEC Support
> has taken every reasonable precaution to minimise this risk, we cannot
> accept liability for any damage which you sustain as a result of software
> viruses. You should carry out your own virus checking procedure before
> opening any attachment.
>
> ______________________________________________________________________
> This message has been checked for all known viruses by  MessageLabs Virus
> Scanning Service.
> ______________________________________________________________________
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org




-- 
Ítalo Rossi
italo at freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170614/4dba35a8/attachment.html>

From zzyroy at qq.com  Wed Jun 14 00:57:40 2017
From: zzyroy at qq.com (=?gb18030?B?enp5?=)
Date: Wed, 14 Jun 2017 08:57:40 +0800
Subject: [Freeswitch-users] How to use originate make A_leg to a queue
Message-ID: <tencent_0E95D3795A95A2624FB35A7D@qq.com>

Dear David,
Thank you for your reply.  This way is put the call center queue in the B leg.  
But I want to call the queue first (put the queue in the A leg).  




------------------ Original ------------------
From: David Villasmil <david.villasmil.work at gmail.com>
Date: 周三,6月 14,2017 03:36
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] How to use originate make A_leg to a queue



Originate blahnlah &callcenter(yourqueue)
Or something like that
On Tue, Jun 13, 2017 at 3:30 PM zzy <zzyroy at qq.com> wrote:


Dear All,

I'm testing mod_callcenter now.

Is there any way to make A_leg to the call center queue first then bridge B_leg by dialplan?

Just like [ originate group/sales+A 8888 XML default ]

Thanks. 

 
_________________________________________________________________________
 Professional FreeSWITCH Consulting Services:
 consulting at freeswitch.org
 http://www.freeswitchsolutions.com
 
 Official FreeSWITCH Sites
 http://www.freeswitch.org
 http://confluence.freeswitch.org
 http://www.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
 http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170614/880eaef1/attachment-0001.html>

From sdpaek21 at gmail.com  Wed Jun 14 10:30:54 2017
From: sdpaek21 at gmail.com (Sp Pho)
Date: Wed, 14 Jun 2017 13:30:54 +0300
Subject: [Freeswitch-users] B-Leg Early Media - Normal Clearing vs
	Originator Cancelled
In-Reply-To: <CAL0o4X=G8iikibMhyaW4c71AJhvc6EPgNss1LpjXpiL=t-9BJw@mail.gmail.com>
References: <CAL0o4X=G8iikibMhyaW4c71AJhvc6EPgNss1LpjXpiL=t-9BJw@mail.gmail.com>
Message-ID: <CAL0o4X=mQGrxw4nWrL17ZprbyjBLviBYov3WAwMjHp2sYOBr0A@mail.gmail.com>

I've pulled a log of the bad case vs good one. A diff of both can be found
here:
https://www.diffchecker.com/zWNNGibE.

This is from the same originating number  - any ideas why on line 96, early
media is detected for the good case and not the bad, would be greatly
appreciated.

thanks in advance

On Tue, Jun 13, 2017 at 3:59 PM, Sp Pho <sdpaek21 at gmail.com> wrote:

> Hi,
>
> I'm creating a bridge to terminate call arriving inbound from a DID.
>
> Everything works fine, however the termination point plays a recorded
> message - which as rightfully so by the switch is seen as early media (with
> pre-answer).
>
> However, when the originator hangs up while the early media is playing,
> the CDRs indicate NORMAL_CLEARING. If ignore early media is enabled, the
> media does not play and ORIGINATOR_CANCELis the hang-up cause.
>
> Is there any way to receive the early media and from the CDR know whether
> or not the call was subsequently answered or not by the agent (i.e. if the
> originator canceled the call during the early media or while on hold)?
>
> Thanks in advance,
> SP
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170614/4c800daa/attachment-0001.html>

From khamlichi.khalil at gmail.com  Wed Jun 14 12:09:34 2017
From: khamlichi.khalil at gmail.com (Khalil Khamlichi)
Date: Wed, 14 Jun 2017 12:09:34 +0000
Subject: [Freeswitch-users] Callcenter module,
 can I originate call for an agent in uuid-standby mode ?
Message-ID: <CAEK98WarT_sPVc4HHZoVJqo+BEpGDduwH87w-Ar3C4NphMEwwg@mail.gmail.com>

Hi,

I need to give my agents ability to make manual calls, hopefully without
leaving their actually established call (they are in uuid-standby mode and
in Idle state so there is no live member on the line ).

my questions:

Is it possible to originate a new call and bridge with agent uuid-standby
session ?
would it not break the callcenter establised uuid-standby session ?
would the agent return to its uuid-standby session after the originated
call is hangup ?

ofcourse if this is too complicated, I would just connect the agent thru a
second line, while leaving his uuid-standby call on the first line, though
it would be so cool to somehow stay on that same uuid-standby session and
enjoy both calllcenter module and manual dialing.

Thanks in advance, and I appreciate your help.

Khalil
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170614/175be7a1/attachment-0001.html>

From khorsmann at gmail.com  Wed Jun 14 14:02:50 2017
From: khorsmann at gmail.com (Karsten Horsmann)
Date: Wed, 14 Jun 2017 16:02:50 +0200
Subject: [Freeswitch-users] [SR-Users] Multiple FreeSWITCH servers
	behind kamailio-websocket
In-Reply-To: <407012cb-dc25-b237-17d9-69462307e05b@gmail.com>
References: <CAFArqsZjjVP63_OscW=qVOh_skaddndKuXzKaWJp2WgHg9KaEw@mail.gmail.com>
 <CAMtXxQkrCf0+iv9-5e0VtQFN-wuAXAPcbboSV4FfPwraAmFDFg@mail.gmail.com>
 <407012cb-dc25-b237-17d9-69462307e05b@gmail.com>
Message-ID: <CAFArqsY8WYdEvibM57SXSqLbnMWBRcCmkEc0MOgHYk_6_G-Cdg@mail.gmail.com>

Hello Daniel,

i will try that and hopfully get an working webrtc loadbalancer in the near
future with Kamailio and FreeSWITCH :).

2017-06-14 10:57 GMT+02:00 Daniel-Constantin Mierla <miconda at gmail.com>:

> Hello,
>
> that combination of dispatcher+websocket+xhttp modules works just fine...
>
> So to load balance the SIP signaling with Kamailio towards FreeSwitch,
> just use the dispatcher module as usual. A sample config is available at:
>
>   - https://www.kamailio.org/docs/modules/stable/modules/
> dispatcher.html#dispatcher.ex.config
>
> You need to add the support for websocket traffic via websocket module:
>
>   - https://www.kamailio.org/docs/modules/stable/modules/
> websocket.html#idp42826164
>
> or extract from the tutorial linked in a previous email on this thread.
>
> Cheers,
> Daniel
>
>
> On 13.06.17 16:38, Joel Serrano wrote:
>
> Hi Karsten,
>
> Have you tried with regular Kamailio (w/ dispatcher+websocket+xhttp
> modules)? I don't see why it wouldn't work...
>
> Joel.
>
>
> On Wed, Jun 7, 2017 at 4:47 AM, Karsten Horsmann <khorsmann at gmail.com>
> wrote:
>
>> Hello List,
>>
>>
>> is there any howto about webrtc loadbalance in combination with kamailio
>> and FreeSWITCH?
>>
>> I want to share one WSS address/endpoint to multiple FreeSWITCH backends.
>> Or is there any other best practice?
>>
>> My callflow is mostly that my internal SIP Servers called my registered
>> webrtc clients.
>>
>> Would be nice to get some input.
>>
>> --
>> Kind Regards
>> *Karsten Horsmann*
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.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
>> http://www.freeswitch.org
>>
>
>
>
> _______________________________________________
> Kamailio (SER) - Users Mailing Listsr-users at lists.kamailio.orghttps://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>
>
>
> --
> Daniel-Constantin Mierlawww.twitter.com/miconda -- www.linkedin.com/in/miconda
> Kamailio Advanced Training - www.asipto.com
> Kamailio World Conference - www.kamailioworld.com
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org
>



-- 
Mit freundlichen Grüßen
*Karsten Horsmann*
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170614/6746f054/attachment.html>

From brian at freeswitch.org  Wed Jun 14 16:26:17 2017
From: brian at freeswitch.org (Brian West)
Date: Wed, 14 Jun 2017 11:26:17 -0500
Subject: [Freeswitch-users] B-Leg Early Media - Normal Clearing vs
 Originator Cancelled
In-Reply-To: <CAL0o4X=mQGrxw4nWrL17ZprbyjBLviBYov3WAwMjHp2sYOBr0A@mail.gmail.com>
References: <CAL0o4X=G8iikibMhyaW4c71AJhvc6EPgNss1LpjXpiL=t-9BJw@mail.gmail.com>
 <CAL0o4X=mQGrxw4nWrL17ZprbyjBLviBYov3WAwMjHp2sYOBr0A@mail.gmail.com>
Message-ID: <CAEJMVkCfOqLm9BXJ2PB6jqTjJgEuBAmOMK_XGvEqfs74JNsUMg@mail.gmail.com>

You're receiving a 487 (timeout) response.

/b


On Wed, Jun 14, 2017 at 5:30 AM, Sp Pho <sdpaek21 at gmail.com> wrote:

> I've pulled a log of the bad case vs good one. A diff of both can be found
> here:
> https://www.diffchecker.com/zWNNGibE.
>
> This is from the same originating number  - any ideas why on line 96,
> early media is detected for the good case and not the bad, would be greatly
> appreciated.
>
> thanks in advance
>
>
> On Tue, Jun 13, 2017 at 3:59 PM, Sp Pho <sdpaek21 at gmail.com> wrote:
>
>> Hi,
>>
>> I'm creating a bridge to terminate call arriving inbound from a DID.
>>
>> Everything works fine, however the termination point plays a recorded
>> message - which as rightfully so by the switch is seen as early media (with
>> pre-answer).
>>
>> However, when the originator hangs up while the early media is playing,
>> the CDRs indicate NORMAL_CLEARING. If ignore early media is enabled, the
>> media does not play and ORIGINATOR_CANCELis the hang-up cause.
>>
>> Is there any way to receive the early media and from the CDR know whether
>> or not the call was subsequently answered or not by the agent (i.e. if the
>> originator canceled the call during the early media or while on hold)?
>>
>> Thanks in advance,
>> SP
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org
>



-- 

*Brian West*
brian at freeswitch.org

*Twitter: @FreeSWITCH , @briankwest*

http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

Book a phone call (CST) <https://freeswitch.com/appointment>

Allison prompts for FreeSWITCH:

*https://www.gofundme.com/allison-prompts-for-freeswitch*
<https://www.gofundme.com/allison-prompts-for-freeswitch>

Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
/r/freeswitch <https://www.reddit.com/r/freeswitch>

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*Skype:*briankwest
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170614/78ec63c5/attachment.html>

From brian at freeswitch.org  Wed Jun 14 16:27:35 2017
From: brian at freeswitch.org (Brian West)
Date: Wed, 14 Jun 2017 11:27:35 -0500
Subject: [Freeswitch-users] Paging
In-Reply-To: <5D4CCF18-DAA0-4F84-8203-4F69D66B251A@magicmail.mooo.com>
References: <8AE4D7F5-1AE6-44F8-B85D-13C4FADC22CA@magicmail.mooo.com>
 <CANybnk5eZ3Yqn4-2gJz6hWaDXskgPOs804_v0CV3WgcCqHYB4A@mail.gmail.com>
 <CAHKqMb7srMTQrm9Hz8ZNreKPZYuO7K11wJCL0_wD9ZJMyTtR=Q@mail.gmail.com>
 <5D4CCF18-DAA0-4F84-8203-4F69D66B251A@magicmail.mooo.com>
Message-ID: <CAEJMVkC0y=W7p4mFumdjHGYqvfxxUk8CJELgDv7-z2SYKUEYRQ@mail.gmail.com>

Without multicast you'll have to review the mad boss example in the vanilla
config.  Doing so in this matter doesn't scale well.

./b

On Wed, Jun 14, 2017 at 6:03 AM, Rick Jarvis <rick at magicmail.mooo.com>
wrote:

> Thanks Nandy. We’ve used PA for stuff, but what we’re trying to accomplish
> here is paging through handsets. It’s kind of strange that there isn’t more
> support for this as it’s quite a standard feature even on the old analog
> PBX systems, Panasonic and so on.
>
>
> On 14 Jun 2017, at 10:43, Nandy Dagondon <nandy1925 at gmail.com> wrote:
>
> I've done this using mod_portaudio with auto-answer from this example:
> https://wiki.freeswitch.org/wiki/Mod_portaudio#PA_System_w.2F_Chime
>
> /nandy
>
> On Fri, May 19, 2017 at 12:04 PM, Lesley Pervis <lesley.pervis at gmail.com>
> wrote:
>
>> I've done this in Lua with outbound conference calls, but it was pretty
>> complicated and you have to have endpoints that will auto-answer.
>> https://freeswitch.org/confluence/display/FREESWITCH/Outboun
>> d+Conference+Calls
>>
>> On Wed, May 17, 2017 at 10:02 AM, Rick Jarvis <rick at magicmail.mooo.com>
>> wrote:
>>
>>> Looking to set up paging (not multicast). What’s the best way of
>>> achieving this? Specifically, I want to have the receiving handset(s)
>>> answer muted for privacy reasons, so it’s literally like a PA system rather
>>> than just auto answer…?
>>>
>>>
>>> ____________________________________________________________
>>> _____________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.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
>>> http://www.freeswitch.org
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.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
>> http://www.freeswitch.org
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org
>



-- 

*Brian West*
brian at freeswitch.org

*Twitter: @FreeSWITCH , @briankwest*

http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

Book a phone call (CST) <https://freeswitch.com/appointment>

Allison prompts for FreeSWITCH:

*https://www.gofundme.com/allison-prompts-for-freeswitch*
<https://www.gofundme.com/allison-prompts-for-freeswitch>

Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
/r/freeswitch <https://www.reddit.com/r/freeswitch>

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*Skype:*briankwest
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170614/daf9b25a/attachment-0001.html>

From admin at tedssupply.com  Wed Jun 14 16:36:04 2017
From: admin at tedssupply.com (admin)
Date: Wed, 14 Jun 2017 12:36:04 -0400
Subject: [Freeswitch-users] When calling out ,
 freeswitch doesn't get DTMF ?
In-Reply-To: <CALXCt0qw4DCJ2Q0cF+kVEOpjCjp2T8eFgWbYx03sN=z5APBGGQ@mail.gmail.com>
References: <f823cdd8-a354-3698-d8c5-484b1b4cc8b4@xbipin.com>
 <CAEJMVkDDQX=Ni3AOQxQOo=joGoAeYroxioVP=cu9w6Y2m7kLPA@mail.gmail.com>
 <15c898b5d58.2779.b07ebdf329620b8089087c7205b03f01@xbipin.com>
 <310cf71c.8fc3.15c8ba9c4f6.Coremail.eastour@163.com>
 <CAFGRPVqFA+tpaOQkAGGCR7s5gY0z0TbdArWFdkfnsaaG4Z9Dgw@mail.gmail.com>
 <2a18422f.ef8.15c99b7b9bb.Coremail.eastour@163.com>
 <CALXCt0pKKEJoSeXXZ=tn04nifEZvQCRSFCDKL8jSu+pfOgvRNQ@mail.gmail.com>
 <6c11dd0f.c2c1.15c9c727dc7.Coremail.eastour@163.com>
 <CAFLo2QwwwE6xeXBN9DkM1VEecEJpwnDMmQ7iT9EvSyx9RqE9mg@mail.gmail.com>
 <2f48505d.d9c5.15ca1d28424.Coremail.eastour@163.com>
 <59401B92020000310000A34D@mail.tedssupply.com>
 <CALXCt0qw4DCJ2Q0cF+kVEOpjCjp2T8eFgWbYx03sN=z5APBGGQ@mail.gmail.com>
Message-ID: <59412DB4020000310000A381@mail.tedssupply.com>


I don't mean to hijack the OP concern with my problem, I just wanted to reinforce I have seen this problem.  And the problem is that the call originated through FS appears not to send user entered DTMF to the receiving phone.  Let's help chenyzhi and then I'll take a turn. 

- James

>>> Giovanni Maruzzelli <gmaruzz at gmail.com> 6/14/2017 03:53 AM >>>

Never heard such problems 


Please pastebin your dialplan, your SIP profile, and the complete, since beginning to end, unedited, debug output of console when receiving a call which does not get DTMFs 







sent from mobile
cell: +39 347 266 56 18
Giovanni Maruzzelli
OpenTelecom.IT 



On Jun 13, 2017 23:07, "admin" <admin at tedssupply.com> wrote:



I am encountering the same issue. I am using an ESL call to api originate, with little else except getting in and out of the port, and the caller DTMF is failing to reach the called number. The called number auto answer attendant does not respond to DTMF, and a call to a test phone confirms no DTMF. This is a new upgrade from 1.2 to 1.6 and I don't know if this was an issue in 1.2, but my users never complained before 1.6 upgrade. Ideas?... 



>>> chenyzhi <eastour at 163.com> 06/13/17 3:32 PM >>>

It's a one leg call .there is no b-leg. 



please make a test on your freeswitch box . 



just type the command "originate user/1001 5000" on the freeswitch console to see if your freeswitch instance can detect dtmf input. 









At 2017-06-12 22:57:17, "Luis Daniel Lucio Quiroz" <luis.daniel.lucio at gmail.com> wrote:



check if you have transcoding, and if you do, check that dftm type-codec on leg b are compatible. 



-- 

Luis Daniel Lucio Quiroz
CISSP, CISM, CISA
Linux, VoIP and much more fun
www.okay.com.mx

Need LCR? Check out LCR for FusionPBX with FreeSWITCH
Need Billing? Check out Billing for FusionPBX with FreeSWITCH 



On Mon, Jun 12, 2017 at 9:16 AM, chenyzhi <eastour at 163.com> wrote:



I don't think It's the DTMF type , because when I dial 5000 from x-lite (which has registered to freeswitch as 1001) ,I can hear the voice and if I press any dtmf on x-lite, freeswitch can recieve the dtmfs. This means that the DTMF type is correct ,otherwise freeswitch coudn't have received the dtmfs; 



when I enter the command "originate user/1001 5000" at the freeswitch console , my xlite will ring ,and I answered ,I can hear the voice ,I press some dtmf,but freeswitch can NOT receive any dtmf. really weird. 









At 2017-06-12 17:37:49, "Giovanni Maruzzelli" <gmaruzz at gmail.com> wrote:



you need to check the DTMF type, you probably are using the wrong one (info-inband-rfc2833), and for some reason they are not negotiated



On 12 June 2017 at 02:32, chenyzhi <eastour at 163.com> wrote:



I have read the logs ,but I didn't find any difference. 



Please make a test to see if this happens in your box. 









At 2017-06-10 06:49:34, "David Villasmil" <david.villasmil.work at gmail.com> wrote:



Have you looked at the log? Bump the logging up and see what shows up... what you're seeing is very weird

David

On Fri, Jun 9, 2017 at 11:18 AM chenyzhi <eastour at 163.com> wrote:





Hi 



call 5000 from x-lite , you can hear the IVR voice and if you press dtmf keys , freeswitch can receive the dtmf keys . 



but , if you enter the command "originate user/1001 5000" , x-lite will ring ,answer it ,you can hear the IVR voice , press some keys ,the freeswitch can NOT receive any dtmf , 



why? 
 





_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.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
http://www.freeswitch.org  
 






_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.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
http://www.freeswitch.org
 




--


Sincerely,

Giovanni Maruzzelli
OpenTelecom.IT
cell: +39 347 266 56 18
 
 






_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.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
http://www.freeswitch.org
 


 






_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.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
http://www.freeswitch.org
 
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170614/7be27ca6/attachment-0001.html>

From david.villasmil.work at gmail.com  Wed Jun 14 16:41:30 2017
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Wed, 14 Jun 2017 16:41:30 +0000
Subject: [Freeswitch-users] When calling out ,
	freeswitch doesn't get DTMF ?
In-Reply-To: <59412DB4020000310000A381@mail.tedssupply.com>
References: <f823cdd8-a354-3698-d8c5-484b1b4cc8b4@xbipin.com>
 <CAEJMVkDDQX=Ni3AOQxQOo=joGoAeYroxioVP=cu9w6Y2m7kLPA@mail.gmail.com>
 <15c898b5d58.2779.b07ebdf329620b8089087c7205b03f01@xbipin.com>
 <310cf71c.8fc3.15c8ba9c4f6.Coremail.eastour@163.com>
 <CAFGRPVqFA+tpaOQkAGGCR7s5gY0z0TbdArWFdkfnsaaG4Z9Dgw@mail.gmail.com>
 <2a18422f.ef8.15c99b7b9bb.Coremail.eastour@163.com>
 <CALXCt0pKKEJoSeXXZ=tn04nifEZvQCRSFCDKL8jSu+pfOgvRNQ@mail.gmail.com>
 <6c11dd0f.c2c1.15c9c727dc7.Coremail.eastour@163.com>
 <CAFLo2QwwwE6xeXBN9DkM1VEecEJpwnDMmQ7iT9EvSyx9RqE9mg@mail.gmail.com>
 <2f48505d.d9c5.15ca1d28424.Coremail.eastour@163.com>
 <59401B92020000310000A34D@mail.tedssupply.com>
 <CALXCt0qw4DCJ2Q0cF+kVEOpjCjp2T8eFgWbYx03sN=z5APBGGQ@mail.gmail.com>
 <59412DB4020000310000A381@mail.tedssupply.com>
Message-ID: <CAFGRPVpX9-M2n0FfebV3S1TTEWGyXS3Wh9NqzH7qC9nqO9csWw@mail.gmail.com>

I do this on a daily basis, originate and send to an ivr to get dtmf
digits, no issues ever found...
On Wed, Jun 14, 2017 at 6:36 PM admin <admin at tedssupply.com> wrote:

> I don't mean to hijack the OP concern with my problem, I just wanted to
> reinforce I have seen this problem.  And the problem is that the call
> originated through FS appears not to send user entered DTMF to the
> receiving phone.  Let's help chenyzhi and then I'll take a turn.
>
> - James
>
> >>> Giovanni Maruzzelli <gmaruzz at gmail.com> 6/14/2017 03:53 AM >>>
>
> Never heard such problems
>
>
> Please pastebin your dialplan, your SIP profile, and the complete, since
> beginning to end, unedited, debug output of console when receiving a call
> which does not get DTMFs
>
>
>
>
> sent from mobile
> cell: +39 347 266 56 18
> Giovanni Maruzzelli
> OpenTelecom.IT
>
>
> On Jun 13, 2017 23:07, "admin" <admin at tedssupply.com> wrote:
>
>> I am encountering the same issue. I am using an ESL call to api
>> originate, with little else except getting in and out of the port, and the
>> caller DTMF is failing to reach the called number. The called number auto
>> answer attendant does not respond to DTMF, and a call to a test phone
>> confirms no DTMF. This is a new upgrade from 1.2 to 1.6 and I don't know if
>> this was an issue in 1.2, but my users never complained before 1.6 upgrade.
>> Ideas?...
>>
>>
>>
>> >>> chenyzhi <eastour at 163.com> 06/13/17 3:32 PM >>>
>>
>> It's a one leg call .there is no b-leg.
>>
>>
>> please make a test on your freeswitch box .
>>
>>
>> just type the command "originate user/1001 5000" on the freeswitch
>> console to see if your freeswitch instance can detect dtmf input.
>>
>>
>>
>>
>>
>>
>>
>>
>> At 2017-06-12 22:57:17, "Luis Daniel Lucio Quiroz" <
>> luis.daniel.lucio at gmail.com> wrote:
>>
>> check if you have transcoding, and if you do, check that dftm type-codec
>> on leg b are compatible.
>>
>>
>> --
>>
>> Luis Daniel Lucio Quiroz
>> CISSP, CISM, CISA
>> Linux, VoIP and much more fun
>> www.okay.com.mx
>>
>> Need LCR? Check out LCR for FusionPBX with FreeSWITCH
>> Need Billing? Check out Billing for FusionPBX with FreeSWITCH
>>
>>
>> On Mon, Jun 12, 2017 at 9:16 AM, chenyzhi <eastour at 163.com> wrote:
>>
>>> I don't think It's the DTMF type , because when I dial 5000 from x-lite
>>> (which has registered to freeswitch as 1001) ,I can hear the voice and if I
>>> press any dtmf on x-lite, freeswitch can recieve the dtmfs. This means that
>>> the DTMF type is correct ,otherwise freeswitch coudn't have received the
>>> dtmfs;
>>>
>>>
>>> when I enter the command "originate user/1001 5000" at the freeswitch
>>> console , my xlite will ring ,and I answered ,I can hear the voice ,I press
>>> some dtmf,but freeswitch can NOT receive any dtmf. really weird.
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>> At 2017-06-12 17:37:49, "Giovanni Maruzzelli" <gmaruzz at gmail.com> wrote:
>>>
>>> you need to check the DTMF type, you probably are using the wrong one
>>> (info-inband-rfc2833), and for some reason they are not negotiated
>>>
>>>
>>> On 12 June 2017 at 02:32, chenyzhi <eastour at 163.com> wrote:
>>>
>>>> I have read the logs ,but I didn't find any difference.
>>>>
>>>>
>>>> Please make a test to see if this happens in your box.
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> At 2017-06-10 06:49:34, "David Villasmil" <
>>>> david.villasmil.work at gmail.com> wrote:
>>>>
>>>> Have you looked at the log? Bump the logging up and see what shows
>>>> up... what you're seeing is very weird
>>>>
>>>> David
>>>>
>>>> On Fri, Jun 9, 2017 at 11:18 AM chenyzhi <eastour at 163.com> wrote:
>>>>
>>>>> Hi
>>>>>
>>>>>
>>>>> call 5000 from x-lite , you can hear the IVR voice and if you press
>>>>> dtmf keys , freeswitch can receive the dtmf keys .
>>>>>
>>>>>
>>>>> but , if you enter the command "originate user/1001 5000" , x-lite
>>>>> will ring ,answer it ,you can hear the IVR voice , press some keys ,the
>>>>> freeswitch can NOT receive any dtmf ,
>>>>>
>>>>>
>>>>> why?
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.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
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.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
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> --
>>>
>>>
>>> Sincerely,
>>>
>>> Giovanni Maruzzelli
>>> OpenTelecom.IT
>>> cell: +39 347 266 56 18
>>>
>>>
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.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
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.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
>> http://www.freeswitch.org
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170614/2f9a2c8a/attachment-0001.html>

From gmaruzz at gmail.com  Wed Jun 14 16:42:11 2017
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Wed, 14 Jun 2017 18:42:11 +0200
Subject: [Freeswitch-users] When calling out ,
	freeswitch doesn't get DTMF ?
In-Reply-To: <59412DB4020000310000A381@mail.tedssupply.com>
References: <f823cdd8-a354-3698-d8c5-484b1b4cc8b4@xbipin.com>
 <CAEJMVkDDQX=Ni3AOQxQOo=joGoAeYroxioVP=cu9w6Y2m7kLPA@mail.gmail.com>
 <15c898b5d58.2779.b07ebdf329620b8089087c7205b03f01@xbipin.com>
 <310cf71c.8fc3.15c8ba9c4f6.Coremail.eastour@163.com>
 <CAFGRPVqFA+tpaOQkAGGCR7s5gY0z0TbdArWFdkfnsaaG4Z9Dgw@mail.gmail.com>
 <2a18422f.ef8.15c99b7b9bb.Coremail.eastour@163.com>
 <CALXCt0pKKEJoSeXXZ=tn04nifEZvQCRSFCDKL8jSu+pfOgvRNQ@mail.gmail.com>
 <6c11dd0f.c2c1.15c9c727dc7.Coremail.eastour@163.com>
 <CAFLo2QwwwE6xeXBN9DkM1VEecEJpwnDMmQ7iT9EvSyx9RqE9mg@mail.gmail.com>
 <2f48505d.d9c5.15ca1d28424.Coremail.eastour@163.com>
 <59401B92020000310000A34D@mail.tedssupply.com>
 <CALXCt0qw4DCJ2Q0cF+kVEOpjCjp2T8eFgWbYx03sN=z5APBGGQ@mail.gmail.com>
 <59412DB4020000310000A381@mail.tedssupply.com>
Message-ID: <CALXCt0rTBTBTfT+e1t0UCfpwmWyOhz1-yeQ2s6T8mSTYAnS-+w@mail.gmail.com>

On 14 June 2017 at 18:36, admin <admin at tedssupply.com> wrote:

> I don't mean to hijack the OP concern with my problem, I just wanted to
> reinforce I have seen this problem.  And the problem is that the call
> originated through FS appears not to send user entered DTMF to the
> receiving phone.  Let's help chenyzhi and then I'll take a turn.
>

I just tested right now from console with:


 bgapi originate user/1011 5000

and

originate user/1011 5000


and it works


Please pastebin your FreeSWITCH version (eg, type "version" in console),
your dialplan, your SIP profile, and the complete, since beginning to end,
unedited, debug output of console when receiving a call which does not get
DTMFs


Maybe helping you we'll help him too


>
> - James
>
> >>> Giovanni Maruzzelli <gmaruzz at gmail.com> 6/14/2017 03:53 AM >>>
>
> Never heard such problems
>
>
> Please pastebin your dialplan, your SIP profile, and the complete, since
> beginning to end, unedited, debug output of console when receiving a call
> which does not get DTMFs
>
>
>
>
> sent from mobile
> cell: +39 347 266 56 18
> Giovanni Maruzzelli
> OpenTelecom.IT
>
>
> On Jun 13, 2017 23:07, "admin" <admin at tedssupply.com> wrote:
>
>> I am encountering the same issue. I am using an ESL call to api
>> originate, with little else except getting in and out of the port, and the
>> caller DTMF is failing to reach the called number. The called number auto
>> answer attendant does not respond to DTMF, and a call to a test phone
>> confirms no DTMF. This is a new upgrade from 1.2 to 1.6 and I don't know if
>> this was an issue in 1.2, but my users never complained before 1.6 upgrade.
>> Ideas?...
>>
>>
>>
>> >>> chenyzhi <eastour at 163.com> 06/13/17 3:32 PM >>>
>>
>> It's a one leg call .there is no b-leg.
>>
>>
>> please make a test on your freeswitch box .
>>
>>
>> just type the command "originate user/1001 5000" on the freeswitch
>> console to see if your freeswitch instance can detect dtmf input.
>>
>>
>>
>>
>>
>>
>>
>>
>> At 2017-06-12 22:57:17, "Luis Daniel Lucio Quiroz" <
>> luis.daniel.lucio at gmail.com> wrote:
>>
>> check if you have transcoding, and if you do, check that dftm type-codec
>> on leg b are compatible.
>>
>>
>> --
>>
>> Luis Daniel Lucio Quiroz
>> CISSP, CISM, CISA
>> Linux, VoIP and much more fun
>> www.okay.com.mx
>>
>> Need LCR? Check out LCR for FusionPBX with FreeSWITCH
>> Need Billing? Check out Billing for FusionPBX with FreeSWITCH
>>
>>
>> On Mon, Jun 12, 2017 at 9:16 AM, chenyzhi <eastour at 163.com> wrote:
>>
>>> I don't think It's the DTMF type , because when I dial 5000 from x-lite
>>> (which has registered to freeswitch as 1001) ,I can hear the voice and if I
>>> press any dtmf on x-lite, freeswitch can recieve the dtmfs. This means that
>>> the DTMF type is correct ,otherwise freeswitch coudn't have received the
>>> dtmfs;
>>>
>>>
>>> when I enter the command "originate user/1001 5000" at the freeswitch
>>> console , my xlite will ring ,and I answered ,I can hear the voice ,I press
>>> some dtmf,but freeswitch can NOT receive any dtmf. really weird.
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>> At 2017-06-12 17:37:49, "Giovanni Maruzzelli" <gmaruzz at gmail.com> wrote:
>>>
>>> you need to check the DTMF type, you probably are using the wrong one
>>> (info-inband-rfc2833), and for some reason they are not negotiated
>>>
>>>
>>> On 12 June 2017 at 02:32, chenyzhi <eastour at 163.com> wrote:
>>>
>>>> I have read the logs ,but I didn't find any difference.
>>>>
>>>>
>>>> Please make a test to see if this happens in your box.
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> At 2017-06-10 06:49:34, "David Villasmil" <
>>>> david.villasmil.work at gmail.com> wrote:
>>>>
>>>> Have you looked at the log? Bump the logging up and see what shows
>>>> up... what you're seeing is very weird
>>>>
>>>> David
>>>>
>>>> On Fri, Jun 9, 2017 at 11:18 AM chenyzhi <eastour at 163.com> wrote:
>>>>
>>>>> Hi
>>>>>
>>>>>
>>>>> call 5000 from x-lite , you can hear the IVR voice and if you press
>>>>> dtmf keys , freeswitch can receive the dtmf keys .
>>>>>
>>>>>
>>>>> but , if you enter the command "originate user/1001 5000" , x-lite
>>>>> will ring ,answer it ,you can hear the IVR voice , press some keys ,the
>>>>> freeswitch can NOT receive any dtmf ,
>>>>>
>>>>>
>>>>> why?
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> ____________________________________________________________
>>>>> _____________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.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
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>>
>>>>
>>>> ____________________________________________________________
>>>> _____________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.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
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> --
>>>
>>>
>>> Sincerely,
>>>
>>> Giovanni Maruzzelli
>>> OpenTelecom.IT
>>> cell: +39 347 266 56 18
>>>
>>>
>>>
>>>
>>>
>>> ____________________________________________________________
>>> _____________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.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
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.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
>> http://www.freeswitch.org
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org
>



-- 

Sincerely,

Giovanni Maruzzelli
OpenTelecom.IT
cell: +39 347 266 56 18
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170614/d324ad61/attachment-0001.html>

From rick at magicmail.mooo.com  Wed Jun 14 16:53:08 2017
From: rick at magicmail.mooo.com (Rick Jarvis)
Date: Wed, 14 Jun 2017 17:53:08 +0100
Subject: [Freeswitch-users] Paging
In-Reply-To: <CAEJMVkC0y=W7p4mFumdjHGYqvfxxUk8CJELgDv7-z2SYKUEYRQ@mail.gmail.com>
References: <8AE4D7F5-1AE6-44F8-B85D-13C4FADC22CA@magicmail.mooo.com>
 <CANybnk5eZ3Yqn4-2gJz6hWaDXskgPOs804_v0CV3WgcCqHYB4A@mail.gmail.com>
 <CAHKqMb7srMTQrm9Hz8ZNreKPZYuO7K11wJCL0_wD9ZJMyTtR=Q@mail.gmail.com>
 <5D4CCF18-DAA0-4F84-8203-4F69D66B251A@magicmail.mooo.com>
 <CAEJMVkC0y=W7p4mFumdjHGYqvfxxUk8CJELgDv7-z2SYKUEYRQ@mail.gmail.com>
Message-ID: <EBBBDB25-0565-47D3-B39B-31BAEAD17BBB@magicmail.mooo.com>

I’m correct in thinking that for multicast to work, the endpoints have to be on the same LAN, am I? Haven’t ever played with multicast.

I don’t know why I hadn’t realised that mad boss looks as if it will do exactly what I’m after, hopefully good for a dozen extensions or so… and it mutes all the handsets in the group it calls, right?

> On 14 Jun 2017, at 17:27, Brian West <brian at freeswitch.org> wrote:
> 
> Without multicast you'll have to review the mad boss example in the vanilla config.  Doing so in this matter doesn't scale well.
> 
> ./b
> 
> On Wed, Jun 14, 2017 at 6:03 AM, Rick Jarvis <rick at magicmail.mooo.com <mailto:rick at magicmail.mooo.com>> wrote:
> Thanks Nandy. We’ve used PA for stuff, but what we’re trying to accomplish here is paging through handsets. It’s kind of strange that there isn’t more support for this as it’s quite a standard feature even on the old analog PBX systems, Panasonic and so on.
> 
> 
>> On 14 Jun 2017, at 10:43, Nandy Dagondon <nandy1925 at gmail.com <mailto:nandy1925 at gmail.com>> wrote:
>> 
>> I've done this using mod_portaudio with auto-answer from this example:
>> https://wiki.freeswitch.org/wiki/Mod_portaudio#PA_System_w.2F_Chime <https://wiki.freeswitch.org/wiki/Mod_portaudio#PA_System_w.2F_Chime>
>> 
>> /nandy
>> 
>> On Fri, May 19, 2017 at 12:04 PM, Lesley Pervis <lesley.pervis at gmail.com <mailto:lesley.pervis at gmail.com>> wrote:
>> I've done this in Lua with outbound conference calls, but it was pretty complicated and you have to have endpoints that will auto-answer. https://freeswitch.org/confluence/display/FREESWITCH/Outbound+Conference+Calls <https://freeswitch.org/confluence/display/FREESWITCH/Outbound+Conference+Calls>
>> 
>> On Wed, May 17, 2017 at 10:02 AM, Rick Jarvis <rick at magicmail.mooo.com <mailto:rick at magicmail.mooo.com>> wrote:
>> Looking to set up paging (not multicast). What’s the best way of achieving this? Specifically, I want to have the receiving handset(s) answer muted for privacy reasons, so it’s literally like a PA system rather than just auto answer…?
>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org <http://www.freeswitch.org/>
>> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
>> http://www.cluecon.com <http://www.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>
>> http://www.freeswitch.org <http://www.freeswitch.org/>
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org <http://www.freeswitch.org/>
>> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
>> http://www.cluecon.com <http://www.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>
>> http://www.freeswitch.org <http://www.freeswitch.org/>
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org <http://www.freeswitch.org/>
>> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
>> http://www.cluecon.com <http://www.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>
>> http://www.freeswitch.org <http://www.freeswitch.org/>
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> http://www.cluecon.com <http://www.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>
> http://www.freeswitch.org <http://www.freeswitch.org/>
> 
> 
> 
> -- 
> Brian West
> brian at freeswitch.org <mailto:brian at freeswitch.org>
> Twitter: @FreeSWITCH , @briankwest
> 
> http://www.freeswitchbook.com <http://www.freeswitchbook.com/> 
> http://www.freeswitchcookbook.com <http://www.freeswitchcookbook.com/>
> Book a phone call (CST) <https://freeswitch.com/appointment>
> 
> Allison prompts for FreeSWITCH:
> 
> https://www.gofundme.com/allison-prompts-for-freeswitch <https://www.gofundme.com/allison-prompts-for-freeswitch>
> Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit: /r/freeswitch <https://www.reddit.com/r/freeswitch>
> T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378)
> Skype:briankwest
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170614/5f200164/attachment.html>

From jnankin at gmail.com  Wed Jun 14 17:34:50 2017
From: jnankin at gmail.com (Josh Nankin)
Date: Wed, 14 Jun 2017 12:34:50 -0500
Subject: [Freeswitch-users] hello
Message-ID: <CAAYWZ8nDAeDnc+sU83fV=5rb6UFjG=hmRpdNAM4NOvjERigq6w@mail.gmail.com>


-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170614/191f79a7/attachment.html>

From jnankin at gmail.com  Wed Jun 14 19:22:15 2017
From: jnankin at gmail.com (Josh Nankin)
Date: Wed, 14 Jun 2017 14:22:15 -0500
Subject: [Freeswitch-users] RPORT still being sent in TCP calls
In-Reply-To: <CAOsHgtuu-2S+yW76VNfnWuRK3WeR94kABGUmAk9gMTPgDPNh_A@mail.gmail.com>
References: <CAOsHgtszqP2ph+LLadnHWmm6cMmU4r6=Og4smfydScoVkMh0wQ@mail.gmail.com>
 <CAKbxfG8n7ycf_vgkFE73=mDX_anW2z284onh89vdSB2mZO=ykw@mail.gmail.com>
 <CAOsHgtt0X+-iLDCqEpu4wc8obgjp1kBe_9zAoprJi+7DU2SZ1g@mail.gmail.com>
 <8E832DB1-BEF7-4EF1-826F-ED57C966E794@jerris.com>
 <CAOsHgtuu-2S+yW76VNfnWuRK3WeR94kABGUmAk9gMTPgDPNh_A@mail.gmail.com>
Message-ID: <CAAYWZ8kbJ4ssOks3FgaFQsRhAXqbHqe+8vEu2BPCcjUd+YKseQ@mail.gmail.com>

Opened https://freeswitch.org/jira/browse/FS-10392 with regards to this

On Wed, Jun 14, 2017 at 1:22 PM, Daniel Greenwald <dig1234 at gmail.com> wrote:

>
> ---------- Forwarded message ----------
> From: Michael Jerris <mike at jerris.com>
> Date: Mon, Jun 12, 2017 at 4:42 PM
> Subject: Re: [Freeswitch-users] RPORT still being sent in TCP calls
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>
>
> Change is still in there, confirmed.  The packages you are using are very
> old (but should also have that patch in it)… I see that you are supplying
> modified sip traces.  If you can reproduce this on master code, please
> create a Jira with configuration and full debug logs with sip trace
> (unmodified) attached.
>
> Thanks
> Mike
>
> On Jun 12, 2017, at 4:32 PM, Daniel Greenwald <dig1234 at gmail.com> wrote:
>
> We have upgraded to latest STABLE from Ubuntu packages but we are still
> seeing rport in TCP calls:
>
>
> INVITE sip:xxxxxxx at sip.freeswitch.com;transport=tcp SIP/2.0
> Via: SIP/2.0/TCP x.x.x.x;rport;branch=z9hG4bKZDKF5N8QcFZcH
> Max-Forwards: 70
> From: "user" <sip:xxxxxx at sip.freeswitch.com>;tag=Z1pQm54mB22De
> To: <sip:xxxxxx at sip.freeswitch.com;transport=tcp>
> Call-ID: c79656fc-d2d0-4446-90c3-060dabf82fd6
> CSeq: 108312748 INVITE
> Contact: <sip:user at x.x.x.x:5060;transport=tcp;gw=provider>
> User-Agent: FreeSWITCH-mod_sofia/1.6.10-4-726448d~64bit
>
> On Fri, Jun 9, 2017 at 2:56 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> Odd number releases are not stable releases they are dev releases.
>> The version you quoted is from may 2015 2 years ago.
>>
>> The latest stable release is 1.6.17 and latest build release changes
>> every day.
>>
>>
>>
>> On Fri, Jun 9, 2017 at 10:50 AM, Daniel Greenwald <dig1234 at gmail.com>
>> wrote:
>>
>>> We have noticed that FS is sending RPORT in TCP calls to a gateway. It
>>> was reported as fixed in this bug:
>>> https://freeswitch.org/jira/browse/FS-6612
>>>
>>> We are running:
>>> 1.5.15b+git~20150512T053645Z~9eb887af47~64bit
>>>
>>> I am not sure why RPORT is still being sent. Is this there a config
>>> parameter which needs to be set to suppress the RPORT? Or was this change
>>> reverted in later versions for some reason. Provider is telling us we
>>> should not be sending RPORT in TCP...
>>>
>>> Any info would be greatly appreciated.
>>>
>>> Thanks!
>>>
>>> ____________________________________________________________
>>> _____________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.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
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Anthony Minessale II       ♬ @anthmfs  ♬ @FreeSWITCH  ♬
>>
>> ☞ http://freeswitch.org/  ☞ http://cluecon.com/  ☞
>> http://twitter.com/FreeSWITCH
>> ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+
>> <http://freeswitch.org/g+>*
>>
>> ClueCon Weekly Development Call
>> ☎ sip:888 at conference.freeswitch.org  ☎ +19193869900 <(919)%20386-9900>
>>
>> https://www.youtube.com/watch?v=9XXgW34t40s
>> https://www.youtube.com/watch?v=NLaDpGQuZDA
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.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
>> http://www.freeswitch.org
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170614/eccb6b40/attachment-0001.html>

From prex5609 at gmail.com  Wed Jun 14 23:07:59 2017
From: prex5609 at gmail.com (Peter Rex)
Date: Wed, 14 Jun 2017 17:07:59 -0600
Subject: [Freeswitch-users] Paging
In-Reply-To: <EBBBDB25-0565-47D3-B39B-31BAEAD17BBB@magicmail.mooo.com>
References: <8AE4D7F5-1AE6-44F8-B85D-13C4FADC22CA@magicmail.mooo.com>
 <CANybnk5eZ3Yqn4-2gJz6hWaDXskgPOs804_v0CV3WgcCqHYB4A@mail.gmail.com>
 <CAHKqMb7srMTQrm9Hz8ZNreKPZYuO7K11wJCL0_wD9ZJMyTtR=Q@mail.gmail.com>
 <5D4CCF18-DAA0-4F84-8203-4F69D66B251A@magicmail.mooo.com>
 <CAEJMVkC0y=W7p4mFumdjHGYqvfxxUk8CJELgDv7-z2SYKUEYRQ@mail.gmail.com>
 <EBBBDB25-0565-47D3-B39B-31BAEAD17BBB@magicmail.mooo.com>
Message-ID: <CAKDN4Ffh3Ukz1kwBYtxm05cWqLxLZwB9kC4+Fkg4Qz+SxFsT3A@mail.gmail.com>

I thin I added "flags{mute} inline" to each leg dialstring.



On Wed, Jun 14, 2017 at 10:53 AM, Rick Jarvis <rick at magicmail.mooo.com>
wrote:

> I’m correct in thinking that for multicast to work, the endpoints have to
> be on the same LAN, am I? Haven’t ever played with multicast.
>
> I don’t know why I hadn’t realised that mad boss looks as if it will do
> exactly what I’m after, hopefully good for a dozen extensions or so… and it
> mutes all the handsets in the group it calls, right?
>
>
> On 14 Jun 2017, at 17:27, Brian West <brian at freeswitch.org> wrote:
>
> Without multicast you'll have to review the mad boss example in the
> vanilla config.  Doing so in this matter doesn't scale well.
>
> ./b
>
> On Wed, Jun 14, 2017 at 6:03 AM, Rick Jarvis <rick at magicmail.mooo.com>
> wrote:
>
>> Thanks Nandy. We’ve used PA for stuff, but what we’re trying to
>> accomplish here is paging through handsets. It’s kind of strange that there
>> isn’t more support for this as it’s quite a standard feature even on the
>> old analog PBX systems, Panasonic and so on.
>>
>>
>> On 14 Jun 2017, at 10:43, Nandy Dagondon <nandy1925 at gmail.com> wrote:
>>
>> I've done this using mod_portaudio with auto-answer from this example:
>> https://wiki.freeswitch.org/wiki/Mod_portaudio#PA_System_w.2F_Chime
>>
>> /nandy
>>
>> On Fri, May 19, 2017 at 12:04 PM, Lesley Pervis <lesley.pervis at gmail.com>
>> wrote:
>>
>>> I've done this in Lua with outbound conference calls, but it was pretty
>>> complicated and you have to have endpoints that will auto-answer.
>>> https://freeswitch.org/confluence/display/FREESWITCH/Outboun
>>> d+Conference+Calls
>>>
>>> On Wed, May 17, 2017 at 10:02 AM, Rick Jarvis <rick at magicmail.mooo.com>
>>> wrote:
>>>
>>>> Looking to set up paging (not multicast). What’s the best way of
>>>> achieving this? Specifically, I want to have the receiving handset(s)
>>>> answer muted for privacy reasons, so it’s literally like a PA system rather
>>>> than just auto answer…?
>>>>
>>>>
>>>> ____________________________________________________________
>>>> _____________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.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/free
>>>> switch-users
>>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> ____________________________________________________________
>>> _____________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.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
>>> http://www.freeswitch.org
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.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
>> http://www.freeswitch.org
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.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
>> http://www.freeswitch.org
>>
>
>
>
> --
>
> *Brian West*
> brian at freeswitch.org
>
> *Twitter: @FreeSWITCH , @briankwest*
>
> http://www.freeswitchbook.com
> http://www.freeswitchcookbook.com
>
> Book a phone call (CST) <https://freeswitch.com/appointment>
>
> Allison prompts for FreeSWITCH:
>
> *https://www.gofundme.com/allison-prompts-for-freeswitch*
> <https://www.gofundme.com/allison-prompts-for-freeswitch>
>
> Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
> /r/freeswitch <https://www.reddit.com/r/freeswitch>
>
> *T:*+19184209001 <(918)%20420-9001> | *F:*+19184209002 <(918)%20420-9002>
> | *M:*+1918424WEST (9378)
> *Skype:*briankwest
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170614/d78bc799/attachment.html>

From agubbe at gmail.com  Thu Jun 15 08:07:01 2017
From: agubbe at gmail.com (=?UTF-8?Q?Agust=C3=AD_Ubalde_Bellot?=)
Date: Thu, 15 Jun 2017 10:07:01 +0200
Subject: [Freeswitch-users] Freeswitch sslv3 support
In-Reply-To: <CAGCz3RBiBhE7ism+Wfr81HShk42TDht4c-Yp_DY0cp35OCNqLQ@mail.gmail.com>
References: <CAGCz3RBiBhE7ism+Wfr81HShk42TDht4c-Yp_DY0cp35OCNqLQ@mail.gmail.com>
Message-ID: <CAGCz3RCjv0GHRkYmaPd31QLjd6pWBTdgxbbBmdeVRYwz0vU=RA@mail.gmail.com>

Hi Brian,

Is possible to disable for web socket secure connections too?


Thanks,
Agustí

2017-06-13 13:24 GMT+02:00 Agustí Ubalde Bellot <agubbe at gmail.com>:

> Hi all,
>
> Is there a FreeSWITCH update where sslv3 support is disabled?
>
>
> Thanks,
> Agustí
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170615/0a890534/attachment.html>

From eastour at 163.com  Thu Jun 15 02:28:43 2017
From: eastour at 163.com (chenyzhi)
Date: Thu, 15 Jun 2017 10:28:43 +0800 (CST)
Subject: [Freeswitch-users] When calling out ,
 freeswitch doesn't get DTMF ?
In-Reply-To: <CALXCt0rTBTBTfT+e1t0UCfpwmWyOhz1-yeQ2s6T8mSTYAnS-+w@mail.gmail.com>
References: <f823cdd8-a354-3698-d8c5-484b1b4cc8b4@xbipin.com>
 <CAEJMVkDDQX=Ni3AOQxQOo=joGoAeYroxioVP=cu9w6Y2m7kLPA@mail.gmail.com>
 <15c898b5d58.2779.b07ebdf329620b8089087c7205b03f01@xbipin.com>
 <310cf71c.8fc3.15c8ba9c4f6.Coremail.eastour@163.com>
 <CAFGRPVqFA+tpaOQkAGGCR7s5gY0z0TbdArWFdkfnsaaG4Z9Dgw@mail.gmail.com>
 <2a18422f.ef8.15c99b7b9bb.Coremail.eastour@163.com>
 <CALXCt0pKKEJoSeXXZ=tn04nifEZvQCRSFCDKL8jSu+pfOgvRNQ@mail.gmail.com>
 <6c11dd0f.c2c1.15c9c727dc7.Coremail.eastour@163.com>
 <CAFLo2QwwwE6xeXBN9DkM1VEecEJpwnDMmQ7iT9EvSyx9RqE9mg@mail.gmail.com>
 <2f48505d.d9c5.15ca1d28424.Coremail.eastour@163.com>
 <59401B92020000310000A34D@mail.tedssupply.com>
 <CALXCt0qw4DCJ2Q0cF+kVEOpjCjp2T8eFgWbYx03sN=z5APBGGQ@mail.gmail.com>
 <59412DB4020000310000A381@mail.tedssupply.com>
 <CALXCt0rTBTBTfT+e1t0UCfpwmWyOhz1-yeQ2s6T8mSTYAnS-+w@mail.gmail.com>
Message-ID: <7d1e3f12.4750.15ca994d89d.Coremail.eastour@163.com>

the version is:   FreeSWITCH Version 1.9.0+git~20170518T231917Z~a1fc18aee5~32bit (git a1fc18a 2017-05-18 23:19:17Z 32bit)


 the complete, since beginning to end, unedited, debug output of console when making a outgoing  call which does not get DTMFs


and the whole conf folder is in the attatchment.


thx!






在 2017-06-15 00:42:11，"Giovanni Maruzzelli" <gmaruzz at gmail.com> 写道：





On 14 June 2017 at 18:36, admin <admin at tedssupply.com> wrote:


I don't mean to hijack the OP concern with my problem, I just wanted to reinforce I have seen this problem.  And the problem is that the call originated through FS appears not to send user entered DTMF to the receiving phone.  Let's help chenyzhi and then I'll take a turn.




I just tested right now from console with:




 bgapi originate user/1011 5000

and

originate user/1011 5000




and it works





Please pastebin your FreeSWITCH version (eg, type "version" in console), your dialplan, your SIP profile, and the complete, since beginning to end, unedited, debug output of console when receiving a call which does not get DTMFs




Maybe helping you we'll help him too

 




- James

>>> Giovanni Maruzzelli <gmaruzz at gmail.com> 6/14/2017 03:53 AM >>>


Never heard such problems




Please pastebin your dialplan, your SIP profile, and the complete, since beginning to end, unedited, debug output of console when receiving a call which does not get DTMFs










sent from mobile
cell: +39 347 266 56 18
Giovanni Maruzzelli
OpenTelecom.IT




On Jun 13, 2017 23:07, "admin" <admin at tedssupply.com> wrote:


I am encountering the same issue. I am using an ESL call to api originate, with little else except getting in and out of the port, and the caller DTMF is failing to reach the called number. The called number auto answer attendant does not respond to DTMF, and a call to a test phone confirms no DTMF. This is a new upgrade from 1.2 to 1.6 and I don't know if this was an issue in 1.2, but my users never complained before 1.6 upgrade. Ideas?...



>>> chenyzhi <eastour at 163.com> 06/13/17 3:32 PM >>>


It's a one leg call .there is no b-leg.




please make a test on your freeswitch box .




just type the command "originate user/1001 5000" on the freeswitch console to see if your freeswitch instance can detect dtmf input.













At 2017-06-12 22:57:17, "Luis Daniel Lucio Quiroz" <luis.daniel.lucio at gmail.com> wrote:


check if you have transcoding, and if you do, check that dftm type-codec on leg b are compatible.




--

Luis Daniel Lucio Quiroz
CISSP, CISM, CISA
Linux, VoIP and much more fun
www.okay.com.mx

Need LCR? Check out LCR for FusionPBX with FreeSWITCH
Need Billing? Check out Billing for FusionPBX with FreeSWITCH




On Mon, Jun 12, 2017 at 9:16 AM, chenyzhi <eastour at 163.com> wrote:


I don't think It's the DTMF type , because when I dial 5000 from x-lite (which has registered to freeswitch as 1001) ,I can hear the voice and if I press any dtmf on x-lite, freeswitch can recieve the dtmfs. This means that the DTMF type is correct ,otherwise freeswitch coudn't have received the dtmfs;




when I enter the command "originate user/1001 5000" at the freeswitch console , my xlite will ring ,and I answered ,I can hear the voice ,I press some dtmf,but freeswitch can NOT receive any dtmf. really weird.













At 2017-06-12 17:37:49, "Giovanni Maruzzelli" <gmaruzz at gmail.com> wrote:


you need to check the DTMF type, you probably are using the wrong one (info-inband-rfc2833), and for some reason they are not negotiated





On 12 June 2017 at 02:32, chenyzhi <eastour at 163.com> wrote:


I have read the logs ,but I didn't find any difference.




Please make a test to see if this happens in your box.













At 2017-06-10 06:49:34, "David Villasmil" <david.villasmil.work at gmail.com> wrote:


Have you looked at the log? Bump the logging up and see what shows up... what you're seeing is very weird

David


On Fri, Jun 9, 2017 at 11:18 AM chenyzhi <eastour at 163.com> wrote:


Hi




call 5000 from x-lite , you can hear the IVR voice and if you press dtmf keys , freeswitch can receive the dtmf keys .




but , if you enter the command "originate user/1001 5000" , x-lite will ring ,answer it ,you can hear the IVR voice , press some keys ,the freeswitch can NOT receive any dtmf ,




why?








_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.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
http://www.freeswitch.org









_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.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
http://www.freeswitch.org





--



Sincerely,

Giovanni Maruzzelli
OpenTelecom.IT
cell: +39 347 266 56 18










_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.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
http://www.freeswitch.org













_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.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
http://www.freeswitch.org



_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.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
http://www.freeswitch.org




--


Sincerely,

Giovanni Maruzzelli
OpenTelecom.IT
cell: +39 347 266 56 18
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170615/ee9c42b8/attachment-0001.html>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: freeswitch-log-4callingoutdoesnotgetdtmf.7z
Type: application/x-7z-compressed
Size: 99772 bytes
Desc: not available
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170615/ee9c42b8/attachment-0001.7z>

From ron.menez at entropysolution.com  Thu Jun 15 07:49:04 2017
From: ron.menez at entropysolution.com (Ron)
Date: Thu, 15 Jun 2017 07:49:04 +0000
Subject: [Freeswitch-users] AMR Codec for Audio File Playback
Message-ID: <34D864D4-4CD0-4F6D-8B3C-83F359411180@entropysolution.com>

Hi All,

Is it possible to execute a command playback for an audio file using AMR Codec?

We tried the following configuration in dialplan and gave different errors:

Configuration with “pre-answer”:
  <extension name="AMR_Testing">
    <condition field="destination_number" expression="^\d{7}$">

      <action application="pre_answer"/>
      <action application="sleep" data="2000"/>

      <action application="playback" data="/usr/local/freeswitch/sounds/aaaaa.wav"/>

    </condition>
  </extension>

Error Log:
EXECUTE sofia/internal/09570000001 at 192.168.1.129<mailto:sofia/internal/09570000001 at 192.168.1.129>:5062 pre_answer()
2017-06-15 15:39:45.285763 [INFO] mod_dptools.c:1355 Sending early media
2017-06-15 15:39:45.285763 [DEBUG] switch_core_media.c:3056 Set Codec sofia/internal/09570000001 at 192.168.1.129<mailto:sofia/internal/09570000001 at 192.168.1.129>:5062 AMR/0 0 ms 160 samples 12200 bits 1 channels
2017-06-15 15:39:45.285763 [DEBUG] switch_core_codec.c:111 sofia/internal/09570000001 at 192.168.1.129<mailto:sofia/internal/09570000001 at 192.168.1.129>:5062 Original read codec set to AMR:96
2017-06-15 15:39:45.285763 [DEBUG] switch_core_media.c:6927 PROXY AUDIO RTP [sofia/internal/09570000001 at 192.168.1.129<mailto:sofia/internal/09570000001 at 192.168.1.129>:5062] 192.168.1.129:62020->192.168.1.129:62020 codec: 98 ms: 20
2017-06-15 15:39:45.285763 [ERR] switch_core_media.c:7549 AUDIO RTP REPORTS ERROR: [Missing local host]
2017-06-15 15:39:45.285763 [NOTICE] switch_core_media.c:7550 Hangup sofia/internal/09570000001 at 192.168.1.129<mailto:sofia/internal/09570000001 at 192.168.1.129>:5062 [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER]

Configuration with “answer”:
  <extension name="AMR_Testing">
    <condition field="destination_number" expression="^\d{7}$">

      <action application="answer"/>
      <action application="sleep" data="2000"/>

      <action application="playback" data="/usr/local/freeswitch/sounds/aaaaa.wav"/>

    </condition>
  </extension>

Error Log:
2017-06-15 15:41:49.965732 [NOTICE] mod_dptools.c:1312 Channel [sofia/internal/09570000001 at 192.168.1.129<mailto:sofia/internal/09570000001 at 192.168.1.129>:5062] has been answered
2017-06-15 15:41:49.965732 [DEBUG] switch_channel.c:3772 (sofia/internal/09570000001 at 192.168.1.129<mailto:sofia/internal/09570000001 at 192.168.1.129>:5062) Callstate Change EARLY -> ACTIVE
2017-06-15 15:41:49.965732 [DEBUG] sofia.c:7048 Channel sofia/internal/09570000001 at 192.168.1.129<mailto:sofia/internal/09570000001 at 192.168.1.129>:5062 entering state [completed][200]
2017-06-15 15:41:49.965732 [DEBUG] sofia.c:7048 Channel sofia/internal/09570000001 at 192.168.1.129<mailto:sofia/internal/09570000001 at 192.168.1.129>:5062 entering state [ready][200]
EXECUTE sofia/internal/09570000001 at 192.168.1.129<mailto:sofia/internal/09570000001 at 192.168.1.129>:5062 sleep(2000)
2017-06-15 15:41:49.985728 [DEBUG] switch_rtp.c:7247 Correct audio ip/port confirmed.
EXECUTE sofia/internal/09570000001 at 192.168.1.129<mailto:sofia/internal/09570000001 at 192.168.1.129>:5062 playback(/usr/local/freeswitch/sounds/aaaaa.wav)
2017-06-15 15:41:51.985732 [DEBUG] switch_core_file.c:342 File /usr/local/freeswitch/sounds/aaaaa.wav sample rate 44100 doesn't match requested rate 8000
2017-06-15 15:41:51.985732 [WARNING] switch_core_file.c:360 File has 2 channels, muxing to 1 channel will occur.
2017-06-15 15:41:51.985732 [DEBUG] switch_ivr_play_say.c:1498 Codec Activated L16 at 8000hz 1 channels 20ms
2017-06-15 15:41:52.005728 [ERR] mod_amr.c:338 This codec is only usable in passthrough mode!
2017-06-15 15:41:52.005728 [ERR] switch_core_io.c:1434 Codec AMR encoder error!
2017-06-15 15:41:52.005728 [DEBUG] switch_ivr_play_say.c:1942 done playing file /usr/local/freeswitch/sounds/aaaaa.wav
2017-06-15 15:41:52.005728 [NOTICE] switch_core_state_machine.c:385 sofia/internal/09570000001 at 192.168.1.129<mailto:sofia/internal/09570000001 at 192.168.1.129>:5062 has executed the last dialplan instruction, hanging up.
2017-06-15 15:41:52.005728 [NOTICE] switch_core_state_machine.c:387 Hangup sofia/internal/09570000001 at 192.168.1.129<mailto:sofia/internal/09570000001 at 192.168.1.129>:5062 [CS_EXECUTE] [NORMAL_CLEARING]

We also tried to use the “fs_encode" and tried AMR codec to encode the wav file but we received the error below:

Opening file aaaaa.wav
Opening file aaaaa.AMR
2017-05-25 16:18:32.078353 [INFO] mod_native_file.c:101 Opening File [aaaaa.AMR] 8000hz
Frame size is 160
2017-05-25 16:18:32.078366 [ERR] mod_amr.c:338 This codec is only usable in passthrough mode!
Codec encoder error
2017-05-25 16:18:32.078372 [WARNING] switch_core_codec.c:920 Codec is not initialized!

We tried using the native file configuration and absolute codec but same errors shown for “pre_answer” and “answer” configuration.

May we request your help if there is another way to run playback for audio files using AMR codec.

Thank you.

Best Regard,

Ron Menez
ron.menez at entropysolution.com<mailto:ron.menez at entropysolution.com>




-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170615/2aa60024/attachment-0001.html>

From USZPELSV at comunycarse.com  Thu Jun 15 07:49:24 2017
From: USZPELSV at comunycarse.com (Sven Uszpelkat)
Date: Thu, 15 Jun 2017 07:49:24 +0000
Subject: [Freeswitch-users] Call recording
Message-ID: <7c47071f86ce436186b4ab8bf58de46c@w12mad02.comunycarse.com>

Hello,

We are using FreeSWITCH as a third-party recording application, i.e. we are receiving SIP calls with the complete audio of conversations taking place on another switch and we are saving this audio to a file. To achieve this we are using a simple script similar to this:

session:answer()
while(session:ready() == true) do
   test = session:recordFile("/usr/local/freeswitch/recordings/test.wav", 18000, 0, 300)
   session:setAutoHangup(false)
   session:hangup()
end

This script will be invoked by the following dialplan:

<extension name="Recording">
      <condition field="destination_number" expression=".*">
            <action application="lua" data="record.lua"/>
      </condition>
</extension>

Basically it seems to work quite well, but sometimes there are missing audio at the end of the recorded file.  Usually it's only a few seconds, but sometimes it seems to be more. (It's like the recording sometimes goes behind the real call and when the hangup event is received the remaining audio is discarded.)

What could be the reason for this behavior? Is there something wrong with the script or is there a better way to achieve our goal?

Many thanks in advance.

Best regards,

Sven Uszpelkat
Departamento I+D
Comunycarse Network Consultants, S.L.


[Descripción: Descripción: http://www.comunycarse.com/email_images/facebook_16.jpg]<http://www.facebook.com/pages/Comunycarse/227218309114> [Descripción: Descripción: http://www.comunycarse.com/email_images/linkedin_16.jpg] <http://www.linkedin.com/company/comunycarse-s.l.>  [Descripción: Descripción: http://www.comunycarse.com/email_images/twitter_16.jpg] <https://twitter.com/#!/comunycarse>  [Descripción: Descripción: http://www.comunycarse.com/email_images/wordpress_16.jpg] <http://www.blog.comunycarse.com/>

Joaquín Turina, 2
28224 Pozuelo de Alarcón MADRID
Tlf. +34 917 498 700
Fax +34 917 498 720

Sabino Arana, 18
08028 BARCELONA
Tlf. +34 934 098 480
Fax +34 934 098 490

http://www.comunycarse.com<http://www.comunycarse.com/>

AVISO LEGAL
La presente comunicación y sus anexos tiene como destinatario la persona a la que va dirigida, por lo que si usted lo recibe por error debe notificarlo al remitente y eliminarlo de su sistema, no pudiendo utilizarlo, total o parcialmente, para ningún fin. Su contenido puede tener información confidencial o protegida legalmente y únicamente expresa la opinión del remitente. El uso del correo electrónico vía internet no permite asegurar ni la confidencialidad de los mensajes ni su correcta recepción. En el caso de que el destinatario no consintiera la utilización del correo electrónico deberá ponerlo en nuestro conocimiento inmediatamente.

DISCLAIMER
This message and its attachments are intended exclusively for the named addressee. If you receive this message by mistake, please delete it immediately from your system and notify the sender. You may not use this message or any part of it for any purpose. The message may contain information that is confidential or protected by law, and any opinions expressed are those of the individual sender. Internet email guarantees neither the confidentiality nor the proper receipt of the message sent. If the addressee of this message does not consent to the use of internet e-mail, please inform us immediately.

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170615/8b417d63/attachment-0001.html>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: image001.jpg
Type: image/jpeg
Size: 1630 bytes
Desc: image001.jpg
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170615/8b417d63/attachment-0004.jpg>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: image002.jpg
Type: image/jpeg
Size: 1694 bytes
Desc: image002.jpg
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170615/8b417d63/attachment-0005.jpg>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: image003.jpg
Type: image/jpeg
Size: 1628 bytes
Desc: image003.jpg
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170615/8b417d63/attachment-0006.jpg>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: image004.jpg
Type: image/jpeg
Size: 1402 bytes
Desc: image004.jpg
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170615/8b417d63/attachment-0007.jpg>

From kkothari157 at gmail.com  Thu Jun 15 06:48:20 2017
From: kkothari157 at gmail.com (Ketan Kothari)
Date: Thu, 15 Jun 2017 12:18:20 +0530
Subject: [Freeswitch-users] Verto failover
Message-ID: <CAGqqW9-YHn5nW5f4OFz0=EedosPyWzcM4Y0Cd3Cn31TX6j4=aQ@mail.gmail.com>

Can we setup Verto communicator *User-Interface* and *FreeSWITCH* on
separate server?
If yes

*--->* We have 2 FreeSWITCH servers and 1 User-Interface server. So how to
load balance or fail-over of Verto?
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170615/902cac9e/attachment.html>

From gmaruzz at gmail.com  Thu Jun 15 09:04:48 2017
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Thu, 15 Jun 2017 11:04:48 +0200
Subject: [Freeswitch-users] When calling out ,
	freeswitch doesn't get DTMF ?
In-Reply-To: <7d1e3f12.4750.15ca994d89d.Coremail.eastour@163.com>
References: <f823cdd8-a354-3698-d8c5-484b1b4cc8b4@xbipin.com>
 <CAEJMVkDDQX=Ni3AOQxQOo=joGoAeYroxioVP=cu9w6Y2m7kLPA@mail.gmail.com>
 <15c898b5d58.2779.b07ebdf329620b8089087c7205b03f01@xbipin.com>
 <310cf71c.8fc3.15c8ba9c4f6.Coremail.eastour@163.com>
 <CAFGRPVqFA+tpaOQkAGGCR7s5gY0z0TbdArWFdkfnsaaG4Z9Dgw@mail.gmail.com>
 <2a18422f.ef8.15c99b7b9bb.Coremail.eastour@163.com>
 <CALXCt0pKKEJoSeXXZ=tn04nifEZvQCRSFCDKL8jSu+pfOgvRNQ@mail.gmail.com>
 <6c11dd0f.c2c1.15c9c727dc7.Coremail.eastour@163.com>
 <CAFLo2QwwwE6xeXBN9DkM1VEecEJpwnDMmQ7iT9EvSyx9RqE9mg@mail.gmail.com>
 <2f48505d.d9c5.15ca1d28424.Coremail.eastour@163.com>
 <59401B92020000310000A34D@mail.tedssupply.com>
 <CALXCt0qw4DCJ2Q0cF+kVEOpjCjp2T8eFgWbYx03sN=z5APBGGQ@mail.gmail.com>
 <59412DB4020000310000A381@mail.tedssupply.com>
 <CALXCt0rTBTBTfT+e1t0UCfpwmWyOhz1-yeQ2s6T8mSTYAnS-+w@mail.gmail.com>
 <7d1e3f12.4750.15ca994d89d.Coremail.eastour@163.com>
Message-ID: <CALXCt0oRdP=8M_vbQpevzUs=NAO-rDVx0BqXJzv6mhy9Q5vZkg@mail.gmail.com>

On 15 June 2017 at 04:28, chenyzhi <eastour at 163.com> wrote:

> the version is:   FreeSWITCH Version 1.9.0+git~20170518T231917Z~a1fc18aee5~32bit
> (git a1fc18a 2017-05-18 23:19:17Z 32bit)
>
>  the complete, since beginning to end, unedited, debug output of console
> when making a outgoing  call which does not get DTMFs
>
> and the whole conf folder is in the attatchment.
>
>
from the log, seems it does not read all the IVR messages correctly to you,
it exits straight away...

are you able to correctly hear all the IVR messages?

also, can you take a SIP trace? (from console: "sofia global siptrace on")

I suspect you have a NAT problem of some sort

Also, I see you are on MASTER git, on Windows, and on 32 bit... Not sure
this is supported... Have you has this problems with stable branch (1.6.x)?







> thx!
>
>
>
>
>
> 在 2017-06-15 00:42:11，"Giovanni Maruzzelli" <gmaruzz at gmail.com> 写道：
>
>
>
> On 14 June 2017 at 18:36, admin <admin at tedssupply.com> wrote:
>
>> I don't mean to hijack the OP concern with my problem, I just wanted to
>> reinforce I have seen this problem.  And the problem is that the call
>> originated through FS appears not to send user entered DTMF to the
>> receiving phone.  Let's help chenyzhi and then I'll take a turn.
>>
>
> I just tested right now from console with:
>
>
>  bgapi originate user/1011 5000
>
> and
>
> originate user/1011 5000
>
>
> and it works
>
>
> Please pastebin your FreeSWITCH version (eg, type "version" in console),
> your dialplan, your SIP profile, and the complete, since beginning to end,
> unedited, debug output of console when receiving a call which does not get
> DTMFs
>
>
> Maybe helping you we'll help him too
>
>
>>
>> - James
>>
>> >>> Giovanni Maruzzelli <gmaruzz at gmail.com> 6/14/2017 03:53 AM >>>
>>
>> Never heard such problems
>>
>>
>> Please pastebin your dialplan, your SIP profile, and the complete, since
>> beginning to end, unedited, debug output of console when receiving a call
>> which does not get DTMFs
>>
>>
>>
>>
>> sent from mobile
>> cell: +39 347 266 56 18
>> Giovanni Maruzzelli
>> OpenTelecom.IT
>>
>>
>> On Jun 13, 2017 23:07, "admin" <admin at tedssupply.com> wrote:
>>
>>> I am encountering the same issue. I am using an ESL call to api
>>> originate, with little else except getting in and out of the port, and the
>>> caller DTMF is failing to reach the called number. The called number auto
>>> answer attendant does not respond to DTMF, and a call to a test phone
>>> confirms no DTMF. This is a new upgrade from 1.2 to 1.6 and I don't know if
>>> this was an issue in 1.2, but my users never complained before 1.6 upgrade.
>>> Ideas?...
>>>
>>>
>>>
>>> >>> chenyzhi <eastour at 163.com> 06/13/17 3:32 PM >>>
>>>
>>> It's a one leg call .there is no b-leg.
>>>
>>>
>>> please make a test on your freeswitch box .
>>>
>>>
>>> just type the command "originate user/1001 5000" on the freeswitch
>>> console to see if your freeswitch instance can detect dtmf input.
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>> At 2017-06-12 22:57:17, "Luis Daniel Lucio Quiroz" <
>>> luis.daniel.lucio at gmail.com> wrote:
>>>
>>> check if you have transcoding, and if you do, check that dftm type-codec
>>> on leg b are compatible.
>>>
>>>
>>> --
>>>
>>> Luis Daniel Lucio Quiroz
>>> CISSP, CISM, CISA
>>> Linux, VoIP and much more fun
>>> www.okay.com.mx
>>>
>>> Need LCR? Check out LCR for FusionPBX with FreeSWITCH
>>> Need Billing? Check out Billing for FusionPBX with FreeSWITCH
>>>
>>>
>>> On Mon, Jun 12, 2017 at 9:16 AM, chenyzhi <eastour at 163.com> wrote:
>>>
>>>> I don't think It's the DTMF type , because when I dial 5000 from x-lite
>>>> (which has registered to freeswitch as 1001) ,I can hear the voice and if I
>>>> press any dtmf on x-lite, freeswitch can recieve the dtmfs. This means that
>>>> the DTMF type is correct ,otherwise freeswitch coudn't have received the
>>>> dtmfs;
>>>>
>>>>
>>>> when I enter the command "originate user/1001 5000" at the freeswitch
>>>> console , my xlite will ring ,and I answered ,I can hear the voice ,I press
>>>> some dtmf,but freeswitch can NOT receive any dtmf. really weird.
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> At 2017-06-12 17:37:49, "Giovanni Maruzzelli" <gmaruzz at gmail.com>
>>>> wrote:
>>>>
>>>> you need to check the DTMF type, you probably are using the wrong one
>>>> (info-inband-rfc2833), and for some reason they are not negotiated
>>>>
>>>>
>>>> On 12 June 2017 at 02:32, chenyzhi <eastour at 163.com> wrote:
>>>>
>>>>> I have read the logs ,but I didn't find any difference.
>>>>>
>>>>>
>>>>> Please make a test to see if this happens in your box.
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> At 2017-06-10 06:49:34, "David Villasmil" <
>>>>> david.villasmil.work at gmail.com> wrote:
>>>>>
>>>>> Have you looked at the log? Bump the logging up and see what shows
>>>>> up... what you're seeing is very weird
>>>>>
>>>>> David
>>>>>
>>>>> On Fri, Jun 9, 2017 at 11:18 AM chenyzhi <eastour at 163.com> wrote:
>>>>>
>>>>>> Hi
>>>>>>
>>>>>>
>>>>>> call 5000 from x-lite , you can hear the IVR voice and if you press
>>>>>> dtmf keys , freeswitch can receive the dtmf keys .
>>>>>>
>>>>>>
>>>>>> but , if you enter the command "originate user/1001 5000" , x-lite
>>>>>> will ring ,answer it ,you can hear the IVR voice , press some keys ,the
>>>>>> freeswitch can NOT receive any dtmf ,
>>>>>>
>>>>>>
>>>>>> why?
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> ____________________________________________________________
>>>>>> _____________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://confluence.freeswitch.org
>>>>>> http://www.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/free
>>>>>> switch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> ____________________________________________________________
>>>>> _____________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.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/free
>>>>> switch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>>
>>>>
>>>> Sincerely,
>>>>
>>>> Giovanni Maruzzelli
>>>> OpenTelecom.IT
>>>> cell: +39 347 266 56 18
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> ____________________________________________________________
>>>> _____________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.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/free
>>>> switch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>>
>>>
>>> ____________________________________________________________
>>> _____________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.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
>>> http://www.freeswitch.org
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.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
>> http://www.freeswitch.org
>>
>
>
>
> --
>
> Sincerely,
>
> Giovanni Maruzzelli
> OpenTelecom.IT
> cell: +39 347 266 56 18
>
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org
>



-- 

Sincerely,

Giovanni Maruzzelli
OpenTelecom.IT
cell: +39 347 266 56 18
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170615/87f1af6d/attachment-0001.html>

From gmaruzz at gmail.com  Thu Jun 15 09:19:09 2017
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Thu, 15 Jun 2017 11:19:09 +0200
Subject: [Freeswitch-users] Call recording
In-Reply-To: <7c47071f86ce436186b4ab8bf58de46c@w12mad02.comunycarse.com>
References: <7c47071f86ce436186b4ab8bf58de46c@w12mad02.comunycarse.com>
Message-ID: <CALXCt0ocv6Hhfq7vfHhQf15047E9d+bWfuSE+wkLbwgLLh6Ptw@mail.gmail.com>

On 15 June 2017 at 09:49, Sven Uszpelkat <USZPELSV at comunycarse.com> wrote:

> Hello,
>
>
>
> We are using FreeSWITCH as a third-party recording application, i.e. we
> are receiving SIP calls with the complete audio of conversations taking
> place on another switch and we are saving this audio to a file. To achieve
> this we are using a simple script similar to this:
>
>
>
> session:answer()
>
> while(session:ready() == true) do
>
>    test = session:recordFile("/usr/local/freeswitch/recordings/test.wav",
> 18000, 0, 300)
>
>    session:setAutoHangup(false)
>
>    session:hangup()
>
> end
>
>
>
> This script will be invoked by the following dialplan:
>
>
>
> <extension name="Recording">
>
>       <condition field="destination_number" expression=".*">
>
>             <action application="lua" data="record.lua"/>
>
>       </condition>
>
> </extension>
>
>
>
> Basically it seems to work quite well, but sometimes there are missing
> audio at the end of the recorded file.  Usually it’s only a few seconds,
> but sometimes it seems to be more. (It’s like the recording sometimes goes
> behind the real call and when the hangup event is received the remaining
> audio is discarded.)
>
>
>
> What could be the reason for this behavior? Is there something wrong with
> the script or is there a better way to achieve our goal?
>

One first question come to my mind: why do you use a script here? A simple
extension can do exactly the same, if you just want to record the session...

https://freeswitch.org/confluence/display/FREESWITCH/mod_dptools:+record_session

Anyway, if you want to use the script, why you first

session:setAutoHangup(false)

and after that you


session:hangup()


?

Also, you made the silence_threshold equal 0 (zero).
Have you has the same problems using a silence_threshold of, let's say, 30
(thirty), like in documentation? (
https://freeswitch.org/confluence/display/FREESWITCH/Lua+API+Reference#LuaAPIReference-session:recordFile
)

Hope this helps,

-giovanni



>
>
> Many thanks in advance.
>
>
>
> Best regards,
>
>
>
> Sven Uszpelkat
> Departamento I+D
> Comunycarse Network Consultants, S.L.
>
>
>
>
>
> [image: Descripción: Descripción:
> http://www.comunycarse.com/email_images/facebook_16.jpg]
> <http://www.facebook.com/pages/Comunycarse/227218309114> [image:
> Descripción: Descripción:
> http://www.comunycarse.com/email_images/linkedin_16.jpg]
> <http://www.linkedin.com/company/comunycarse-s.l.> [image: Descripción:
> Descripción: http://www.comunycarse.com/email_images/twitter_16.jpg]
> <https://twitter.com/#%21/comunycarse> [image: Descripción: Descripción:
> http://www.comunycarse.com/email_images/wordpress_16.jpg]
> <http://www.blog.comunycarse.com/>
>
> Joaquín Turina, 2
> 28224 Pozuelo de Alarcón MADRID
> Tlf. +34 917 498 700 <+34%20917%2049%2087%2000>
> Fax +34 917 498 720 <+34%20917%2049%2087%2020>
>
> Sabino Arana, 18
> 08028 BARCELONA
> Tlf. +34 934 098 480 <+34%20934%2009%2084%2080>
> Fax +34 934 098 490 <+34%20934%2009%2084%2090>
>
> http://www.comunycarse.com
>
>
> AVISO LEGAL
> La presente comunicación y sus anexos tiene como destinatario la persona a
> la que va dirigida, por lo que si usted lo recibe por error debe
> notificarlo al remitente y eliminarlo de su sistema, no pudiendo
> utilizarlo, total o parcialmente, para ningún fin. Su contenido puede tener
> información confidencial o protegida legalmente y únicamente expresa la
> opinión del remitente. El uso del correo electrónico vía internet no
> permite asegurar ni la confidencialidad de los mensajes ni su correcta
> recepción. En el caso de que el destinatario no consintiera la utilización
> del correo electrónico deberá ponerlo en nuestro conocimiento
> inmediatamente.
>
> DISCLAIMER
> This message and its attachments are intended exclusively for the named
> addressee. If you receive this message by mistake, please delete it
> immediately from your system and notify the sender. You may not use this
> message or any part of it for any purpose. The message may contain
> information that is confidential or protected by law, and any opinions
> expressed are those of the individual sender. Internet email guarantees
> neither the confidentiality nor the proper receipt of the message sent. If
> the addressee of this message does not consent to the use of internet
> e-mail, please inform us immediately.
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org
>



-- 

Sincerely,

Giovanni Maruzzelli
OpenTelecom.IT
cell: +39 347 266 56 18
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170615/c38154d1/attachment-0001.html>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: image001.jpg
Type: image/jpeg
Size: 1630 bytes
Desc: not available
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170615/c38154d1/attachment-0004.jpg>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: image004.jpg
Type: image/jpeg
Size: 1402 bytes
Desc: not available
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170615/c38154d1/attachment-0005.jpg>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: image002.jpg
Type: image/jpeg
Size: 1694 bytes
Desc: not available
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170615/c38154d1/attachment-0006.jpg>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: image003.jpg
Type: image/jpeg
Size: 1628 bytes
Desc: not available
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170615/c38154d1/attachment-0007.jpg>

From igorolhovskiy at gmail.com  Thu Jun 15 09:40:00 2017
From: igorolhovskiy at gmail.com (Igor Olhovskiy)
Date: Thu, 15 Jun 2017 12:40:00 +0300
Subject: [Freeswitch-users] question about HA solution
In-Reply-To: <CAEwTsAhpLWcAoK8jfaEZ+HS2NeNLLEgbyAZkNRiTJ4RMNRet4Q@mail.gmail.com>
References: <CADcrYG+RHGQ2fC7dSKK9x9WfVvYcTczOtbppCLgttRCfMGqArA@mail.gmail.com>
 <CAFLo2QxxbZ5Ged0J0p9_VfQ2nkGtuV0CDp9G763EjjuvqUv8iw@mail.gmail.com>
 <CADcrYG+sO0+WYUtYVN6pkpU-5SwyfurfshdUdhxb-c4Az1s49Q@mail.gmail.com>
 <8A13E0EC-FA13-4267-80F6-CE1A8E8360CF@jerris.com>
 <CADcrYGLbN1Ut6riF1Lf7UvwG9ypvR2ekWxt1m=e5BmqOzP_c=w@mail.gmail.com>
 <E55363B0-487F-4F65-8F3F-78620AB1AB73@jerris.com>
 <CADcrYGKK=cTmVzG5oY0MV3b+4CLqg0=YzGvCiDZNZdBqQe3mMA@mail.gmail.com>
 <CAEwTsAhpLWcAoK8jfaEZ+HS2NeNLLEgbyAZkNRiTJ4RMNRet4Q@mail.gmail.com>
Message-ID: <CAJTkRNuenLDn-mu34cZ4dgxNH_P0Wq8nPx4VF21dmgBFB3Zq0w@mail.gmail.com>

Hi!
Same situation here.
Idea is:
I’m having Freeswitch HA (keepalived, working, same database, calls
recovering…)
If I look on «show calls» at slave node, I see calls on master node.
I crash master node (with «fsctl crash»), calls are transferred to slave
node, restored, but when I run «show calls» on this (slave) node again, I
see 0 calls. But calls are actually going on.

So, it’s seems impossible to have 2nd recover on already recovered call.


In DB logs seen an errors like
insert into channels (uuid,direction,created,created_epoch,
name,state,callstate,dialplan,context,hostname,initial_cid_name,initial_cid_num,initial_ip_addr,initial_dest,initial_dialplan,initial_context)
values('57904410-a8ad-4c28-a88a-83bd2280e146','outbound','2017-06-15
19:30:32','1497519032','sofia/internal/113-akbepcb59gt2a at 172.17.240.50:5060
','CS_INIT','DOWN','XML','sip303.empowervoice.com
','blueAPACHE_test','103','103','172.17.240.50','113-akbepcb59gt2a','XML','
sip303.empowervoice.com')
Jun 15 19:30:31 E2-EVL-T-DB-01 postgres[28042]: [109-1] 2017-06-15 19:30:31
AEST [28042-103] freeswitch at freeswitch ERROR:  duplicate key value violates
unique constraint «channels_pkey"
Or like
statement: insert into calls
(call_uuid,call_created,call_created_epoch,caller_uuid,callee_uuid,hostname)
values ('ffaf3eb5-3fc5-47fe-adef-cc4dddf53bab','2017-06-15
19:30:32','1497519032','ffaf3eb5-3fc5-47fe-adef-cc4dddf53bab','57904410-a8ad-4c28-a88a-83bd2280e146','blueAPACHE_test')
Jun 15 19:30:31 E2-EVL-T-DB-01 postgres[28042]: [147-1] 2017-06-15 19:30:31
AEST [28042-142] freeswitch at freeswitch ERROR:  duplicate key value violates
unique constraint «calls_pkey"

Also I see much queries like this
delete from calls where (caller_uuid=‘ffaf3eb5-3fc5-47fe-adef-cc4dddf53bab’
or callee_uuid='ffaf3eb5-3fc5-47fe-adef-cc4dddf53bab')
delete from recovery where
runtime_uuid!=‘91f571c5-e0d2-462e-aa84-e4ca07052119’ and
technology=‘sofia’….
when calls are switched.

So, can this help to point an issue?

2017-06-08 18:48 GMT+03:00 Michael Jerris <mike at jerris.com>:

> check your db logs as nothing we are doing should be clearing those.
>
> On Thu, Jun 8, 2017 at 4:08 AM Denys Pozniak <denys.pozniak at crazycall.com>
> wrote:
>
>> Hello!
>>
>> My configs:
>>
>> *switch.conf.xml*
>>
>>     <param name="switchname" value="freeswitch01"/>
>>     <param name="core-recovery-db-dsn" value="odbc://freeswitch:root:
>> ubuntu"/>
>>     <param name="core-db-dsn" value="odbc://freeswitch:root:ubuntu"/>
>>     <param name="auto-create-schemas" value="false"/>
>>     <param name="auto-clear-sql" value="false"/>
>>
>>
>> *external.conf.xml*
>>
>>    <param name="track-calls" value="true"/>
>>    <param name="odbc-dsn" value="odbc://freeswitch:root:ubuntu"/>
>>
>>
>> On 7 June 2017 at 17:35, Michael Jerris <mike at jerris.com> wrote:
>>
>>> That param should keep it from doing so, if its not you are not setting
>>> it somehow or something else is wiping the db.
>>>
>>> On Jun 5, 2017, at 1:50 PM, Denys Pozniak <denys.pozniak at crazycall.com>
>>> wrote:
>>>
>>> Yes, correct. But when you restart FS on slave, it will erase database.
>>> And option auto-clear-sql=false not working for me.
>>>
>>> On Jun 5, 2017 6:32 PM, "Michael Jerris" <mike at jerris.com> wrote:
>>>
>>> recovered calls will get new entries in the table.
>>>
>>> On Jun 5, 2017, at 7:41 AM, Denys Pozniak <denys.pozniak at crazycall.com>
>>> wrote:
>>>
>>> Hello!
>>>
>>> Thank you *Raymond* about your explanation, but I dont agree with some
>>> point:
>>> *If it really need an answer about your question -- "if it is possible
>>> to move calls back".   I think it's unnecessary.*  - in my case I have
>>> two not equal servers, so I need to have only one as a master.
>>> If switchover happens I need to have ability to restore master back.
>>>
>>> Thank you *Luis* for your link, you can do simple test to understand
>>> what I am talking about: do call -> check on master and slave #show
>>> channels -> restart FS on slave -> check on master #show channels. In my
>>> case I dont see any active calls after this, so restoring back is not
>>> possible.
>>>
>>>
>>>
>>> On 3 June 2017 at 22:16, Luis Daniel Lucio Quiroz <
>>> luis.daniel.lucio at gmail.com> wrote:
>>>
>>>> You may want to read this article.
>>>>
>>>> http://inside-out.xyz/technology/how-to-configure-
>>>> freeswitch-for-ha.html
>>>>
>>>> Le 31 mai 2017 6:29 PM, "Denys Pozniak" <denys.pozniak at crazycall.com>
>>>> a écrit :
>>>>
>>>> Hello!
>>>>
>>>> I built FS HA solution based on keepalived and mysql master-master.
>>>> It works ok generally, but as I understand FS after restarting cleaning
>>>> own database.
>>>>
>>>> So when node1 fails calls jump to node2, after script restarts node1 it
>>>> is not possible to move calls back.
>>>>
>>>> Tried options in switch.conf.xml, but no luck:
>>>>
>>>>     <param name="auto-create-schemas" value="false"/>
>>>>     <param name="auto-clear-sql" value="false"/>
>>>>
>>>> Is there is a way to solve this?
>>>>
>>>>
>>>
>>> ____________________________________________________________
>>> _____________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.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
>>> http://www.freeswitch.org
>>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.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
>> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org
>



-- 
Best regards,
Igor
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170615/d15f222c/attachment.html>

From matt at supportedbusiness.com  Thu Jun 15 09:54:32 2017
From: matt at supportedbusiness.com (Matt Broad)
Date: Thu, 15 Jun 2017 10:54:32 +0100
Subject: [Freeswitch-users] group_confirm_file multiple files
Message-ID: <CAEeV61jMv9ytvzX_8URn+M7wiC=vpW0+3Nr7VjbE-Ld8jvR5eA@mail.gmail.com>

Hi,

I'm wondering if it is possible to play multiple files using the
group_confirm_file function.

I have 2 audio files that I would like to play 1 after the other and then
wait for the confirm key.


I have tried using mod_file_string, but get an error "Error from mpg123:
File access error. (code 22)", I assume this is due to the fact it is
reading the file string as one file rather than 2 separated by the !
delimiter.


thanks
Matt

Matt Broad
Tel: +44 (0)203 011 1313 <+44%2020%203011%201313>
Web: www.supportedbusiness.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170615/b81daa5c/attachment.html>

From s.safarov at gmail.com  Thu Jun 15 11:16:35 2017
From: s.safarov at gmail.com (Sergey Safarov)
Date: Thu, 15 Jun 2017 11:16:35 +0000
Subject: [Freeswitch-users] Verto failover
In-Reply-To: <CAGqqW9-YHn5nW5f4OFz0=EedosPyWzcM4Y0Cd3Cn31TX6j4=aQ@mail.gmail.com>
References: <CAGqqW9-YHn5nW5f4OFz0=EedosPyWzcM4Y0Cd3Cn31TX6j4=aQ@mail.gmail.com>
Message-ID: <CAHtxdDd885V0+LjRB24THaka1o40dWy1x-kO2Ko_jZZ5aXiyBQ@mail.gmail.com>

Yes.
Balancing can be done using:
1) at dns level;
2) using haproxy daemon;
3) using nginx as proxy.

чт, 15 июн. 2017 г. в 11:41, Ketan Kothari <kkothari157 at gmail.com>:

> Can we setup Verto communicator *User-Interface* and *FreeSWITCH* on
> separate server?
> If yes
>
> *--->* We have 2 FreeSWITCH servers and 1 User-Interface server. So how
> to load balance or fail-over of Verto?
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170615/a9bbd0b4/attachment-0001.html>

From gmaruzz at gmail.com  Thu Jun 15 11:16:52 2017
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Thu, 15 Jun 2017 13:16:52 +0200
Subject: [Freeswitch-users] Call recording
In-Reply-To: <5323e6cfb2844e59a19957fad4e8044c@w12mad02.comunycarse.com>
References: <7c47071f86ce436186b4ab8bf58de46c@w12mad02.comunycarse.com>
 <CALXCt0ocv6Hhfq7vfHhQf15047E9d+bWfuSE+wkLbwgLLh6Ptw@mail.gmail.com>
 <5323e6cfb2844e59a19957fad4e8044c@w12mad02.comunycarse.com>
Message-ID: <CALXCt0ohebExQsNw+VCB0sPT-7hkRQHAw_pS9dMsOD0cFtxriw@mail.gmail.com>

On 15 June 2017 at 12:50, Sven Uszpelkat <USZPELSV at comunycarse.com> wrote:


> 3.)    We set the silence threshold to 0 because the documentation is not
> very clear how to disable the silence detection. We don’t want the
> recording to stop in response to a period of silence. The point is to
> record everything.
>
>
The silence_threshold determines what is considered silence, eg below what
level of acoustic energy we state the stream is containing silence. Then,
we wait for "how_many_silence_seconds" or until hangup before stopping
recording.

So, maybe you are right, and setting it to 0 will consider silence only
when there is absolute silence in the stream, so for all practical
purposes, until hangup. I have no mean to check source code now.

On another hand, I can think at other possible causes for the premature end
of the recorded file: maybe you move the file before it has been flushed by
FreeSWITCH or by operating system? Maybe the hangup in the script
interrupts the recording in the script and close the file descriptor before
is flushed? (I am shooting in the dark)

You can try to leave out those two lines, and test again.

Also, you can insert a line that sync (flush) the filesystem before
exiting, just to be sure.

I would insert it after the while(session:ready())

A system(sync), or something similar will probably do.

Hope this helps,
-giovanni






>
>
> Best regards,
>
> Sven
>
>
>
> *De:* FreeSWITCH-users [mailto:freeswitch-users-bounc
> es at lists.freeswitch.org <freeswitch-users-bounces at lists.freeswitch.org>] *En
> nombre de *Giovanni Maruzzelli
> *Enviado el:* jueves, 15 de junio de 2017 11:19
> *Para:* FreeSWITCH Users Help
> *Asunto:* Re: [Freeswitch-users] Call recording
>
>
>
>
>
>
>
> On 15 June 2017 at 09:49, Sven Uszpelkat <USZPELSV at comunycarse.com> wrote:
>
> Hello,
>
>
>
> We are using FreeSWITCH as a third-party recording application, i.e. we
> are receiving SIP calls with the complete audio of conversations taking
> place on another switch and we are saving this audio to a file. To achieve
> this we are using a simple script similar to this:
>
>
>
> session:answer()
>
> while(session:ready() == true) do
>
>    test = session:recordFile("/usr/local/freeswitch/recordings/test.wav",
> 18000, 0, 300)
>
>    session:setAutoHangup(false)
>
>    session:hangup()
>
> end
>
>
>
> This script will be invoked by the following dialplan:
>
>
>
> <extension name="Recording">
>
>       <condition field="destination_number" expression=".*">
>
>             <action application="lua" data="record.lua"/>
>
>       </condition>
>
> </extension>
>
>
>
> Basically it seems to work quite well, but sometimes there are missing
> audio at the end of the recorded file.  Usually it’s only a few seconds,
> but sometimes it seems to be more. (It’s like the recording sometimes goes
> behind the real call and when the hangup event is received the remaining
> audio is discarded.)
>
>
>
> What could be the reason for this behavior? Is there something wrong with
> the script or is there a better way to achieve our goal?
>
>
>
> One first question come to my mind: why do you use a script here? A simple
> extension can do exactly the same, if you just want to record the session...
>
> https://freeswitch.org/confluence/display/FREESWITCH/mod_
> dptools:+record_session
>
> Anyway, if you want to use the script, why you first
>
> session:setAutoHangup(false)
>
> and after that you
>
>
>
> session:hangup()
>
>
>
> ?
>
>
>
> Also, you made the silence_threshold equal 0 (zero).
> Have you has the same problems using a silence_threshold of, let's say, 30
> (thirty), like in documentation? ( https://freeswitch.org/conflue
> nce/display/FREESWITCH/Lua+API+Reference#LuaAPIReference-
> session:recordFile )
>
>
>
> Hope this helps,
>
> -giovanni
>
>
>
>
>
>
> Many thanks in advance.
>
>
>
> Best regards,
>
>
>
> Sven Uszpelkat
> Departamento I+D
> Comunycarse Network Consultants, S.L.
>
>
>
>
>
> [image: Descripción: Descripción:
> http://www.comunycarse.com/email_images/facebook_16.jpg]
> <http://www.facebook.com/pages/Comunycarse/227218309114> [image:
> Descripción: Descripción:
> http://www.comunycarse.com/email_images/linkedin_16.jpg]
> <http://www.linkedin.com/company/comunycarse-s.l.> [image: Descripción:
> Descripción: http://www.comunycarse.com/email_images/twitter_16.jpg]
> <https://twitter.com/#%21/comunycarse> [image: Descripción: Descripción:
> http://www.comunycarse.com/email_images/wordpress_16.jpg]
> <http://www.blog.comunycarse.com/>
>
> Joaquín Turina, 2
> 28224 Pozuelo de Alarcón MADRID
> Tlf. +34 917 498 700 <+34%20917%2049%2087%2000>
> Fax +34 917 498 720 <+34%20917%2049%2087%2020>
>
> Sabino Arana, 18
> 08028 BARCELONA
> Tlf. +34 934 098 480 <+34%20934%2009%2084%2080>
> Fax +34 934 098 490 <+34%20934%2009%2084%2090>
>
> http://www.comunycarse.com
>
>
> AVISO LEGAL
> La presente comunicación y sus anexos tiene como destinatario la persona a
> la que va dirigida, por lo que si usted lo recibe por error debe
> notificarlo al remitente y eliminarlo de su sistema, no pudiendo
> utilizarlo, total o parcialmente, para ningún fin. Su contenido puede tener
> información confidencial o protegida legalmente y únicamente expresa la
> opinión del remitente. El uso del correo electrónico vía internet no
> permite asegurar ni la confidencialidad de los mensajes ni su correcta
> recepción. En el caso de que el destinatario no consintiera la utilización
> del correo electrónico deberá ponerlo en nuestro conocimiento
> inmediatamente.
>
> DISCLAIMER
> This message and its attachments are intended exclusively for the named
> addressee. If you receive this message by mistake, please delete it
> immediately from your system and notify the sender. You may not use this
> message or any part of it for any purpose. The message may contain
> information that is confidential or protected by law, and any opinions
> expressed are those of the individual sender. Internet email guarantees
> neither the confidentiality nor the proper receipt of the message sent. If
> the addressee of this message does not consent to the use of internet
> e-mail, please inform us immediately.
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org
>
>
>
>
> --
>
>
> Sincerely,
>
> Giovanni Maruzzelli
> OpenTelecom.IT
> cell: +39 347 266 56 18
>



-- 

Sincerely,

Giovanni Maruzzelli
OpenTelecom.IT
cell: +39 347 266 56 18
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170615/ada8f1e3/attachment-0001.html>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: image003.jpg
Type: image/jpeg
Size: 1628 bytes
Desc: not available
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170615/ada8f1e3/attachment-0004.jpg>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: image002.jpg
Type: image/jpeg
Size: 1694 bytes
Desc: not available
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170615/ada8f1e3/attachment-0005.jpg>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: image001.jpg
Type: image/jpeg
Size: 1630 bytes
Desc: not available
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170615/ada8f1e3/attachment-0006.jpg>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: image004.jpg
Type: image/jpeg
Size: 1402 bytes
Desc: not available
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170615/ada8f1e3/attachment-0007.jpg>

From adrian.worutowicz at esifrance.net  Thu Jun 15 09:44:16 2017
From: adrian.worutowicz at esifrance.net (Adrian Worutowicz)
Date: Thu, 15 Jun 2017 11:44:16 +0200
Subject: [Freeswitch-users] Build Problem in VS2015
Message-ID: <001001d2e5bb$f4519cc0$dcf4d640$@worutowicz@esifrance.net>

Hello,

 

I try to recompile FS in VS2015 without success.

I took FS sources from git master.

 

git config --global core.autocrlf false

git clone https://stash.freeswitch.org/scm/fs/freeswitch.git
/c/ESI/Components/FreeSwitch/
<https://stash.freeswitch.org/scm/fs/freeswitch.git%20/c/ESI/Components/Free
Switch/> 

 

I have wix311 for VS2015 installed.

 

For example it searches in folder 'openssl-1.0.2k' while only a folder
'openssl' exists.

 

I tried to recompile mod_PortAudio, and I got
c:\ESI\components\freeswitch\src\mod\endpoints\mod_portaudio\pablio.h(55):
fatal error C1083: Impossible d'ouvrir le fichier include : 'portaudio.h' :
No such file or directory.

 

Indeed 'portaudio.h' does not exist.

Plenty of other errors in the attached file.

 

What do I miss?

 

Thanks in advance,

Adrian.

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170615/dd0a1c8b/attachment-0001.html>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: vs2015BuildReport.zip
Type: application/octet-stream
Size: 13747 bytes
Desc: not available
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170615/dd0a1c8b/attachment-0001.obj>

From USZPELSV at comunycarse.com  Thu Jun 15 10:50:03 2017
From: USZPELSV at comunycarse.com (Sven Uszpelkat)
Date: Thu, 15 Jun 2017 10:50:03 +0000
Subject: [Freeswitch-users] Call recording
In-Reply-To: <CALXCt0ocv6Hhfq7vfHhQf15047E9d+bWfuSE+wkLbwgLLh6Ptw@mail.gmail.com>
References: <7c47071f86ce436186b4ab8bf58de46c@w12mad02.comunycarse.com>
 <CALXCt0ocv6Hhfq7vfHhQf15047E9d+bWfuSE+wkLbwgLLh6Ptw@mail.gmail.com>
Message-ID: <5323e6cfb2844e59a19957fad4e8044c@w12mad02.comunycarse.com>

Hi Giovanni,

Thank you for your response.


1.)    The code example I provided was only a fragment of the script. The actual script is doing more than just recording but we will try the dialplan approach as well.

2.)    The session:hangup was thought to leave the while loop but maybe it’s not necessary if the session is already finished or we change the while for an if. I think I copied this from another example I found.

3.)    We set the silence threshold to 0 because the documentation is not very clear how to disable the silence detection. We don’t want the recording to stop in response to a period of silence. The point is to record everything.

Best regards,
Sven

De: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] En nombre de Giovanni Maruzzelli
Enviado el: jueves, 15 de junio de 2017 11:19
Para: FreeSWITCH Users Help
Asunto: Re: [Freeswitch-users] Call recording



On 15 June 2017 at 09:49, Sven Uszpelkat <USZPELSV at comunycarse.com<mailto:USZPELSV at comunycarse.com>> wrote:
Hello,

We are using FreeSWITCH as a third-party recording application, i.e. we are receiving SIP calls with the complete audio of conversations taking place on another switch and we are saving this audio to a file. To achieve this we are using a simple script similar to this:

session:answer()
while(session:ready() == true) do
   test = session:recordFile("/usr/local/freeswitch/recordings/test.wav", 18000, 0, 300)
   session:setAutoHangup(false)
   session:hangup()
end

This script will be invoked by the following dialplan:

<extension name="Recording">
      <condition field="destination_number" expression=".*">
            <action application="lua" data="record.lua"/>
      </condition>
</extension>

Basically it seems to work quite well, but sometimes there are missing audio at the end of the recorded file.  Usually it’s only a few seconds, but sometimes it seems to be more. (It’s like the recording sometimes goes behind the real call and when the hangup event is received the remaining audio is discarded.)

What could be the reason for this behavior? Is there something wrong with the script or is there a better way to achieve our goal?

One first question come to my mind: why do you use a script here? A simple extension can do exactly the same, if you just want to record the session...

https://freeswitch.org/confluence/display/FREESWITCH/mod_dptools:+record_session
Anyway, if you want to use the script, why you first

session:setAutoHangup(false)
and after that you

session:hangup()

?

Also, you made the silence_threshold equal 0 (zero).
Have you has the same problems using a silence_threshold of, let's say, 30 (thirty), like in documentation? ( https://freeswitch.org/confluence/display/FREESWITCH/Lua+API+Reference#LuaAPIReference-session:recordFile )

Hope this helps,
-giovanni



Many thanks in advance.

Best regards,

Sven Uszpelkat
Departamento I+D
Comunycarse Network Consultants, S.L.


[Descripción: Descripción: http://www.comunycarse.com/email_images/facebook_16.jpg]<http://www.facebook.com/pages/Comunycarse/227218309114> [Descripción: Descripción: http://www.comunycarse.com/email_images/linkedin_16.jpg] <http://www.linkedin.com/company/comunycarse-s.l.>  [Descripción: Descripción: http://www.comunycarse.com/email_images/twitter_16.jpg] <https://twitter.com/#%21/comunycarse>  [Descripción: Descripción: http://www.comunycarse.com/email_images/wordpress_16.jpg] <http://www.blog.comunycarse.com/>

Joaquín Turina, 2
28224 Pozuelo de Alarcón MADRID
Tlf. +34 917 498 700<tel:+34%20917%2049%2087%2000>
Fax +34 917 498 720<tel:+34%20917%2049%2087%2020>

Sabino Arana, 18
08028 BARCELONA
Tlf. +34 934 098 480<tel:+34%20934%2009%2084%2080>
Fax +34 934 098 490<tel:+34%20934%2009%2084%2090>

http://www.comunycarse.com<http://www.comunycarse.com/>

AVISO LEGAL
La presente comunicación y sus anexos tiene como destinatario la persona a la que va dirigida, por lo que si usted lo recibe por error debe notificarlo al remitente y eliminarlo de su sistema, no pudiendo utilizarlo, total o parcialmente, para ningún fin. Su contenido puede tener información confidencial o protegida legalmente y únicamente expresa la opinión del remitente. El uso del correo electrónico vía internet no permite asegurar ni la confidencialidad de los mensajes ni su correcta recepción. En el caso de que el destinatario no consintiera la utilización del correo electrónico deberá ponerlo en nuestro conocimiento inmediatamente.

DISCLAIMER
This message and its attachments are intended exclusively for the named addressee. If you receive this message by mistake, please delete it immediately from your system and notify the sender. You may not use this message or any part of it for any purpose. The message may contain information that is confidential or protected by law, and any opinions expressed are those of the individual sender. Internet email guarantees neither the confidentiality nor the proper receipt of the message sent. If the addressee of this message does not consent to the use of internet e-mail, please inform us immediately.


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.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
http://www.freeswitch.org



--

Sincerely,

Giovanni Maruzzelli
OpenTelecom.IT
cell: +39 347 266 56 18
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170615/67a049b7/attachment-0001.html>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: image001.jpg
Type: image/jpeg
Size: 1630 bytes
Desc: image001.jpg
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170615/67a049b7/attachment-0004.jpg>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: image002.jpg
Type: image/jpeg
Size: 1694 bytes
Desc: image002.jpg
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170615/67a049b7/attachment-0005.jpg>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: image003.jpg
Type: image/jpeg
Size: 1628 bytes
Desc: image003.jpg
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170615/67a049b7/attachment-0006.jpg>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: image004.jpg
Type: image/jpeg
Size: 1402 bytes
Desc: image004.jpg
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170615/67a049b7/attachment-0007.jpg>

From kkothari157 at gmail.com  Thu Jun 15 12:55:32 2017
From: kkothari157 at gmail.com (Ketan Kothari)
Date: Thu, 15 Jun 2017 18:25:32 +0530
Subject: [Freeswitch-users] Verto failover
In-Reply-To: <CAHtxdDd885V0+LjRB24THaka1o40dWy1x-kO2Ko_jZZ5aXiyBQ@mail.gmail.com>
References: <CAGqqW9-YHn5nW5f4OFz0=EedosPyWzcM4Y0Cd3Cn31TX6j4=aQ@mail.gmail.com>
 <CAHtxdDd885V0+LjRB24THaka1o40dWy1x-kO2Ko_jZZ5aXiyBQ@mail.gmail.com>
Message-ID: <CAGqqW9_5e2Q5NvRQcPDnTmQaHLa8qpr5kbNfv1FR5tMOEnp3KQ@mail.gmail.com>

Hello Sergey,

Thanks for your response.


*3) using nginx as proxy.*
Could you please tell me some reference to do it.


On Thu, Jun 15, 2017 at 4:46 PM, Sergey Safarov <s.safarov at gmail.com> wrote:

> Yes.
> Balancing can be done using:
> 1) at dns level;
> 2) using haproxy daemon;
> 3) using nginx as proxy.
>
> чт, 15 июн. 2017 г. в 11:41, Ketan Kothari <kkothari157 at gmail.com>:
>
>> Can we setup Verto communicator *User-Interface* and *FreeSWITCH* on
>> separate server?
>> If yes
>>
>> *--->* We have 2 FreeSWITCH servers and 1 User-Interface server. So how
>> to load balance or fail-over of Verto?
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.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
>> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170615/6e47d503/attachment-0001.html>

From rmundkowsky at ets.org  Thu Jun 15 13:51:53 2017
From: rmundkowsky at ets.org (Mundkowsky, Robert)
Date: Thu, 15 Jun 2017 13:51:53 +0000
Subject: [Freeswitch-users] Verto failover
In-Reply-To: <CAHtxdDd885V0+LjRB24THaka1o40dWy1x-kO2Ko_jZZ5aXiyBQ@mail.gmail.com>
References: <CAGqqW9-YHn5nW5f4OFz0=EedosPyWzcM4Y0Cd3Cn31TX6j4=aQ@mail.gmail.com>
 <CAHtxdDd885V0+LjRB24THaka1o40dWy1x-kO2Ko_jZZ5aXiyBQ@mail.gmail.com>
Message-ID: <BN6PR07MB3556AA8421B7AA8E8439856FBEC00@BN6PR07MB3556.namprd07.prod.outlook.com>

Verto is just frontend client side code (javascript), you can put it anywhere and point to a FreeSWITCH server anywhere else.

I am curious what type of failover is handled by Sergey’s suggestion.

Would this support an active call to continue? Or are you talking about failover for next call?


Robert Mundkowsky

From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Sergey Safarov
Sent: Thursday, June 15, 2017 7:17 AM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Verto failover

Yes.
Balancing can be done using:
1) at dns level;
2) using haproxy daemon;
3) using nginx as proxy.

чт, 15 июн. 2017 г. в 11:41, Ketan Kothari <kkothari157 at gmail.com<mailto:kkothari157 at gmail.com>>:
Can we setup Verto communicator User-Interface and FreeSWITCH on separate server?
If yes

---> We have 2 FreeSWITCH servers and 1 User-Interface server. So how to load balance or fail-over of Verto?
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com<https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.freeswitchsolutions.com&data=02%7C01%7Crmundkowsky%40ets.org%7Cad70a93ec7c5477c6e5108d4b3e0af98%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636331224986585344&sdata=rabcpCufsZMmPOC35CVZIeWm4YeP2t%2B67cYi0ND5V3s%3D&reserved=0>

Official FreeSWITCH Sites
http://www.freeswitch.org<https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.freeswitch.org&data=02%7C01%7Crmundkowsky%40ets.org%7Cad70a93ec7c5477c6e5108d4b3e0af98%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636331224986585344&sdata=rmrLXKiFmbYBsk38ZuQFYxyyYWm4mT9%2BKS2aZshgMEQ%3D&reserved=0>
http://confluence.freeswitch.org<https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fconfluence.freeswitch.org&data=02%7C01%7Crmundkowsky%40ets.org%7Cad70a93ec7c5477c6e5108d4b3e0af98%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636331224986585344&sdata=P5vBRCKyjO9am3mHuUGyAbXKopi64iTjVaoL64oGTE4%3D&reserved=0>
http://www.cluecon.com<https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.cluecon.com&data=02%7C01%7Crmundkowsky%40ets.org%7Cad70a93ec7c5477c6e5108d4b3e0af98%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636331224986585344&sdata=5wem7ZBJIWbgHc5BVkS08hI4li6dAlitksUO06zBkMU%3D&reserved=0>

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<https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Flists.freeswitch.org%2Fmailman%2Flistinfo%2Ffreeswitch-users&data=02%7C01%7Crmundkowsky%40ets.org%7Cad70a93ec7c5477c6e5108d4b3e0af98%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636331224986585344&sdata=0b9wmreHehfx%2Ft%2FxRZOxPHVqk70qdxIMkG0GPaCCDr4%3D&reserved=0>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users<https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Flists.freeswitch.org%2Fmailman%2Foptions%2Ffreeswitch-users&data=02%7C01%7Crmundkowsky%40ets.org%7Cad70a93ec7c5477c6e5108d4b3e0af98%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636331224986585344&sdata=LBMZOQVvh23pws%2FN8rGsIIdg6tVxagZIRFY1%2B8N7Uzk%3D&reserved=0>
http://www.freeswitch.org<https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.freeswitch.org&data=02%7C01%7Crmundkowsky%40ets.org%7Cad70a93ec7c5477c6e5108d4b3e0af98%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636331224986585344&sdata=rmrLXKiFmbYBsk38ZuQFYxyyYWm4mT9%2BKS2aZshgMEQ%3D&reserved=0>

________________________________

This e-mail and any files transmitted with it may contain privileged or confidential information. It is solely for use by the individual for whom it is intended, even if addressed incorrectly. If you received this e-mail in error, please notify the sender; do not disclose, copy, distribute, or take any action in reliance on the contents of this information; and delete it from your system. Any other use of this e-mail is prohibited.


Thank you for your compliance.

________________________________
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170615/df29e479/attachment-0001.html>

From soapee01.fs at stubbornroses.com  Thu Jun 15 22:06:14 2017
From: soapee01.fs at stubbornroses.com (soapee01.fs at stubbornroses.com)
Date: Thu, 15 Jun 2017 17:06:14 -0500
Subject: [Freeswitch-users] Missed Calls with Hunt Group
Message-ID: <594304D6.7070207@stubbornroses.com>

Hi,

Here's the command that I'm running.

originate user/102 at domain &bridge(user/100 at domain,user/101 at domain)


On FS version 1.2.22, if the user 100 answers the call, user 101 will 
not see a missed call notification.

On FS version FS 1.6.5 if the user 100 answers the call, user 101  will 
show a missed call.

Is there something in the docs I've missed? I'd really like to set it 
back to the old way, but it would be really cool if there's a variable 
somewhere I've missed that lets you choose the behavior.

Thanks!

James


From jungleboogie0 at gmail.com  Thu Jun 15 23:30:45 2017
From: jungleboogie0 at gmail.com (jungle Boogie)
Date: Thu, 15 Jun 2017 16:30:45 -0700
Subject: [Freeswitch-users]
	=?utf-8?q?FreeSWITCH_Week_in_Review_=28Master_?=
	=?utf-8?q?Branch=29_December_5th_=E2=80=93_December_12th?=
In-Reply-To: <5671969d94b11_2c9b52d334780b@resque-worker.8.mail>
References: <5671969d94b11_2c9b52d334780b@resque-worker.8.mail>
Message-ID: <CAKE2PDuexrLMXzHZ+AXtf6ptse52hed8KXchisJJaAwJ_Vgkag@mail.gmail.com>

Hi All,
On 16 December 2015 at 08:51, Ken Rice <krice at freeswitch.org> wrote:
> New Post on freeswitch.org from Kathleen King
> check it out at http://ift.tt/1P8dyo3
> FreeSWITCH Week in Review (Master Branch) December 5th – December 12th


What happened with these emails? They're really informative and give
people an idea on what's changing.


From brian at freeswitch.org  Fri Jun 16 00:23:30 2017
From: brian at freeswitch.org (Brian West)
Date: Thu, 15 Jun 2017 19:23:30 -0500
Subject: [Freeswitch-users] Missed Calls with Hunt Group
In-Reply-To: <594304D6.7070207@stubbornroses.com>
References: <594304D6.7070207@stubbornroses.com>
Message-ID: <CAEJMVkAjhP_kv6FmiSrwraVknuE6QzSswXOBaj15cc9umWHZVA@mail.gmail.com>

Verify the answered elsewhere header is there, Thats what makes that
happen, plus try 1.6.18.

/b


On Thu, Jun 15, 2017 at 5:06 PM, <soapee01.fs at stubbornroses.com> wrote:

> Hi,
>
> Here's the command that I'm running.
>
> originate user/102 at domain &bridge(user/100 at domain,user/101 at domain)
>
>
> On FS version 1.2.22, if the user 100 answers the call, user 101 will not
> see a missed call notification.
>
> On FS version FS 1.6.5 if the user 100 answers the call, user 101  will
> show a missed call.
>
> Is there something in the docs I've missed? I'd really like to set it back
> to the old way, but it would be really cool if there's a variable somewhere
> I've missed that lets you choose the behavior.
>
> Thanks!
>
> James
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org




-- 

*Brian West*
brian at freeswitch.org

*Twitter: @FreeSWITCH , @briankwest*

http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

Book a phone call (CST) <https://freeswitch.com/appointment>

Allison prompts for FreeSWITCH:

*https://www.gofundme.com/allison-prompts-for-freeswitch*
<https://www.gofundme.com/allison-prompts-for-freeswitch>

Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
/r/freeswitch <https://www.reddit.com/r/freeswitch>

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*Skype:*briankwest
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170615/7272589f/attachment.html>

From tculjaga at gmail.com  Fri Jun 16 08:57:14 2017
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Fri, 16 Jun 2017 10:57:14 +0200
Subject: [Freeswitch-users] WebRTC and Carrier NAT ( CgNAT)
Message-ID: <CAJoKKdU_iQigcaoiWVaHtd8Si=gfvOyJ3MEGdUSG62i3UP32rA@mail.gmail.com>

hello,

does anyone have experience with WebRTC via carrier NAT ?

client_LAN(192.168.x.y) <NAT> carrer_prvate_WAN(10.x.x.x) <NAT>
carrier_public_WAN(public_ip) <> internet


in this scenario, WebRTC calls using STUN only will not work. What about
TURN ?

did anyone try that ?

Regards,
Tihomir.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170616/9a6f0dcd/attachment.html>

From hardikitpl at gmail.com  Fri Jun 16 09:23:20 2017
From: hardikitpl at gmail.com (Hardik Patel)
Date: Fri, 16 Jun 2017 14:53:20 +0530
Subject: [Freeswitch-users] Fax receive issue with t30 codec
In-Reply-To: <CALXCt0pPS1MnPOFpvENhb0T646E7zUeTg3UzT+Kzs8Q=8tf-vA@mail.gmail.com>
References: <CAHdn4vjxr5vVoO7Oc1P0u-xg6dJNYBhjxU=y5iz6VP5_JEcUjA@mail.gmail.com>
 <CAEJMVkDtY8VeG+14cB42wHee5fjgzsXE5trA31dtp=F4rxoeRg@mail.gmail.com>
 <CAJoKKdVBBBbZSmtT20f6ANgDJnpXjV+7a1==ivsdQ5qcaN3BqA@mail.gmail.com>
 <CAEJMVkBFD6b=8porYC9Vm-si5Qy3OsDCNKQrmOBDWmm4A-qTEA@mail.gmail.com>
 <CAHdn4vgRHZ5zNFLQYhDkUfrMGDNzqnEwQdQ3coORHfqQmOMOQA@mail.gmail.com>
 <CAHdn4vhNkD9naQ9dOh5NLFEjgcemeyr6PqYU1dAsUMQGmMte9A@mail.gmail.com>
 <CALXCt0pPS1MnPOFpvENhb0T646E7zUeTg3UzT+Kzs8Q=8tf-vA@mail.gmail.com>
Message-ID: <CAHdn4vhesgQNwtaOQJ3u9VkP=TEwL7WqLevnP13c3EQCcmnp_A@mail.gmail.com>

Hi All

Finally.......!!!  It works for me and it's nice experience to troubleshoot
on it.


Thank you very much you all who support me on this.

Thanks a lot.


On Wed, Jun 14, 2017 at 5:28 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>
wrote:

> try setting the transfer to 9600 (fax_disable_v17)
>
> Also, you may find this page useful:
>
> https://freeswitch.org/confluence/display/FREESWITCH/
> mod_spandsp#mod_spandsp-Fax
>
>
>
> On 14 June 2017 at 13:52, Hardik Patel <hardikitpl at gmail.com> wrote:
>
>> Hi Brian,
>>
>> Is there any solution for that?
>>
>> On Sat, Jun 10, 2017 at 9:46 AM, Hardik Patel <hardikitpl at gmail.com>
>> wrote:
>>
>>> Hi Brian,
>>>
>>> Thanks for the support.
>>>
>>> We are testing receive fax functionality using real fax machine and here
>>> i have listed the model which we are using to send fax.
>>>
>>> the models of real fax machines that we have used are group 3 CCITT /
>>> ITU, they are the following:
>>> 1 ) konica minolta bizhub-c220
>>> 2 ) HP Officejet 4500
>>> 3 ) HP Officejet G85
>>>
>>>
>>> From above list one of our machine is sending fax without T38 support
>>> and we got the failure with same error which have posted on bug but if we
>>> use T38 support then it works fine for us.
>>>
>>>
>>>
>>> *ERROR :*
>>> 2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:508 Fax processing
>>> not successful - result (3) Timed out waiting for the first message.
>>> 2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:513 Remote station
>>> id:
>>> 2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:514 Local station
>>> id:  SpanDSP Fax Ident
>>> 2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:515 Pages
>>> transferred: 0
>>> 2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:517 Total fax
>>> pages:   0
>>> 2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:518 Image
>>> resolution:  0x0
>>> 2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:519 Transfer
>>> Rate:     14400
>>> 2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:521 ECM
>>> status         off
>>> 2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:522 remote
>>> country:
>>> 2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:523 remote
>>> vendor:
>>> 2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:524 remote
>>> model:
>>> 2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:526
>>> ==============================
>>>
>>> On Sat, Jun 10, 2017 at 2:42 AM, Brian West <brian at freeswitch.org>
>>> wrote:
>>>
>>>> Your log snip doesn't really help, I know without a single doubt faxing
>>>> works fine.  So what are you doing and how are you doing it?
>>>>
>>>> On Fri, Jun 9, 2017 at 3:42 PM, Tihomir Culjaga <tculjaga at gmail.com>
>>>> wrote:
>>>>
>>>>> hi, whats new on faxing in 1.6.17  ?
>>>>>
>>>>> T.
>>>>>
>>>>> On 6 June 2017 at 16:00, Brian West <brian at freeswitch.org> wrote:
>>>>>
>>>>>> You'll have to use 1.6.17 if you ever want any faxing to work in all
>>>>>> test cases.
>>>>>>
>>>>>> /b
>>>>>>
>>>>>> On Tue, Jun 6, 2017 at 6:11 AM, Hardik Patel <hardikitpl at gmail.com>
>>>>>> wrote:
>>>>>>
>>>>>>> Hello,
>>>>>>>
>>>>>>> I am using opensips as entry point using dispatcher. opensips(
>>>>>>> 127.0.0.1), i am routing call to freeswitch server (127.0.0.3).
>>>>>>>
>>>>>>> Now I am trying to receive fax, my issue is when i try to send fax
>>>>>>> in softphone(Zoiper) from the log i am seeing that it is sending fax using
>>>>>>> t30 codec. and i am not receiving the fax at destination, is it because of
>>>>>>> codec, should it only work with t38 codec? if that is the issue than how am
>>>>>>> i be able to send the fax using t38 from zoiper?
>>>>>>>
>>>>>>> Here i am attaching the fs log with loglevel 9 and sip trace is also
>>>>>>> enabled.
>>>>>>>
>>>>>>> 127.0.0.2  => carrier/provider IP
>>>>>>> 123456789 => Fax number
>>>>>>> test at gamil.com => Email Address
>>>>>>> 127.0.0.4 =>UI IP
>>>>>>>
>>>>>>> Pastebin link:https://pastebin.freeswitch.org/view/9ec52715
>>>>>>>
>>>>>>> ____________________________________________________________
>>>>>>> _____________
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>
>>>>>>> Official FreeSWITCH Sites
>>>>>>> http://www.freeswitch.org
>>>>>>> http://confluence.freeswitch.org
>>>>>>> http://www.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/free
>>>>>>> switch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> --
>>>>>>
>>>>>> *Brian West*
>>>>>> brian at freeswitch.org
>>>>>>
>>>>>> *Twitter: @FreeSWITCH , @briankwest*
>>>>>>
>>>>>> http://www.freeswitchbook.com
>>>>>> http://www.freeswitchcookbook.com
>>>>>>
>>>>>> Book a phone call (CST) <https://freeswitch.com/appointment>
>>>>>>
>>>>>> Allison prompts for FreeSWITCH:
>>>>>>
>>>>>> *https://www.gofundme.com/allison-prompts-for-freeswitch*
>>>>>> <https://www.gofundme.com/allison-prompts-for-freeswitch>
>>>>>>
>>>>>> Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
>>>>>> /r/freeswitch <https://www.reddit.com/r/freeswitch>
>>>>>>
>>>>>> *T:*+19184209001 <+1%20918-420-9001> | *F:*+19184209002
>>>>>> <+1%20918-420-9002> | *M:*+1918424WEST (9378)
>>>>>> *Skype:*briankwest
>>>>>>
>>>>>> ____________________________________________________________
>>>>>> _____________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://confluence.freeswitch.org
>>>>>> http://www.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/free
>>>>>> switch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>> ____________________________________________________________
>>>>> _____________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.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/free
>>>>> switch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>>
>>>> *Brian West*
>>>> brian at freeswitch.org
>>>>
>>>> *Twitter: @FreeSWITCH , @briankwest*
>>>>
>>>> http://www.freeswitchbook.com
>>>> http://www.freeswitchcookbook.com
>>>>
>>>> Book a phone call (CST) <https://freeswitch.com/appointment>
>>>>
>>>> Allison prompts for FreeSWITCH:
>>>>
>>>> *https://www.gofundme.com/allison-prompts-for-freeswitch*
>>>> <https://www.gofundme.com/allison-prompts-for-freeswitch>
>>>>
>>>> Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
>>>> /r/freeswitch <https://www.reddit.com/r/freeswitch>
>>>>
>>>> *T:*+19184209001 <(918)%20420-9001> | *F:*+19184209002
>>>> <(918)%20420-9002> | *M:*+1918424WEST (9378)
>>>> *Skype:*briankwest
>>>>
>>>> ____________________________________________________________
>>>> _____________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.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/free
>>>> switch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> --
>>> Hardik Patel
>>> iNextrix Technologies Pvt Ltd
>>>
>>
>>
>>
>> --
>> Hardik Patel
>> iNextrix Technologies Pvt Ltd
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.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
>> http://www.freeswitch.org
>>
>
>
>
> --
>
> Sincerely,
>
> Giovanni Maruzzelli
> OpenTelecom.IT
> cell: +39 347 266 56 18
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org
>



-- 
Hardik Patel
iNextrix Technologies Pvt Ltd
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170616/1026a452/attachment-0001.html>

From kkothari157 at gmail.com  Fri Jun 16 07:02:39 2017
From: kkothari157 at gmail.com (Ketan Kothari)
Date: Fri, 16 Jun 2017 12:32:39 +0530
Subject: [Freeswitch-users] Verto failover
In-Reply-To: <BN6PR07MB3556AA8421B7AA8E8439856FBEC00@BN6PR07MB3556.namprd07.prod.outlook.com>
References: <CAGqqW9-YHn5nW5f4OFz0=EedosPyWzcM4Y0Cd3Cn31TX6j4=aQ@mail.gmail.com>
 <CAHtxdDd885V0+LjRB24THaka1o40dWy1x-kO2Ko_jZZ5aXiyBQ@mail.gmail.com>
 <BN6PR07MB3556AA8421B7AA8E8439856FBEC00@BN6PR07MB3556.namprd07.prod.outlook.com>
Message-ID: <CAGqqW98aEn_LGs=hvPmPqXkPTYpLyPoS1tEtz-+af714yJHnjw@mail.gmail.com>

Hello Robert,

I'm looking for fail-over for next call and web-interface as well as if one
FreeSWITCH failed then verto continue working from other FreeSWITCH server.

If you have any suggestion or reference link then please pass it will
helpful for me.



On Thu, Jun 15, 2017 at 7:21 PM, Mundkowsky, Robert <rmundkowsky at ets.org>
wrote:

> Verto is just frontend client side code (javascript), you can put it
> anywhere and point to a FreeSWITCH server anywhere else.
>
>
>
> I am curious what type of failover is handled by Sergey’s suggestion.
>
>
>
> Would this support an active call to continue? Or are you talking about
> failover for next call?
>
>
>
>
>
> Robert Mundkowsky
>
>
>
> *From:* FreeSWITCH-users [mailto:freeswitch-users-
> bounces at lists.freeswitch.org] *On Behalf Of *Sergey Safarov
> *Sent:* Thursday, June 15, 2017 7:17 AM
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Subject:* Re: [Freeswitch-users] Verto failover
>
>
>
> Yes.
> Balancing can be done using:
>
> 1) at dns level;
>
> 2) using haproxy daemon;
>
> 3) using nginx as proxy.
>
>
>
> чт, 15 июн. 2017 г. в 11:41, Ketan Kothari <kkothari157 at gmail.com>:
>
> Can we setup Verto communicator *User-Interface* and *FreeSWITCH* on
> separate server?
> If yes
>
> *--->* We have 2 FreeSWITCH servers and 1 User-Interface server. So how
> to load balance or fail-over of Verto?
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.freeswitchsolutions.com&data=02%7C01%7Crmundkowsky%40ets.org%7Cad70a93ec7c5477c6e5108d4b3e0af98%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636331224986585344&sdata=rabcpCufsZMmPOC35CVZIeWm4YeP2t%2B67cYi0ND5V3s%3D&reserved=0>
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.freeswitch.org&data=02%7C01%7Crmundkowsky%40ets.org%7Cad70a93ec7c5477c6e5108d4b3e0af98%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636331224986585344&sdata=rmrLXKiFmbYBsk38ZuQFYxyyYWm4mT9%2BKS2aZshgMEQ%3D&reserved=0>
> http://confluence.freeswitch.org
> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fconfluence.freeswitch.org&data=02%7C01%7Crmundkowsky%40ets.org%7Cad70a93ec7c5477c6e5108d4b3e0af98%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636331224986585344&sdata=P5vBRCKyjO9am3mHuUGyAbXKopi64iTjVaoL64oGTE4%3D&reserved=0>
> http://www.cluecon.com
> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.cluecon.com&data=02%7C01%7Crmundkowsky%40ets.org%7Cad70a93ec7c5477c6e5108d4b3e0af98%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636331224986585344&sdata=5wem7ZBJIWbgHc5BVkS08hI4li6dAlitksUO06zBkMU%3D&reserved=0>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Flists.freeswitch.org%2Fmailman%2Flistinfo%2Ffreeswitch-users&data=02%7C01%7Crmundkowsky%40ets.org%7Cad70a93ec7c5477c6e5108d4b3e0af98%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636331224986585344&sdata=0b9wmreHehfx%2Ft%2FxRZOxPHVqk70qdxIMkG0GPaCCDr4%3D&reserved=0>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Flists.freeswitch.org%2Fmailman%2Foptions%2Ffreeswitch-users&data=02%7C01%7Crmundkowsky%40ets.org%7Cad70a93ec7c5477c6e5108d4b3e0af98%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636331224986585344&sdata=LBMZOQVvh23pws%2FN8rGsIIdg6tVxagZIRFY1%2B8N7Uzk%3D&reserved=0>
> http://www.freeswitch.org
> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.freeswitch.org&data=02%7C01%7Crmundkowsky%40ets.org%7Cad70a93ec7c5477c6e5108d4b3e0af98%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636331224986585344&sdata=rmrLXKiFmbYBsk38ZuQFYxyyYWm4mT9%2BKS2aZshgMEQ%3D&reserved=0>
>
>
> ------------------------------
>
> This e-mail and any files transmitted with it may contain privileged or
> confidential information. It is solely for use by the individual for whom
> it is intended, even if addressed incorrectly. If you received this e-mail
> in error, please notify the sender; do not disclose, copy, distribute, or
> take any action in reliance on the contents of this information; and delete
> it from your system. Any other use of this e-mail is prohibited.
>
> Thank you for your compliance.
> ------------------------------
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170616/649841f8/attachment.html>

From eastour at 163.com  Fri Jun 16 04:02:14 2017
From: eastour at 163.com (chenyzhi)
Date: Fri, 16 Jun 2017 12:02:14 +0800 (CST)
Subject: [Freeswitch-users] When calling out ,
 freeswitch doesn't get DTMF ?
In-Reply-To: <CALXCt0oRdP=8M_vbQpevzUs=NAO-rDVx0BqXJzv6mhy9Q5vZkg@mail.gmail.com>
References: <f823cdd8-a354-3698-d8c5-484b1b4cc8b4@xbipin.com>
 <CAEJMVkDDQX=Ni3AOQxQOo=joGoAeYroxioVP=cu9w6Y2m7kLPA@mail.gmail.com>
 <15c898b5d58.2779.b07ebdf329620b8089087c7205b03f01@xbipin.com>
 <310cf71c.8fc3.15c8ba9c4f6.Coremail.eastour@163.com>
 <CAFGRPVqFA+tpaOQkAGGCR7s5gY0z0TbdArWFdkfnsaaG4Z9Dgw@mail.gmail.com>
 <2a18422f.ef8.15c99b7b9bb.Coremail.eastour@163.com>
 <CALXCt0pKKEJoSeXXZ=tn04nifEZvQCRSFCDKL8jSu+pfOgvRNQ@mail.gmail.com>
 <6c11dd0f.c2c1.15c9c727dc7.Coremail.eastour@163.com>
 <CAFLo2QwwwE6xeXBN9DkM1VEecEJpwnDMmQ7iT9EvSyx9RqE9mg@mail.gmail.com>
 <2f48505d.d9c5.15ca1d28424.Coremail.eastour@163.com>
 <59401B92020000310000A34D@mail.tedssupply.com>
 <CALXCt0qw4DCJ2Q0cF+kVEOpjCjp2T8eFgWbYx03sN=z5APBGGQ@mail.gmail.com>
 <59412DB4020000310000A381@mail.tedssupply.com>
 <CALXCt0rTBTBTfT+e1t0UCfpwmWyOhz1-yeQ2s6T8mSTYAnS-+w@mail.gmail.com>
 <7d1e3f12.4750.15ca994d89d.Coremail.eastour@163.com>
 <CALXCt0oRdP=8M_vbQpevzUs=NAO-rDVx0BqXJzv6mhy9Q5vZkg@mail.gmail.com>
Message-ID: <7611bb49.72be.15caf10d2a0.Coremail.eastour@163.com>

Yes ,I can hear all the IVR prompt voices correctly.


I don't think it's  a NAT problem ,because both the x-lite and the freeswitch are in the same LAN.


The sip trace log is in the attatchment. Thank you.


PS  I tested this on another freeswitch box ,version:
FreeSWITCH Version 1.6.16+git~20170403T142423Z~e6d643b29c~32bit (git e6d643b 2017-04-03 14:24:23Z 32bit)
It can detect dtmf on outgoing calls.  Maybe this only happens on FreeSWITCH Version 1.9.0+git~20170518T231917Z~a1fc18aee5~32bit (git a1fc18a 2017-05-18 23:19:17Z 32bit)






在 2017-06-15 17:04:48，"Giovanni Maruzzelli" <gmaruzz at gmail.com> 写道：





On 15 June 2017 at 04:28, chenyzhi <eastour at 163.com> wrote:

the version is:   FreeSWITCH Version 1.9.0+git~20170518T231917Z~a1fc18aee5~32bit (git a1fc18a 2017-05-18 23:19:17Z 32bit)


 the complete, since beginning to end, unedited, debug output of console when making a outgoing  call which does not get DTMFs


and the whole conf folder is in the attatchment.




from the log, seems it does not read all the IVR messages correctly to you, it exits straight away...


are you able to correctly hear all the IVR messages?


also, can you take a SIP trace? (from console: "sofia global siptrace on")


I suspect you have a NAT problem of some sort


Also, I see you are on MASTER git, on Windows, and on 32 bit... Not sure this is supported... Have you has this problems with stable branch (1.6.x)?






 
thx!






在 2017-06-15 00:42:11，"Giovanni Maruzzelli" <gmaruzz at gmail.com> 写道：





On 14 June 2017 at 18:36, admin <admin at tedssupply.com> wrote:


I don't mean to hijack the OP concern with my problem, I just wanted to reinforce I have seen this problem.  And the problem is that the call originated through FS appears not to send user entered DTMF to the receiving phone.  Let's help chenyzhi and then I'll take a turn.




I just tested right now from console with:




 bgapi originate user/1011 5000

and

originate user/1011 5000




and it works





Please pastebin your FreeSWITCH version (eg, type "version" in console), your dialplan, your SIP profile, and the complete, since beginning to end, unedited, debug output of console when receiving a call which does not get DTMFs




Maybe helping you we'll help him too

 




- James

>>> Giovanni Maruzzelli <gmaruzz at gmail.com> 6/14/2017 03:53 AM >>>


Never heard such problems




Please pastebin your dialplan, your SIP profile, and the complete, since beginning to end, unedited, debug output of console when receiving a call which does not get DTMFs










sent from mobile
cell: +39 347 266 56 18
Giovanni Maruzzelli
OpenTelecom.IT




On Jun 13, 2017 23:07, "admin" <admin at tedssupply.com> wrote:


I am encountering the same issue. I am using an ESL call to api originate, with little else except getting in and out of the port, and the caller DTMF is failing to reach the called number. The called number auto answer attendant does not respond to DTMF, and a call to a test phone confirms no DTMF. This is a new upgrade from 1.2 to 1.6 and I don't know if this was an issue in 1.2, but my users never complained before 1.6 upgrade. Ideas?...



>>> chenyzhi <eastour at 163.com> 06/13/17 3:32 PM >>>


It's a one leg call .there is no b-leg.




please make a test on your freeswitch box .




just type the command "originate user/1001 5000" on the freeswitch console to see if your freeswitch instance can detect dtmf input.













At 2017-06-12 22:57:17, "Luis Daniel Lucio Quiroz" <luis.daniel.lucio at gmail.com> wrote:


check if you have transcoding, and if you do, check that dftm type-codec on leg b are compatible.




--

Luis Daniel Lucio Quiroz
CISSP, CISM, CISA
Linux, VoIP and much more fun
www.okay.com.mx

Need LCR? Check out LCR for FusionPBX with FreeSWITCH
Need Billing? Check out Billing for FusionPBX with FreeSWITCH




On Mon, Jun 12, 2017 at 9:16 AM, chenyzhi <eastour at 163.com> wrote:


I don't think It's the DTMF type , because when I dial 5000 from x-lite (which has registered to freeswitch as 1001) ,I can hear the voice and if I press any dtmf on x-lite, freeswitch can recieve the dtmfs. This means that the DTMF type is correct ,otherwise freeswitch coudn't have received the dtmfs;




when I enter the command "originate user/1001 5000" at the freeswitch console , my xlite will ring ,and I answered ,I can hear the voice ,I press some dtmf,but freeswitch can NOT receive any dtmf. really weird.













At 2017-06-12 17:37:49, "Giovanni Maruzzelli" <gmaruzz at gmail.com> wrote:


you need to check the DTMF type, you probably are using the wrong one (info-inband-rfc2833), and for some reason they are not negotiated





On 12 June 2017 at 02:32, chenyzhi <eastour at 163.com> wrote:


I have read the logs ,but I didn't find any difference.




Please make a test to see if this happens in your box.













At 2017-06-10 06:49:34, "David Villasmil" <david.villasmil.work at gmail.com> wrote:


Have you looked at the log? Bump the logging up and see what shows up... what you're seeing is very weird

David


On Fri, Jun 9, 2017 at 11:18 AM chenyzhi <eastour at 163.com> wrote:


Hi




call 5000 from x-lite , you can hear the IVR voice and if you press dtmf keys , freeswitch can receive the dtmf keys .




but , if you enter the command "originate user/1001 5000" , x-lite will ring ,answer it ,you can hear the IVR voice , press some keys ,the freeswitch can NOT receive any dtmf ,




why?








_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.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
http://www.freeswitch.org









_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.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
http://www.freeswitch.org





--



Sincerely,

Giovanni Maruzzelli
OpenTelecom.IT
cell: +39 347 266 56 18










_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.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
http://www.freeswitch.org













_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.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
http://www.freeswitch.org



_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.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
http://www.freeswitch.org




--


Sincerely,

Giovanni Maruzzelli
OpenTelecom.IT
cell: +39 347 266 56 18





 


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.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
http://www.freeswitch.org




--


Sincerely,

Giovanni Maruzzelli
OpenTelecom.IT
cell: +39 347 266 56 18
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170616/3fa9acd9/attachment-0001.html>
-------------- next part --------------
An embedded and charset-unspecified text was scrubbed...
Name: sip-trace-on-log-calling-out-not-receiving-dtmf-log.txt
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170616/3fa9acd9/attachment-0001.txt>

From krice at tollfreegateway.com  Thu Jun 15 16:28:59 2017
From: krice at tollfreegateway.com (krice at tollfreegateway.com)
Date: Thu, 15 Jun 2017 11:28:59 -0500
Subject: [Freeswitch-users] Build Problem in VS2015
In-Reply-To: <001001d2e5bb$f4519cc0$dcf4d640$@worutowicz@esifrance.net>
References: <001001d2e5bb$f4519cc0$dcf4d640$@worutowicz@esifrance.net>
Message-ID: <80c901d2e5f4$7e26f930$7a74eb90$@tollfreegateway.com>

Not sure whats you are doing incorrect here, but I have just built master, I
use the built in git bits with VS2015, and then drop to a command prompt
(via the team explorer tab, select branches, right click the repo and select
open command prompt)

 

Then git pull, git clean -fdx, git reset -hard origin/master , git pull

 

>From here back to the solution explorer open the FreeSWITCH.2015 solution
file and build as normal.

 

I think you have something skewed there old ssl vs new ssl bits 

 

From: FreeSWITCH-users
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Adrian
Worutowicz
Sent: Thursday, June 15, 2017 4:44 AM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Build Problem in VS2015

 

Hello,

 

I try to recompile FS in VS2015 without success.

I took FS sources from git master.

 

git config --global core.autocrlf false

git clone https://stash.freeswitch.org/scm/fs/freeswitch.git
/c/ESI/Components/FreeSwitch/
<https://stash.freeswitch.org/scm/fs/freeswitch.git%20/c/ESI/Components/Free
Switch/> 

 

I have wix311 for VS2015 installed.

 

For example it searches in folder 'openssl-1.0.2k' while only a folder
'openssl' exists.

 

I tried to recompile mod_PortAudio, and I got
c:\ESI\components\freeswitch\src\mod\endpoints\mod_portaudio\pablio.h(55):
fatal error C1083: Impossible d'ouvrir le fichier include : 'portaudio.h' :
No such file or directory.

 

Indeed 'portaudio.h' does not exist.

Plenty of other errors in the attached file.

 

What do I miss?

 

Thanks in advance,

Adrian.

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170615/49102304/attachment-0001.html>

From USZPELSV at comunycarse.com  Fri Jun 16 07:53:30 2017
From: USZPELSV at comunycarse.com (Sven Uszpelkat)
Date: Fri, 16 Jun 2017 07:53:30 +0000
Subject: [Freeswitch-users] Call recording
In-Reply-To: <CALXCt0ohebExQsNw+VCB0sPT-7hkRQHAw_pS9dMsOD0cFtxriw@mail.gmail.com>
References: <7c47071f86ce436186b4ab8bf58de46c@w12mad02.comunycarse.com>
 <CALXCt0ocv6Hhfq7vfHhQf15047E9d+bWfuSE+wkLbwgLLh6Ptw@mail.gmail.com>
 <5323e6cfb2844e59a19957fad4e8044c@w12mad02.comunycarse.com>
 <CALXCt0ohebExQsNw+VCB0sPT-7hkRQHAw_pS9dMsOD0cFtxriw@mail.gmail.com>
Message-ID: <e9ad96d9cb344b128e04bb768be6fc83@w12mad02.comunycarse.com>

Hi Giovanni,

Thank you for your help. We changed our script to something like this:

session:answer()
while(session:ready() == true) do
   test = session:recordFile("/usr/local/freeswitch/recordings/test.wav", 18000, 0, 300)
   os.execute("sync");
   session:setAutoHangup(false)
   session:hangup()
end

As result we note an improvement primarily in short recordings (<1:30 min). These are now practically all complete. In longer recordings there are still losses and it seems that they are increasing with the duration of the recording. I’m not sure how to interpret this but to me it looks like this: with the sync call we achieved to write the buffer content to the file, however in longer calls there is remaining audio which hasn’t even been read to the buffer. Is that correct? If so then the recording function doesn’t ensure to read the remaining audio after hangup. (With tcdump we checked that all audio packets arrived before the BYE message) Does this mean that this behavior is by design?

Best regards,
Sven

De: Giovanni Maruzzelli [mailto:gmaruzz at gmail.com]
Enviado el: jueves, 15 de junio de 2017 13:17
Para: Sven Uszpelkat
CC: FreeSWITCH Users Help
Asunto: Re: [Freeswitch-users] Call recording



On 15 June 2017 at 12:50, Sven Uszpelkat <USZPELSV at comunycarse.com<mailto:USZPELSV at comunycarse.com>> wrote:


3.)    We set the silence threshold to 0 because the documentation is not very clear how to disable the silence detection. We don’t want the recording to stop in response to a period of silence. The point is to record everything.

The silence_threshold determines what is considered silence, eg below what level of acoustic energy we state the stream is containing silence. Then, we wait for "how_many_silence_seconds" or until hangup before stopping recording.
So, maybe you are right, and setting it to 0 will consider silence only when there is absolute silence in the stream, so for all practical purposes, until hangup. I have no mean to check source code now.
On another hand, I can think at other possible causes for the premature end of the recorded file: maybe you move the file before it has been flushed by FreeSWITCH or by operating system? Maybe the hangup in the script interrupts the recording in the script and close the file descriptor before is flushed? (I am shooting in the dark)
You can try to leave out those two lines, and test again.
Also, you can insert a line that sync (flush) the filesystem before exiting, just to be sure.
I would insert it after the while(session:ready())
A system(sync), or something similar will probably do.

Hope this helps,
-giovanni






Best regards,
Sven

De: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] En nombre de Giovanni Maruzzelli
Enviado el: jueves, 15 de junio de 2017 11:19
Para: FreeSWITCH Users Help
Asunto: Re: [Freeswitch-users] Call recording



On 15 June 2017 at 09:49, Sven Uszpelkat <USZPELSV at comunycarse.com<mailto:USZPELSV at comunycarse.com>> wrote:
Hello,

We are using FreeSWITCH as a third-party recording application, i.e. we are receiving SIP calls with the complete audio of conversations taking place on another switch and we are saving this audio to a file. To achieve this we are using a simple script similar to this:

session:answer()
while(session:ready() == true) do
   test = session:recordFile("/usr/local/freeswitch/recordings/test.wav", 18000, 0, 300)
   session:setAutoHangup(false)
   session:hangup()
end

This script will be invoked by the following dialplan:

<extension name="Recording">
      <condition field="destination_number" expression=".*">
            <action application="lua" data="record.lua"/>
      </condition>
</extension>

Basically it seems to work quite well, but sometimes there are missing audio at the end of the recorded file.  Usually it’s only a few seconds, but sometimes it seems to be more. (It’s like the recording sometimes goes behind the real call and when the hangup event is received the remaining audio is discarded.)

What could be the reason for this behavior? Is there something wrong with the script or is there a better way to achieve our goal?

One first question come to my mind: why do you use a script here? A simple extension can do exactly the same, if you just want to record the session...

https://freeswitch.org/confluence/display/FREESWITCH/mod_dptools:+record_session
Anyway, if you want to use the script, why you first

session:setAutoHangup(false)
and after that you

session:hangup()

?

Also, you made the silence_threshold equal 0 (zero).
Have you has the same problems using a silence_threshold of, let's say, 30 (thirty), like in documentation? ( https://freeswitch.org/confluence/display/FREESWITCH/Lua+API+Reference#LuaAPIReference-session:recordFile )

Hope this helps,
-giovanni



Many thanks in advance.

Best regards,

Sven Uszpelkat
Departamento I+D
Comunycarse Network Consultants, S.L.


[Descripción: Descripción: http://www.comunycarse.com/email_images/facebook_16.jpg]<http://www.facebook.com/pages/Comunycarse/227218309114> [Descripción: Descripción: http://www.comunycarse.com/email_images/linkedin_16.jpg] <http://www.linkedin.com/company/comunycarse-s.l.>  [Descripción: Descripción: http://www.comunycarse.com/email_images/twitter_16.jpg] <https://twitter.com/#%21/comunycarse>  [Descripción: Descripción: http://www.comunycarse.com/email_images/wordpress_16.jpg] <http://www.blog.comunycarse.com/>

Joaquín Turina, 2
28224 Pozuelo de Alarcón MADRID
Tlf. +34 917 498 700<tel:+34%20917%2049%2087%2000>
Fax +34 917 498 720<tel:+34%20917%2049%2087%2020>

Sabino Arana, 18
08028 BARCELONA
Tlf. +34 934 098 480<tel:+34%20934%2009%2084%2080>
Fax +34 934 098 490<tel:+34%20934%2009%2084%2090>

http://www.comunycarse.com<http://www.comunycarse.com/>

AVISO LEGAL
La presente comunicación y sus anexos tiene como destinatario la persona a la que va dirigida, por lo que si usted lo recibe por error debe notificarlo al remitente y eliminarlo de su sistema, no pudiendo utilizarlo, total o parcialmente, para ningún fin. Su contenido puede tener información confidencial o protegida legalmente y únicamente expresa la opinión del remitente. El uso del correo electrónico vía internet no permite asegurar ni la confidencialidad de los mensajes ni su correcta recepción. En el caso de que el destinatario no consintiera la utilización del correo electrónico deberá ponerlo en nuestro conocimiento inmediatamente.

DISCLAIMER
This message and its attachments are intended exclusively for the named addressee. If you receive this message by mistake, please delete it immediately from your system and notify the sender. You may not use this message or any part of it for any purpose. The message may contain information that is confidential or protected by law, and any opinions expressed are those of the individual sender. Internet email guarantees neither the confidentiality nor the proper receipt of the message sent. If the addressee of this message does not consent to the use of internet e-mail, please inform us immediately.


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.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
http://www.freeswitch.org



--

Sincerely,

Giovanni Maruzzelli
OpenTelecom.IT
cell: +39 347 266 56 18



--

Sincerely,

Giovanni Maruzzelli
OpenTelecom.IT
cell: +39 347 266 56 18
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170616/1f08947b/attachment-0001.html>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: image001.jpg
Type: image/jpeg
Size: 1630 bytes
Desc: image001.jpg
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170616/1f08947b/attachment-0004.jpg>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: image002.jpg
Type: image/jpeg
Size: 1694 bytes
Desc: image002.jpg
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170616/1f08947b/attachment-0005.jpg>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: image003.jpg
Type: image/jpeg
Size: 1628 bytes
Desc: image003.jpg
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170616/1f08947b/attachment-0006.jpg>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: image004.jpg
Type: image/jpeg
Size: 1402 bytes
Desc: image004.jpg
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170616/1f08947b/attachment-0007.jpg>

From admin at tedssupply.com  Fri Jun 16 14:24:12 2017
From: admin at tedssupply.com (admin)
Date: Fri, 16 Jun 2017 10:24:12 -0400
Subject: [Freeswitch-users] When calling out ,
 freeswitch doesn't get DTMF ?
In-Reply-To: <CALXCt0oRdP=8M_vbQpevzUs=NAO-rDVx0BqXJzv6mhy9Q5vZkg@mail.gmail.com>
References: <f823cdd8-a354-3698-d8c5-484b1b4cc8b4@xbipin.com>
 <CAEJMVkDDQX=Ni3AOQxQOo=joGoAeYroxioVP=cu9w6Y2m7kLPA@mail.gmail.com>
 <15c898b5d58.2779.b07ebdf329620b8089087c7205b03f01@xbipin.com>
 <310cf71c.8fc3.15c8ba9c4f6.Coremail.eastour@163.com>
 <CAFGRPVqFA+tpaOQkAGGCR7s5gY0z0TbdArWFdkfnsaaG4Z9Dgw@mail.gmail.com>
 <2a18422f.ef8.15c99b7b9bb.Coremail.eastour@163.com>
 <CALXCt0pKKEJoSeXXZ=tn04nifEZvQCRSFCDKL8jSu+pfOgvRNQ@mail.gmail.com>
 <6c11dd0f.c2c1.15c9c727dc7.Coremail.eastour@163.com>
 <CAFLo2QwwwE6xeXBN9DkM1VEecEJpwnDMmQ7iT9EvSyx9RqE9mg@mail.gmail.com>
 <2f48505d.d9c5.15ca1d28424.Coremail.eastour@163.com>
 <59401B92020000310000A34D@mail.tedssupply.com>
 <CALXCt0qw4DCJ2Q0cF+kVEOpjCjp2T8eFgWbYx03sN=z5APBGGQ@mail.gmail.com>
 <59412DB4020000310000A381@mail.tedssupply.com>
 <CALXCt0rTBTBTfT+e1t0UCfpwmWyOhz1-yeQ2s6T8mSTYAnS-+w@mail.gmail.com>
 <7d1e3f12.4750.15ca994d89d.Coremail.eastour@163.com>
 <CALXCt0oRdP=8M_vbQpevzUs=NAO-rDVx0BqXJzv6mhy9Q5vZkg@mail.gmail.com>
Message-ID: <5943B1CC020000310000A3F1@mail.tedssupply.com>


I am finally onsite and have prepared the logs requested, but I am
having a bit of trouble with Pastebin.  Is there a max line count?  I
logged a good call, bad call, good sip trace, bad sip trace and
including dialplan is about 17,000 lines -- too many?  May just be a
temp problem with server, will try again later. 

Thanks - James

>>> Giovanni Maruzzelli <gmaruzz at gmail.com> 6/15/2017 05:04 AM >>>





On 15 June 2017 at 04:28, chenyzhi <eastour at 163.com> wrote:



the version is: FreeSWITCH Version
1.9.0+git~20170518T231917Z~a1fc18aee5~32bit (git a1fc18a 2017-05-18
23:19:17Z 32bit) 



the complete, since beginning to end, unedited, debug output of console
when making a outgoing call which does not get DTMFs 



and the whole conf folder is in the attatchment. 


 



from the log, seems it does not read all the IVR messages correctly to
you, it exits straight away...


are you able to correctly hear all the IVR messages?


also, can you take a SIP trace? (from console: "sofia global siptrace
on")


I suspect you have a NAT problem of some sort


Also, I see you are on MASTER git, on Windows, and on 32 bit... Not
sure this is supported... Have you has this problems with stable branch
(1.6.x)?










thx! 









在 2017-06-15 00:42:11，"Giovanni Maruzzelli" <gmaruzz at gmail.com> 写道：







On 14 June 2017 at 18:36, admin <admin at tedssupply.com> wrote:



I don't mean to hijack the OP concern with my problem, I just wanted to
reinforce I have seen this problem. And the problem is that the call
originated through FS appears not to send user entered DTMF to the
receiving phone. Let's help chenyzhi and then I'll take a turn. 
 



I just tested right now from console with: 


bgapi originate user/1011 5000 
and 
originate user/1011 5000 


and it works



Please pastebin your FreeSWITCH version (eg, type "version" in
console), your dialplan, your SIP profile, and the complete, since
beginning to end, unedited, debug output of console when receiving a
call which does not get DTMFs 



Maybe helping you we'll help him too







- James

>>> Giovanni Maruzzelli <gmaruzz at gmail.com> 6/14/2017 03:53 AM >>>

Never heard such problems 


Please pastebin your dialplan, your SIP profile, and the complete,
since beginning to end, unedited, debug output of console when receiving
a call which does not get DTMFs 







sent from mobile
cell: +39 347 266 56 18
Giovanni Maruzzelli
OpenTelecom.IT 



On Jun 13, 2017 23:07, "admin" <admin at tedssupply.com> wrote:



I am encountering the same issue. I am using an ESL call to api
originate, with little else except getting in and out of the port, and
the caller DTMF is failing to reach the called number. The called number
auto answer attendant does not respond to DTMF, and a call to a test
phone confirms no DTMF. This is a new upgrade from 1.2 to 1.6 and I
don't know if this was an issue in 1.2, but my users never complained
before 1.6 upgrade. Ideas?... 



>>> chenyzhi <eastour at 163.com> 06/13/17 3:32 PM >>>

It's a one leg call .there is no b-leg. 



please make a test on your freeswitch box . 



just type the command "originate user/1001 5000" on the freeswitch
console to see if your freeswitch instance can detect dtmf input. 











At 2017-06-12 22:57:17, "Luis Daniel Lucio Quiroz"
<luis.daniel.lucio at gmail.com> wrote:



check if you have transcoding, and if you do, check that dftm
type-codec on leg b are compatible. 



-- 

Luis Daniel Lucio Quiroz
CISSP, CISM, CISA
Linux, VoIP and much more fun
www.okay.com.mx

Need LCR? Check out LCR for FusionPBX with FreeSWITCH
Need Billing? Check out Billing for FusionPBX with FreeSWITCH 



On Mon, Jun 12, 2017 at 9:16 AM, chenyzhi <eastour at 163.com> wrote:



I don't think It's the DTMF type , because when I dial 5000 from x-lite
(which has registered to freeswitch as 1001) ,I can hear the voice and
if I press any dtmf on x-lite, freeswitch can recieve the dtmfs. This
means that the DTMF type is correct ,otherwise freeswitch coudn't have
received the dtmfs; 



when I enter the command "originate user/1001 5000" at the freeswitch
console , my xlite will ring ,and I answered ,I can hear the voice ,I
press some dtmf,but freeswitch can NOT receive any dtmf. really weird. 











At 2017-06-12 17:37:49, "Giovanni Maruzzelli" <gmaruzz at gmail.com>
wrote:



you need to check the DTMF type, you probably are using the wrong one
(info-inband-rfc2833), and for some reason they are not negotiated



On 12 June 2017 at 02:32, chenyzhi <eastour at 163.com> wrote:



I have read the logs ,but I didn't find any difference. 



Please make a test to see if this happens in your box. 











At 2017-06-10 06:49:34, "David Villasmil"
<david.villasmil.work at gmail.com> wrote:



Have you looked at the log? Bump the logging up and see what shows
up... what you're seeing is very weird

David

On Fri, Jun 9, 2017 at 11:18 AM chenyzhi <eastour at 163.com> wrote:





Hi 



call 5000 from x-lite , you can hear the IVR voice and if you press
dtmf keys , freeswitch can receive the dtmf keys . 



but , if you enter the command "originate user/1001 5000" , x-lite will
ring ,answer it ,you can hear the IVR voice , press some keys ,the
freeswitch can NOT receive any dtmf , 



why? 
 






_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.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
http://www.freeswitch.org  
 







_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.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
http://www.freeswitch.org
 




--


Sincerely,

Giovanni Maruzzelli
OpenTelecom.IT
cell: +39 347 266 56 18
 
 







_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.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
http://www.freeswitch.org
 


 







_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.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
http://www.freeswitch.org
 


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.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
http://www.freeswitch.org
 




--


Sincerely,

Giovanni Maruzzelli
OpenTelecom.IT
cell: +39 347 266 56 18
 
 






_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.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
http://www.freeswitch.org
 




--


Sincerely,

Giovanni Maruzzelli
OpenTelecom.IT
cell: +39 347 266 56 18
 
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170616/1c73eceb/attachment-0001.html>

From tculjaga at gmail.com  Fri Jun 16 16:20:40 2017
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Fri, 16 Jun 2017 18:20:40 +0200
Subject: [Freeswitch-users] Call recording
In-Reply-To: <e9ad96d9cb344b128e04bb768be6fc83@w12mad02.comunycarse.com>
References: <7c47071f86ce436186b4ab8bf58de46c@w12mad02.comunycarse.com>
 <CALXCt0ocv6Hhfq7vfHhQf15047E9d+bWfuSE+wkLbwgLLh6Ptw@mail.gmail.com>
 <5323e6cfb2844e59a19957fad4e8044c@w12mad02.comunycarse.com>
 <CALXCt0ohebExQsNw+VCB0sPT-7hkRQHAw_pS9dMsOD0cFtxriw@mail.gmail.com>
 <e9ad96d9cb344b128e04bb768be6fc83@w12mad02.comunycarse.com>
Message-ID: <6B44BE94-4FF7-4D38-98A5-F54BAB4A510A@gmail.com>

I would not use lua, dialplan works great ;)

Sent from my iPhone

> On 16 Jun 2017, at 09:53, Sven Uszpelkat <USZPELSV at comunycarse.com> wrote:
> 
> Hi Giovanni,
>  
> Thank you for your help. We changed our script to something like this:
>  
> session:answer()
> while(session:ready() == true) do
>    test = session:recordFile("/usr/local/freeswitch/recordings/test.wav", 18000, 0, 300)
>    os.execute("sync");
>    session:setAutoHangup(false)
>    session:hangup()
> end
>  
> As result we note an improvement primarily in short recordings (<1:30 min). These are now practically all complete. In longer recordings there are still losses and it seems that they are increasing with the duration of the recording. I’m not sure how to interpret this but to me it looks like this: with the sync call we achieved to write the buffer content to the file, however in longer calls there is remaining audio which hasn’t even been read to the buffer. Is that correct? If so then the recording function doesn’t ensure to read the remaining audio after hangup. (With tcdump we checked that all audio packets arrived before the BYE message) Does this mean that this behavior is by design?
>  
> Best regards,
> Sven
>  
> De: Giovanni Maruzzelli [mailto:gmaruzz at gmail.com] 
> Enviado el: jueves, 15 de junio de 2017 13:17
> Para: Sven Uszpelkat
> CC: FreeSWITCH Users Help
> Asunto: Re: [Freeswitch-users] Call recording
>  
>  
>  
> On 15 June 2017 at 12:50, Sven Uszpelkat <USZPELSV at comunycarse.com> wrote:
>  
> 3.)    We set the silence threshold to 0 because the documentation is not very clear how to disable the silence detection. We don’t want the recording to stop in response to a period of silence. The point is to record everything.
> 
>  
> The silence_threshold determines what is considered silence, eg below what level of acoustic energy we state the stream is containing silence. Then, we wait for "how_many_silence_seconds" or until hangup before stopping recording.
> 
> So, maybe you are right, and setting it to 0 will consider silence only when there is absolute silence in the stream, so for all practical purposes, until hangup. I have no mean to check source code now.
> 
> On another hand, I can think at other possible causes for the premature end of the recorded file: maybe you move the file before it has been flushed by FreeSWITCH or by operating system? Maybe the hangup in the script interrupts the recording in the script and close the file descriptor before is flushed? (I am shooting in the dark)
> 
> You can try to leave out those two lines, and test again.
> 
> Also, you can insert a line that sync (flush) the filesystem before exiting, just to be sure.
> 
> I would insert it after the while(session:ready())
> 
> A system(sync), or something similar will probably do.
>  
> Hope this helps,
> -giovanni
> 
> 
> 
> 
>  
>  
> Best regards,
> Sven
>  
> De: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] En nombre de Giovanni Maruzzelli
> Enviado el: jueves, 15 de junio de 2017 11:19
> Para: FreeSWITCH Users Help
> Asunto: Re: [Freeswitch-users] Call recording
>  
>  
>  
> On 15 June 2017 at 09:49, Sven Uszpelkat <USZPELSV at comunycarse.com> wrote:
> Hello,
>  
> We are using FreeSWITCH as a third-party recording application, i.e. we are receiving SIP calls with the complete audio of conversations taking place on another switch and we are saving this audio to a file. To achieve this we are using a simple script similar to this:
>  
> session:answer()
> while(session:ready() == true) do
>    test = session:recordFile("/usr/local/freeswitch/recordings/test.wav", 18000, 0, 300)
>    session:setAutoHangup(false)
>    session:hangup()
> end
>  
> This script will be invoked by the following dialplan:
>  
> <extension name="Recording">
>       <condition field="destination_number" expression=".*">
>             <action application="lua" data="record.lua"/>
>       </condition>
> </extension>
>  
> Basically it seems to work quite well, but sometimes there are missing audio at the end of the recorded file.  Usually it’s only a few seconds, but sometimes it seems to be more. (It’s like the recording sometimes goes behind the real call and when the hangup event is received the remaining audio is discarded.)
>  
> What could be the reason for this behavior? Is there something wrong with the script or is there a better way to achieve our goal?
>  
> One first question come to my mind: why do you use a script here? A simple extension can do exactly the same, if you just want to record the session...
> 
> https://freeswitch.org/confluence/display/FREESWITCH/mod_dptools:+record_session
> 
> Anyway, if you want to use the script, why you first 
> 
> session:setAutoHangup(false)
> 
> and after that you
>  
> session:hangup()
>  
> ?
>  
> Also, you made the silence_threshold equal 0 (zero). 
> Have you has the same problems using a silence_threshold of, let's say, 30 (thirty), like in documentation? ( https://freeswitch.org/confluence/display/FREESWITCH/Lua+API+Reference#LuaAPIReference-session:recordFile )
>  
> Hope this helps,
> 
> -giovanni
> 
>  
>  
> Many thanks in advance.
>  
> Best regards,
>  
> Sven Uszpelkat
> Departamento I+D
> Comunycarse Network Consultants, S.L.
>  
>  
> <image001.jpg> <image002.jpg> <image003.jpg> <image004.jpg>
> 
> Joaquín Turina, 2
> 28224 Pozuelo de Alarcón MADRID
> Tlf. +34 917 498 700
> Fax +34 917 498 720 
> 
> Sabino Arana, 18
> 08028 BARCELONA
> Tlf. +34 934 098 480
> Fax +34 934 098 490
> 
> http://www.comunycarse.com
> 
> AVISO LEGAL
> La presente comunicación y sus anexos tiene como destinatario la persona a la que va dirigida, por lo que si usted lo recibe por error debe notificarlo al remitente y eliminarlo de su sistema, no pudiendo utilizarlo, total o parcialmente, para ningún fin. Su contenido puede tener información confidencial o protegida legalmente y únicamente expresa la opinión del remitente. El uso del correo electrónico vía internet no permite asegurar ni la confidencialidad de los mensajes ni su correcta recepción. En el caso de que el destinatario no consintiera la utilización del correo electrónico deberá ponerlo en nuestro conocimiento inmediatamente.
> 
> DISCLAIMER
> This message and its attachments are intended exclusively for the named addressee. If you receive this message by mistake, please delete it immediately from your system and notify the sender. You may not use this message or any part of it for any purpose. The message may contain information that is confidential or protected by law, and any opinions expressed are those of the individual sender. Internet email guarantees neither the confidentiality nor the proper receipt of the message sent. If the addressee of this message does not consent to the use of internet e-mail, please inform us immediately.
>  
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org
> 
> 
> 
> --
> 
> Sincerely,
> 
> Giovanni Maruzzelli
> OpenTelecom.IT
> cell: +39 347 266 56 18
> 
> 
> 
> --
> 
> Sincerely,
> 
> Giovanni Maruzzelli
> OpenTelecom.IT
> cell: +39 347 266 56 18
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170616/d3b11616/attachment-0001.html>

From soapee01.fs at stubbornroses.com  Fri Jun 16 17:06:06 2017
From: soapee01.fs at stubbornroses.com (soapee01.fs at stubbornroses.com)
Date: Fri, 16 Jun 2017 12:06:06 -0500
Subject: [Freeswitch-users] Missed Calls with Hunt Group
In-Reply-To: <594304D6.7070207@stubbornroses.com>
References: <594304D6.7070207@stubbornroses.com>
Message-ID: <59440FFE.1040905@stubbornroses.com>

All:

Gill tested this for me on 1.6.17, and the hunt group does not show 
missed calls on phones that did not answer the call on 1.6.17.

Regards,

James

On 6/15/2017 5:06 PM, soapee01.fs at stubbornroses.com wrote:
> Hi,
>
> Here's the command that I'm running.
>
> originate user/102 at domain &bridge(user/100 at domain,user/101 at domain)
>
>
> On FS version 1.2.22, if the user 100 answers the call, user 101 will 
> not see a missed call notification.
>
> On FS version FS 1.6.5 if the user 100 answers the call, user 101  
> will show a missed call.
>
> Is there something in the docs I've missed? I'd really like to set it 
> back to the old way, but it would be really cool if there's a variable 
> somewhere I've missed that lets you choose the behavior.
>
> Thanks!
>
> James
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org



From adrian.worutowicz at esifrance.net  Fri Jun 16 13:17:05 2017
From: adrian.worutowicz at esifrance.net (Adrian Worutowicz)
Date: Fri, 16 Jun 2017 15:17:05 +0200
Subject: [Freeswitch-users] Build Problem in VS2015
In-Reply-To: <80c901d2e5f4$7e26f930$7a74eb90$@tollfreegateway.com>
References: <001001d2e5bb$f4519cc0$dcf4d640$@worutowicz@esifrance.net>
 <80c901d2e5f4$7e26f930$7a74eb90$@tollfreegateway.com>
Message-ID: <005101d2e6a2$d9b0d1a0$8d1274e0$@worutowicz@esifrance.net>

I followed your steps, but unfortunately I got the same result.

Probably I’m missing something in my VS install.

 

Thanks a lot anyway


 

 

 

De : FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org]
De la part de krice at tollfreegateway.com
Envoyé : jeudi 15 juin 2017 18:29
À : 'FreeSWITCH Users Help'
Objet : Re: [Freeswitch-users] Build Problem in VS2015

 

Not sure whats you are doing incorrect here, but I have just built master, I
use the built in git bits with VS2015, and then drop to a command prompt
(via the team explorer tab, select branches, right click the repo and select
open command prompt)

 

Then git pull, git clean -fdx, git reset –hard origin/master , git pull

 

>From here back to the solution explorer open the FreeSWITCH.2015 solution
file and build as normal


 

I think you have something skewed there old ssl vs new ssl bits 

 

From: FreeSWITCH-users
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Adrian
Worutowicz
Sent: Thursday, June 15, 2017 4:44 AM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Build Problem in VS2015

 

Hello,

 

I try to recompile FS in VS2015 without success.

I took FS sources from git master.

 

git config --global core.autocrlf false

git clone https://stash.freeswitch.org/scm/fs/freeswitch.git
/c/ESI/Components/FreeSwitch/
<https://stash.freeswitch.org/scm/fs/freeswitch.git%20/c/ESI/Components/Free
Switch/> 

 

I have wix311 for VS2015 installed.

 

For example it searches in folder 'openssl-1.0.2k' while only a folder
'openssl' exists.

 

I tried to recompile mod_PortAudio, and I got
c:\ESI\components\freeswitch\src\mod\endpoints\mod_portaudio\pablio.h(55):
fatal error C1083: Impossible d'ouvrir le fichier include : 'portaudio.h' :
No such file or directory.

 

Indeed 'portaudio.h' does not exist.

Plenty of other errors in the attached file.

 

What do I miss?

 

Thanks in advance,

Adrian.

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170616/ecaa4797/attachment-0001.html>

From rmundkowsky at ets.org  Fri Jun 16 14:21:24 2017
From: rmundkowsky at ets.org (Mundkowsky, Robert)
Date: Fri, 16 Jun 2017 14:21:24 +0000
Subject: [Freeswitch-users] Verto failover
In-Reply-To: <CAGqqW98aEn_LGs=hvPmPqXkPTYpLyPoS1tEtz-+af714yJHnjw@mail.gmail.com>
References: <CAGqqW9-YHn5nW5f4OFz0=EedosPyWzcM4Y0Cd3Cn31TX6j4=aQ@mail.gmail.com>
 <CAHtxdDd885V0+LjRB24THaka1o40dWy1x-kO2Ko_jZZ5aXiyBQ@mail.gmail.com>
 <BN6PR07MB3556AA8421B7AA8E8439856FBEC00@BN6PR07MB3556.namprd07.prod.outlook.com>
 <CAGqqW98aEn_LGs=hvPmPqXkPTYpLyPoS1tEtz-+af714yJHnjw@mail.gmail.com>
Message-ID: <BN6PR07MB35565A904489A5DC396E1BBBBEC10@BN6PR07MB3556.namprd07.prod.outlook.com>

I think Sergey’s answer is good for you.

I was just curious if Sergey knew if his solution supports active call failover as well, which I doubt, but I am not sure.

Robert Mundkowsky

From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ketan Kothari
Sent: Friday, June 16, 2017 3:03 AM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Verto failover

Hello Robert,
I'm looking for fail-over for next call and web-interface as well as if one FreeSWITCH failed then verto continue working from other FreeSWITCH server.
If you have any suggestion or reference link then please pass it will helpful for me.


On Thu, Jun 15, 2017 at 7:21 PM, Mundkowsky, Robert <rmundkowsky at ets.org<mailto:rmundkowsky at ets.org>> wrote:
Verto is just frontend client side code (javascript), you can put it anywhere and point to a FreeSWITCH server anywhere else.

I am curious what type of failover is handled by Sergey’s suggestion.

Would this support an active call to continue? Or are you talking about failover for next call?


Robert Mundkowsky

From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Sergey Safarov
Sent: Thursday, June 15, 2017 7:17 AM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
Subject: Re: [Freeswitch-users] Verto failover

Yes.
Balancing can be done using:
1) at dns level;
2) using haproxy daemon;
3) using nginx as proxy.

чт, 15 июн. 2017 г. в 11:41, Ketan Kothari <kkothari157 at gmail.com<mailto:kkothari157 at gmail.com>>:
Can we setup Verto communicator User-Interface and FreeSWITCH on separate server?
If yes

---> We have 2 FreeSWITCH servers and 1 User-Interface server. So how to load balance or fail-over of Verto?
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com<https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.freeswitchsolutions.com&data=02%7C01%7Crmundkowsky%40ets.org%7Cad70a93ec7c5477c6e5108d4b3e0af98%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636331224986585344&sdata=rabcpCufsZMmPOC35CVZIeWm4YeP2t%2B67cYi0ND5V3s%3D&reserved=0>

Official FreeSWITCH Sites
http://www.freeswitch.org<https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.freeswitch.org&data=02%7C01%7Crmundkowsky%40ets.org%7Cad70a93ec7c5477c6e5108d4b3e0af98%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636331224986585344&sdata=rmrLXKiFmbYBsk38ZuQFYxyyYWm4mT9%2BKS2aZshgMEQ%3D&reserved=0>
http://confluence.freeswitch.org<https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fconfluence.freeswitch.org&data=02%7C01%7Crmundkowsky%40ets.org%7Cad70a93ec7c5477c6e5108d4b3e0af98%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636331224986585344&sdata=P5vBRCKyjO9am3mHuUGyAbXKopi64iTjVaoL64oGTE4%3D&reserved=0>
http://www.cluecon.com<https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.cluecon.com&data=02%7C01%7Crmundkowsky%40ets.org%7Cad70a93ec7c5477c6e5108d4b3e0af98%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636331224986585344&sdata=5wem7ZBJIWbgHc5BVkS08hI4li6dAlitksUO06zBkMU%3D&reserved=0>

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<https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Flists.freeswitch.org%2Fmailman%2Flistinfo%2Ffreeswitch-users&data=02%7C01%7Crmundkowsky%40ets.org%7Cad70a93ec7c5477c6e5108d4b3e0af98%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636331224986585344&sdata=0b9wmreHehfx%2Ft%2FxRZOxPHVqk70qdxIMkG0GPaCCDr4%3D&reserved=0>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users<https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Flists.freeswitch.org%2Fmailman%2Foptions%2Ffreeswitch-users&data=02%7C01%7Crmundkowsky%40ets.org%7Cad70a93ec7c5477c6e5108d4b3e0af98%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636331224986585344&sdata=LBMZOQVvh23pws%2FN8rGsIIdg6tVxagZIRFY1%2B8N7Uzk%3D&reserved=0>
http://www.freeswitch.org<https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.freeswitch.org&data=02%7C01%7Crmundkowsky%40ets.org%7Cad70a93ec7c5477c6e5108d4b3e0af98%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636331224986585344&sdata=rmrLXKiFmbYBsk38ZuQFYxyyYWm4mT9%2BKS2aZshgMEQ%3D&reserved=0>

________________________________

This e-mail and any files transmitted with it may contain privileged or confidential information. It is solely for use by the individual for whom it is intended, even if addressed incorrectly. If you received this e-mail in error, please notify the sender; do not disclose, copy, distribute, or take any action in reliance on the contents of this information; and delete it from your system. Any other use of this e-mail is prohibited.


Thank you for your compliance.

________________________________

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com<https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.freeswitchsolutions.com&data=02%7C01%7Crmundkowsky%40ets.org%7C67ee24ca36584a3aa50e08d4b4a6c013%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636332075656856073&sdata=TAD4IIf7QhMlxB9CBA%2B%2FkVQleeLeqzdXxzo03t%2B1iwE%3D&reserved=0>

Official FreeSWITCH Sites
http://www.freeswitch.org<https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.freeswitch.org&data=02%7C01%7Crmundkowsky%40ets.org%7C67ee24ca36584a3aa50e08d4b4a6c013%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636332075656856073&sdata=CwaxkYB60mxd5DyuGzQgP%2FYJxr5S%2FY1hh%2FgnAC7%2Boio%3D&reserved=0>
http://confluence.freeswitch.org<https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fconfluence.freeswitch.org&data=02%7C01%7Crmundkowsky%40ets.org%7C67ee24ca36584a3aa50e08d4b4a6c013%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636332075656856073&sdata=fPF%2FxnAUlk8UdTYJKaema7gGz9I%2FC9mXdJalJ%2FaaxFk%3D&reserved=0>
http://www.cluecon.com<https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.cluecon.com&data=02%7C01%7Crmundkowsky%40ets.org%7C67ee24ca36584a3aa50e08d4b4a6c013%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636332075656856073&sdata=sE6J2uV06bEIgUuQn4GedFhzuH%2F%2BMi%2B7u70N7pOSnRQ%3D&reserved=0>

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<https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Flists.freeswitch.org%2Fmailman%2Flistinfo%2Ffreeswitch-users&data=02%7C01%7Crmundkowsky%40ets.org%7C67ee24ca36584a3aa50e08d4b4a6c013%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636332075656856073&sdata=9uO8IwFPpyirgVRkma%2BnNZt%2BkE50ZI1H19abeuV%2Fv%2FY%3D&reserved=0>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users<https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Flists.freeswitch.org%2Fmailman%2Foptions%2Ffreeswitch-users&data=02%7C01%7Crmundkowsky%40ets.org%7C67ee24ca36584a3aa50e08d4b4a6c013%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636332075656856073&sdata=Bbbe9WDpl%2BJ1jP8ndc9y%2BZ0v4OPecNhPE8sv%2BtNMtBg%3D&reserved=0>
http://www.freeswitch.org<https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.freeswitch.org&data=02%7C01%7Crmundkowsky%40ets.org%7C67ee24ca36584a3aa50e08d4b4a6c013%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636332075656866082&sdata=Pgpmdw02f%2FkQ%2BlEKao88ty4hjIzu4UDpT3fnoclNaHo%3D&reserved=0>


________________________________

This e-mail and any files transmitted with it may contain privileged or confidential information. It is solely for use by the individual for whom it is intended, even if addressed incorrectly. If you received this e-mail in error, please notify the sender; do not disclose, copy, distribute, or take any action in reliance on the contents of this information; and delete it from your system. Any other use of this e-mail is prohibited.


Thank you for your compliance.

________________________________
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170616/86057eb5/attachment-0001.html>

From s.safarov at gmail.com  Fri Jun 16 17:38:08 2017
From: s.safarov at gmail.com (Sergey Safarov)
Date: Fri, 16 Jun 2017 17:38:08 +0000
Subject: [Freeswitch-users] Verto failover
In-Reply-To: <CAGqqW9_5e2Q5NvRQcPDnTmQaHLa8qpr5kbNfv1FR5tMOEnp3KQ@mail.gmail.com>
References: <CAGqqW9-YHn5nW5f4OFz0=EedosPyWzcM4Y0Cd3Cn31TX6j4=aQ@mail.gmail.com>
 <CAHtxdDd885V0+LjRB24THaka1o40dWy1x-kO2Ko_jZZ5aXiyBQ@mail.gmail.com>
 <CAGqqW9_5e2Q5NvRQcPDnTmQaHLa8qpr5kbNfv1FR5tMOEnp3KQ@mail.gmail.com>
Message-ID: <CAHtxdDdestZ4qyDAbpx41rxtgdcevSNHZW+o4AmDLivod0Z3Rg@mail.gmail.com>

NGINX
example https://www.nginx.com/blog/nginx-nodejs-websockets-socketio/

Also you may look iptables solution
https://www.webair.com/community/simple-stateful-load-balancer-with-iptables-and-nat/


чт, 15 июн. 2017 г. в 17:18, Ketan Kothari <kkothari157 at gmail.com>:

> Hello Sergey,
>
> Thanks for your response.
>
>
>
> *3) using nginx as proxy.*
> Could you please tell me some reference to do it.
>
>
> On Thu, Jun 15, 2017 at 4:46 PM, Sergey Safarov <s.safarov at gmail.com>
> wrote:
>
>> Yes.
>> Balancing can be done using:
>> 1) at dns level;
>> 2) using haproxy daemon;
>> 3) using nginx as proxy.
>>
>> чт, 15 июн. 2017 г. в 11:41, Ketan Kothari <kkothari157 at gmail.com>:
>>
>>> Can we setup Verto communicator *User-Interface* and *FreeSWITCH* on
>>> separate server?
>>> If yes
>>>
>>> *--->* We have 2 FreeSWITCH servers and 1 User-Interface server. So how
>>> to load balance or fail-over of Verto?
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.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
>>> http://www.freeswitch.org
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.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
>> http://www.freeswitch.org
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170616/be62fffc/attachment.html>

From s.safarov at gmail.com  Fri Jun 16 18:45:27 2017
From: s.safarov at gmail.com (Sergey Safarov)
Date: Fri, 16 Jun 2017 18:45:27 +0000
Subject: [Freeswitch-users] Verto failover
In-Reply-To: <BN6PR07MB35565A904489A5DC396E1BBBBEC10@BN6PR07MB3556.namprd07.prod.outlook.com>
References: <CAGqqW9-YHn5nW5f4OFz0=EedosPyWzcM4Y0Cd3Cn31TX6j4=aQ@mail.gmail.com>
 <CAHtxdDd885V0+LjRB24THaka1o40dWy1x-kO2Ko_jZZ5aXiyBQ@mail.gmail.com>
 <BN6PR07MB3556AA8421B7AA8E8439856FBEC00@BN6PR07MB3556.namprd07.prod.outlook.com>
 <CAGqqW98aEn_LGs=hvPmPqXkPTYpLyPoS1tEtz-+af714yJHnjw@mail.gmail.com>
 <BN6PR07MB35565A904489A5DC396E1BBBBEC10@BN6PR07MB3556.namprd07.prod.outlook.com>
Message-ID: <CAHtxdDfshFiujyQva5jM=S5OatOYTRB2XkQsxsbpZ8_QQ2WCbQ@mail.gmail.com>

пт, 16 июн. 2017 г. в 20:40, Mundkowsky, Robert <rmundkowsky at ets.org>:

> I think Sergey’s answer is good for you.
>
>
>
> I was just curious if Sergey knew if his solution supports active call
> failover as well, which I doubt, but I am not sure.
>
Failover  must be supported on on server side and on client side.

I tested Verto client and may say failover is works on client side.
Also i tested nginx in many cases, failover is works too.

I not tested Verto failover on FreeSwitch side.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170616/3664fb7a/attachment.html>

From s.safarov at gmail.com  Fri Jun 16 18:52:28 2017
From: s.safarov at gmail.com (Sergey Safarov)
Date: Fri, 16 Jun 2017 18:52:28 +0000
Subject: [Freeswitch-users] WebRTC and Carrier NAT ( CgNAT)
In-Reply-To: <CAJoKKdU_iQigcaoiWVaHtd8Si=gfvOyJ3MEGdUSG62i3UP32rA@mail.gmail.com>
References: <CAJoKKdU_iQigcaoiWVaHtd8Si=gfvOyJ3MEGdUSG62i3UP32rA@mail.gmail.com>
Message-ID: <CAHtxdDdur5sMFA_Xhu-FtboyoOm1R0xrLqpuv1CLZiGaYqu4cg@mail.gmail.com>

Many carrier use this NAT schema.
At home i have double NAT.
1) my home WiFi router
2) carrier cone NAT

All works as expected. Tested WebRTC (sipML5) and Verto.



пт, 16 июн. 2017 г. в 12:01, Tihomir Culjaga <tculjaga at gmail.com>:

> hello,
>
> does anyone have experience with WebRTC via carrier NAT ?
>
> client_LAN(192.168.x.y) <NAT> carrer_prvate_WAN(10.x.x.x) <NAT>
> carrier_public_WAN(public_ip) <> internet
>
>
> in this scenario, WebRTC calls using STUN only will not work. What about
> TURN ?
>
> did anyone try that ?
>
> Regards,
> Tihomir.
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170616/a0c0ff30/attachment.html>

From gregor at infomedia.si  Fri Jun 16 19:19:07 2017
From: gregor at infomedia.si (Gregor Nanger)
Date: Fri, 16 Jun 2017 19:19:07 +0000
Subject: [Freeswitch-users] Build Problem in VS2015
In-Reply-To: <594417cf.462ded0a.12ca4.f4aeSMTPIN_ADDED_BROKEN@mx.google.com>
References: <80c901d2e5f4$7e26f930$7a74eb90$@tollfreegateway.com>
 <594417cf.462ded0a.12ca4.f4aeSMTPIN_ADDED_BROKEN@mx.google.com>
Message-ID: <CAC7ttFE9Bq_UJx4J-iwf6gP7g1qrOmBxjmdJtEn_FKucs06njA@mail.gmail.com>

I do not have problems compiling with visual Studio. Except for cloning, I
use same command as stated in wiki. Then open in visual Studio and Build
solution.

On Fri, Jun 16, 2017, 19:39 Adrian Worutowicz <
adrian.worutowicz at esifrance.net> wrote:

> I followed your steps, but unfortunately I got the same result.
>
> Probably I’m missing something in my VS install.
>
>
>
> Thanks a lot anyway…
>
>
>
>
>
>
>
> *De :* FreeSWITCH-users [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *De la part de*
> krice at tollfreegateway.com
> *Envoyé :* jeudi 15 juin 2017 18:29
> *À :* 'FreeSWITCH Users Help'
> *Objet :* Re: [Freeswitch-users] Build Problem in VS2015
>
>
>
> Not sure whats you are doing incorrect here, but I have just built master,
> I use the built in git bits with VS2015, and then drop to a command prompt
> (via the team explorer tab, select branches, right click the repo and
> select open command prompt)
>
>
>
> Then git pull, git clean -fdx, git reset –hard origin/master , git pull
>
>
>
> From here back to the solution explorer open the FreeSWITCH.2015 solution
> file and build as normal…
>
>
>
> I think you have something skewed there old ssl vs new ssl bits
>
>
>
> *From:* FreeSWITCH-users [
> mailto:freeswitch-users-bounces at lists.freeswitch.org
> <freeswitch-users-bounces at lists.freeswitch.org>] *On Behalf Of *Adrian
> Worutowicz
> *Sent:* Thursday, June 15, 2017 4:44 AM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* [Freeswitch-users] Build Problem in VS2015
>
>
>
> Hello,
>
>
>
> I try to recompile FS in VS2015 without success.
>
> I took FS sources from git master.
>
>
>
> git config --global core.autocrlf false
>
> git clone https://stash.freeswitch.org/scm/fs/freeswitch.git
> /c/ESI/Components/FreeSwitch/
>
>
>
> I have wix311 for VS2015 installed.
>
>
>
> For example it searches in folder 'openssl-1.0.2k' while only a folder
> 'openssl' exists.
>
>
>
> I tried to recompile mod_PortAudio, and I got
> c:\ESI\components\freeswitch\src\mod\endpoints\mod_portaudio\pablio.h(55):
> fatal error C1083: Impossible d'ouvrir le fichier include : 'portaudio.h' :
> No such file or directory.
>
>
>
> Indeed 'portaudio.h' does not exist.
>
> Plenty of other errors in the attached file.
>
>
>
> What do I miss?
>
>
>
> Thanks in advance,
>
> Adrian.
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org

-- 
Gregor Nanger

*CTO*
t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485
• Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia
• www.infomedia.si
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170616/f3064a7a/attachment-0001.html>

From tculjaga at gmail.com  Fri Jun 16 19:53:53 2017
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Fri, 16 Jun 2017 21:53:53 +0200
Subject: [Freeswitch-users] WebRTC and Carrier NAT ( CgNAT)
In-Reply-To: <CAHtxdDdur5sMFA_Xhu-FtboyoOm1R0xrLqpuv1CLZiGaYqu4cg@mail.gmail.com>
References: <CAJoKKdU_iQigcaoiWVaHtd8Si=gfvOyJ3MEGdUSG62i3UP32rA@mail.gmail.com>
 <CAHtxdDdur5sMFA_Xhu-FtboyoOm1R0xrLqpuv1CLZiGaYqu4cg@mail.gmail.com>
Message-ID: <CAJoKKdUZ_VcRHxray+NRhGOps=T+ua4mKS0mh2m0brXo6VJkzA@mail.gmail.com>

did you try restricted ( address, port ) cone NAT and Symmetric NAT as well
?

i guess for Symmetric we do need a TURN server... not sure about other NAT
types/Methods of translation

On 16 June 2017 at 20:52, Sergey Safarov <s.safarov at gmail.com> wrote:

> Many carrier use this NAT schema.
> At home i have double NAT.
> 1) my home WiFi router
> 2) carrier cone NAT
>
> All works as expected. Tested WebRTC (sipML5) and Verto.
>
>
>
> пт, 16 июн. 2017 г. в 12:01, Tihomir Culjaga <tculjaga at gmail.com>:
>
>> hello,
>>
>> does anyone have experience with WebRTC via carrier NAT ?
>>
>> client_LAN(192.168.x.y) <NAT> carrer_prvate_WAN(10.x.x.x) <NAT>
>> carrier_public_WAN(public_ip) <> internet
>>
>>
>> in this scenario, WebRTC calls using STUN only will not work. What about
>> TURN ?
>>
>> did anyone try that ?
>>
>> Regards,
>> Tihomir.
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.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
>> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170616/4eaa3483/attachment.html>

From brian at freeswitch.org  Fri Jun 16 21:22:26 2017
From: brian at freeswitch.org (Brian West)
Date: Fri, 16 Jun 2017 16:22:26 -0500
Subject: [Freeswitch-users] WebRTC and Carrier NAT ( CgNAT)
In-Reply-To: <CAJoKKdUZ_VcRHxray+NRhGOps=T+ua4mKS0mh2m0brXo6VJkzA@mail.gmail.com>
References: <CAJoKKdU_iQigcaoiWVaHtd8Si=gfvOyJ3MEGdUSG62i3UP32rA@mail.gmail.com>
 <CAHtxdDdur5sMFA_Xhu-FtboyoOm1R0xrLqpuv1CLZiGaYqu4cg@mail.gmail.com>
 <CAJoKKdUZ_VcRHxray+NRhGOps=T+ua4mKS0mh2m0brXo6VJkzA@mail.gmail.com>
Message-ID: <CAEJMVkBicARmNbRKk++f-mSre5_LfLxfu7V8882sWgYD+tmuSw@mail.gmail.com>

If you're going to be doing Carrier NAT, You should use the Carrier NAT
Network Range.  That would be 100.64.0.0/10 :)

On Fri, Jun 16, 2017 at 2:53 PM, Tihomir Culjaga <tculjaga at gmail.com> wrote:

> did you try restricted ( address, port ) cone NAT and Symmetric NAT as
> well ?
>
> i guess for Symmetric we do need a TURN server... not sure about other NAT
> types/Methods of translation
>
> On 16 June 2017 at 20:52, Sergey Safarov <s.safarov at gmail.com> wrote:
>
>> Many carrier use this NAT schema.
>> At home i have double NAT.
>> 1) my home WiFi router
>> 2) carrier cone NAT
>>
>> All works as expected. Tested WebRTC (sipML5) and Verto.
>>
>>
>>
>> пт, 16 июн. 2017 г. в 12:01, Tihomir Culjaga <tculjaga at gmail.com>:
>>
>>> hello,
>>>
>>> does anyone have experience with WebRTC via carrier NAT ?
>>>
>>> client_LAN(192.168.x.y) <NAT> carrer_prvate_WAN(10.x.x.x) <NAT>
>>> carrier_public_WAN(public_ip) <> internet
>>>
>>>
>>> in this scenario, WebRTC calls using STUN only will not work. What about
>>> TURN ?
>>>
>>> did anyone try that ?
>>>
>>> Regards,
>>> Tihomir.
>>> ____________________________________________________________
>>> _____________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.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
>>> http://www.freeswitch.org
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.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
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org
>



-- 

*Brian West*
brian at freeswitch.org

*Twitter: @FreeSWITCH , @briankwest*

http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

Book a phone call (CST) <https://freeswitch.com/appointment>

Allison prompts for FreeSWITCH:

*https://www.gofundme.com/allison-prompts-for-freeswitch*
<https://www.gofundme.com/allison-prompts-for-freeswitch>

Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
/r/freeswitch <https://www.reddit.com/r/freeswitch>

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*Skype:*briankwest
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170616/a1365510/attachment.html>

From gmaruzz at gmail.com  Sat Jun 17 08:30:38 2017
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Sat, 17 Jun 2017 10:30:38 +0200
Subject: [Freeswitch-users] Call recording
In-Reply-To: <6B44BE94-4FF7-4D38-98A5-F54BAB4A510A@gmail.com>
References: <7c47071f86ce436186b4ab8bf58de46c@w12mad02.comunycarse.com>
 <CALXCt0ocv6Hhfq7vfHhQf15047E9d+bWfuSE+wkLbwgLLh6Ptw@mail.gmail.com>
 <5323e6cfb2844e59a19957fad4e8044c@w12mad02.comunycarse.com>
 <CALXCt0ohebExQsNw+VCB0sPT-7hkRQHAw_pS9dMsOD0cFtxriw@mail.gmail.com>
 <e9ad96d9cb344b128e04bb768be6fc83@w12mad02.comunycarse.com>
 <6B44BE94-4FF7-4D38-98A5-F54BAB4A510A@gmail.com>
Message-ID: <CALXCt0ojayVnDzHnqhrFrcpE7JKNoj+q=fxbu6=mpMV1RbXa3Q@mail.gmail.com>

On 16 June 2017 at 18:20, Tihomir Culjaga <tculjaga at gmail.com> wrote:

> I would not use lua, dialplan works great ;)
>
> Sent from my iPhone
>
> On 16 Jun 2017, at 09:53, Sven Uszpelkat <USZPELSV at comunycarse.com> wrote:
>
> Hi Giovanni,
>
>
>
> Thank you for your help. We changed our script to something like this:
>
>
>
> session:answer()
>
> while(session:ready() == true) do
>
>    test = session:recordFile("/usr/local/freeswitch/recordings/test.wav",
> 18000, 0, 300)
>
>    os.execute("sync");
>
>    session:setAutoHangup(false)
>
>    session:hangup()
>
> end
>
>
I would put an "if-end" instead of a "while-do", and put sync outside the
"ifend", setautohangup before the ifend, hangup after the ifend

you can also use sync-sleeponesecond-syncagain (after the ifend block),
just to be very proactive :)


-giovanni



>
>
> As result we note an improvement primarily in short recordings (<1:30
> min). These are now practically all complete. In longer recordings there
> are still losses and it seems that they are increasing with the duration of
> the recording. I’m not sure how to interpret this but to me it looks like
> this: with the sync call we achieved to write the buffer content to the
> file, however in longer calls there is remaining audio which hasn’t even
> been read to the buffer. Is that correct? If so then the recording function
> doesn’t ensure to read the remaining audio after hangup. (With tcdump we
> checked that all audio packets arrived before the BYE message) Does this
> mean that this behavior is by design?
>
>
>
> Best regards,
>
> Sven
>
>
>
> *De:* Giovanni Maruzzelli [mailto:gmaruzz at gmail.com <gmaruzz at gmail.com>]
> *Enviado el:* jueves, 15 de junio de 2017 13:17
> *Para:* Sven Uszpelkat
> *CC:* FreeSWITCH Users Help
> *Asunto:* Re: [Freeswitch-users] Call recording
>
>
>
>
>
>
>
> On 15 June 2017 at 12:50, Sven Uszpelkat <USZPELSV at comunycarse.com> wrote:
>
>
>
> 3.)    We set the silence threshold to 0 because the documentation is not
> very clear how to disable the silence detection. We don’t want the
> recording to stop in response to a period of silence. The point is to
> record everything.
>
>
>
> The silence_threshold determines what is considered silence, eg below what
> level of acoustic energy we state the stream is containing silence. Then,
> we wait for "how_many_silence_seconds" or until hangup before stopping
> recording.
>
> So, maybe you are right, and setting it to 0 will consider silence only
> when there is absolute silence in the stream, so for all practical
> purposes, until hangup. I have no mean to check source code now.
>
> On another hand, I can think at other possible causes for the premature
> end of the recorded file: maybe you move the file before it has been
> flushed by FreeSWITCH or by operating system? Maybe the hangup in the
> script interrupts the recording in the script and close the file descriptor
> before is flushed? (I am shooting in the dark)
>
> You can try to leave out those two lines, and test again.
>
> Also, you can insert a line that sync (flush) the filesystem before
> exiting, just to be sure.
>
> I would insert it after the while(session:ready())
>
> A system(sync), or something similar will probably do.
>
>
>
> Hope this helps,
>
> -giovanni
>
>
>
>
>
>
>
>
>
> Best regards,
>
> Sven
>
>
>
> *De:* FreeSWITCH-users [mailto:freeswitch-users-
> bounces at lists.freeswitch.org
> <freeswitch-users-bounces at lists.freeswitch.org>] *En nombre de *Giovanni
> Maruzzelli
> *Enviado el:* jueves, 15 de junio de 2017 11:19
> *Para:* FreeSWITCH Users Help
> *Asunto:* Re: [Freeswitch-users] Call recording
>
>
>
>
>
>
>
> On 15 June 2017 at 09:49, Sven Uszpelkat <USZPELSV at comunycarse.com> wrote:
>
> Hello,
>
>
>
> We are using FreeSWITCH as a third-party recording application, i.e. we
> are receiving SIP calls with the complete audio of conversations taking
> place on another switch and we are saving this audio to a file. To achieve
> this we are using a simple script similar to this:
>
>
>
> session:answer()
>
> while(session:ready() == true) do
>
>    test = session:recordFile("/usr/local/freeswitch/recordings/test.wav",
> 18000, 0, 300)
>
>    session:setAutoHangup(false)
>
>    session:hangup()
>
> end
>
>
>
> This script will be invoked by the following dialplan:
>
>
>
> <extension name="Recording">
>
>       <condition field="destination_number" expression=".*">
>
>             <action application="lua" data="record.lua"/>
>
>       </condition>
>
> </extension>
>
>
>
> Basically it seems to work quite well, but sometimes there are missing
> audio at the end of the recorded file.  Usually it’s only a few seconds,
> but sometimes it seems to be more. (It’s like the recording sometimes goes
> behind the real call and when the hangup event is received the remaining
> audio is discarded.)
>
>
>
> What could be the reason for this behavior? Is there something wrong with
> the script or is there a better way to achieve our goal?
>
>
>
> One first question come to my mind: why do you use a script here? A simple
> extension can do exactly the same, if you just want to record the session...
>
> https://freeswitch.org/confluence/display/FREESWITCH/
> mod_dptools:+record_session
>
> Anyway, if you want to use the script, why you first
>
> session:setAutoHangup(false)
>
> and after that you
>
>
>
> session:hangup()
>
>
>
> ?
>
>
>
> Also, you made the silence_threshold equal 0 (zero).
> Have you has the same problems using a silence_threshold of, let's say, 30
> (thirty), like in documentation? ( https://freeswitch.org/
> confluence/display/FREESWITCH/Lua+API+Reference#LuaAPIReference-session:
> recordFile )
>
>
>
> Hope this helps,
>
> -giovanni
>
>
>
>
>
>
> Many thanks in advance.
>
>
>
> Best regards,
>
>
>
> Sven Uszpelkat
> Departamento I+D
> Comunycarse Network Consultants, S.L.
>
>
>
>
>
> <image001.jpg> <http://www.facebook.com/pages/Comunycarse/227218309114>
> <image002.jpg> <http://www.linkedin.com/company/comunycarse-s.l.>
> <image003.jpg> <https://twitter.com/#%21/comunycarse> <image004.jpg>
> <http://www.blog.comunycarse.com/>
>
>
> Joaquín Turina, 2
> 28224 Pozuelo de Alarcón MADRID
> Tlf. +34 917 498 700 <+34%20917%2049%2087%2000>
> Fax +34 917 498 720 <+34%20917%2049%2087%2020>
>
> Sabino Arana, 18
> 08028 BARCELONA
> Tlf. +34 934 098 480 <+34%20934%2009%2084%2080>
> Fax +34 934 098 490 <+34%20934%2009%2084%2090>
>
> http://www.comunycarse.com
>
>
> AVISO LEGAL
> La presente comunicación y sus anexos tiene como destinatario la persona a
> la que va dirigida, por lo que si usted lo recibe por error debe
> notificarlo al remitente y eliminarlo de su sistema, no pudiendo
> utilizarlo, total o parcialmente, para ningún fin. Su contenido puede tener
> información confidencial o protegida legalmente y únicamente expresa la
> opinión del remitente. El uso del correo electrónico vía internet no
> permite asegurar ni la confidencialidad de los mensajes ni su correcta
> recepción. En el caso de que el destinatario no consintiera la utilización
> del correo electrónico deberá ponerlo en nuestro conocimiento
> inmediatamente.
>
> DISCLAIMER
> This message and its attachments are intended exclusively for the named
> addressee. If you receive this message by mistake, please delete it
> immediately from your system and notify the sender. You may not use this
> message or any part of it for any purpose. The message may contain
> information that is confidential or protected by law, and any opinions
> expressed are those of the individual sender. Internet email guarantees
> neither the confidentiality nor the proper receipt of the message sent. If
> the addressee of this message does not consent to the use of internet
> e-mail, please inform us immediately.
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org
>
>
>
>
> --
>
>
> Sincerely,
>
> Giovanni Maruzzelli
> OpenTelecom.IT
> cell: +39 347 266 56 18
>
>
>
>
> --
>
>
> Sincerely,
>
> Giovanni Maruzzelli
> OpenTelecom.IT
> cell: +39 347 266 56 18
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org
>
>


-- 

Sincerely,

Giovanni Maruzzelli
OpenTelecom.IT
cell: +39 347 266 56 18
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170617/34ae375a/attachment-0001.html>

From gmaruzz at gmail.com  Sat Jun 17 08:32:17 2017
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Sat, 17 Jun 2017 10:32:17 +0200
Subject: [Freeswitch-users] When calling out ,
	freeswitch doesn't get DTMF ?
In-Reply-To: <7611bb49.72be.15caf10d2a0.Coremail.eastour@163.com>
References: <f823cdd8-a354-3698-d8c5-484b1b4cc8b4@xbipin.com>
 <CAEJMVkDDQX=Ni3AOQxQOo=joGoAeYroxioVP=cu9w6Y2m7kLPA@mail.gmail.com>
 <15c898b5d58.2779.b07ebdf329620b8089087c7205b03f01@xbipin.com>
 <310cf71c.8fc3.15c8ba9c4f6.Coremail.eastour@163.com>
 <CAFGRPVqFA+tpaOQkAGGCR7s5gY0z0TbdArWFdkfnsaaG4Z9Dgw@mail.gmail.com>
 <2a18422f.ef8.15c99b7b9bb.Coremail.eastour@163.com>
 <CALXCt0pKKEJoSeXXZ=tn04nifEZvQCRSFCDKL8jSu+pfOgvRNQ@mail.gmail.com>
 <6c11dd0f.c2c1.15c9c727dc7.Coremail.eastour@163.com>
 <CAFLo2QwwwE6xeXBN9DkM1VEecEJpwnDMmQ7iT9EvSyx9RqE9mg@mail.gmail.com>
 <2f48505d.d9c5.15ca1d28424.Coremail.eastour@163.com>
 <59401B92020000310000A34D@mail.tedssupply.com>
 <CALXCt0qw4DCJ2Q0cF+kVEOpjCjp2T8eFgWbYx03sN=z5APBGGQ@mail.gmail.com>
 <59412DB4020000310000A381@mail.tedssupply.com>
 <CALXCt0rTBTBTfT+e1t0UCfpwmWyOhz1-yeQ2s6T8mSTYAnS-+w@mail.gmail.com>
 <7d1e3f12.4750.15ca994d89d.Coremail.eastour@163.com>
 <CALXCt0oRdP=8M_vbQpevzUs=NAO-rDVx0BqXJzv6mhy9Q5vZkg@mail.gmail.com>
 <7611bb49.72be.15caf10d2a0.Coremail.eastour@163.com>
Message-ID: <CALXCt0q_s1CByZPrQLN8fa=1-_Kat5QZB4FVohZp589Q9kf=0Q@mail.gmail.com>

On 16 June 2017 at 06:02, chenyzhi <eastour at 163.com> wrote:

> Yes ,I can hear all the IVR prompt voices correctly.
>
> I don't think it's  a NAT problem ,because both the x-lite and the
> freeswitch are in the same LAN.
>
> The sip trace log is in the attatchment. Thank you.
>
> PS  I tested this on another freeswitch box ,version:
> FreeSWITCH Version 1.6.16+git~20170403T142423Z~e6d643b29c~32bit (git
> e6d643b 2017-04-03 14:24:23Z 32bit)
> It can detect dtmf on outgoing calls.  Maybe this only happens on
> FreeSWITCH Version 1.9.0+git~20170518T231917Z~a1fc18aee5~32bit (git
> a1fc18a 2017-05-18 23:19:17Z 32bit)
>


then use the stable version, and open a jira for this issue citing the
master version you are using
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170617/fbda1caa/attachment.html>

From richard.mace at gmail.com  Sat Jun 17 09:47:18 2017
From: richard.mace at gmail.com (Richard Mace)
Date: Sat, 17 Jun 2017 10:47:18 +0100
Subject: [Freeswitch-users] Showing caller number
In-Reply-To: <CAK+KT39G8m1SruQP5AzP74gr=kUcDQRk0nRMyewefYCNKQG3Fw@mail.gmail.com>
References: <CAK+KT3-_O9GAY_mO2=k0QwqgOumc9rv4VgrTTRjNsf=PW9NDRw@mail.gmail.com>
 <CAK+KT3_YruiKUmbRdcXhENOPONhLtEMSum4sG0H3H1NV9T-v8g@mail.gmail.com>
 <CAK+KT39_DsgA0udXCF0WjEFMt_tpd2SHymy04-RMikrXgQpocw@mail.gmail.com>
 <CAK+KT3_FR0zNrsSmTaJ_9_ebVr-Hvn=PsjLOJm9yvgjExfH0sA@mail.gmail.com>
 <CAK+KT3_SnBjTW6hXYH2oi0H3mfc6T-OYtFQ6zuERBA8pXLLpMw@mail.gmail.com>
 <CAK+KT3_AFmjA8AupC+pBBN4uGfQhJK1dRsDsuZ4TN10vpSG7Dw@mail.gmail.com>
 <CAK+KT3_1BWUaH4Bwts9UPeugfaKrec8g3zOcKcLprR=UCFWdpQ@mail.gmail.com>
 <CAK+KT39jxJFrWFH16mXy9KrUuafST-yc_AOJy9X27cbPbgk29Q@mail.gmail.com>
 <CAK+KT3_e8ka1+LiBhh8O7Y3LnZCVBRZt3+mGXKv0oFwD8ZdAQw@mail.gmail.com>
 <CAK+KT3_-bE0_AEUDiDAGqR0nhMW3SiLFY3a6JBJ+HJhs-LQ2mg@mail.gmail.com>
 <CAK+KT39nPESM3OguwZ-qDK_2FnXrezr=_x1iMTXAXJmb8Q+tRA@mail.gmail.com>
 <CAK+KT39G8m1SruQP5AzP74gr=kUcDQRk0nRMyewefYCNKQG3Fw@mail.gmail.com>
Message-ID: <CAK+KT3_XqSkODhHkgULBxfPz5Yf_8ojvyXd-XBWQA2v_Q_qc9Q@mail.gmail.com>

Hi,

I have a situation where I have users dialling a number on my freeswitch
system starting with 01794 that then immediately calls a mobile number.
When the mobile rings, it looks like the 01794 number is calling the number.

How can I adjust the dial string so that when I call the mobile from
freeswitch, it looks like like original number calling the 01794 number is
ringing the mobile directly?

Thanks

Richard
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170617/66d22510/attachment.html>

From gmaruzz at gmail.com  Sat Jun 17 11:01:16 2017
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Sat, 17 Jun 2017 13:01:16 +0200
Subject: [Freeswitch-users] Showing caller number
In-Reply-To: <CAK+KT3_XqSkODhHkgULBxfPz5Yf_8ojvyXd-XBWQA2v_Q_qc9Q@mail.gmail.com>
References: <CAK+KT3-_O9GAY_mO2=k0QwqgOumc9rv4VgrTTRjNsf=PW9NDRw@mail.gmail.com>
 <CAK+KT3_YruiKUmbRdcXhENOPONhLtEMSum4sG0H3H1NV9T-v8g@mail.gmail.com>
 <CAK+KT39_DsgA0udXCF0WjEFMt_tpd2SHymy04-RMikrXgQpocw@mail.gmail.com>
 <CAK+KT3_FR0zNrsSmTaJ_9_ebVr-Hvn=PsjLOJm9yvgjExfH0sA@mail.gmail.com>
 <CAK+KT3_SnBjTW6hXYH2oi0H3mfc6T-OYtFQ6zuERBA8pXLLpMw@mail.gmail.com>
 <CAK+KT3_AFmjA8AupC+pBBN4uGfQhJK1dRsDsuZ4TN10vpSG7Dw@mail.gmail.com>
 <CAK+KT3_1BWUaH4Bwts9UPeugfaKrec8g3zOcKcLprR=UCFWdpQ@mail.gmail.com>
 <CAK+KT39jxJFrWFH16mXy9KrUuafST-yc_AOJy9X27cbPbgk29Q@mail.gmail.com>
 <CAK+KT3_e8ka1+LiBhh8O7Y3LnZCVBRZt3+mGXKv0oFwD8ZdAQw@mail.gmail.com>
 <CAK+KT3_-bE0_AEUDiDAGqR0nhMW3SiLFY3a6JBJ+HJhs-LQ2mg@mail.gmail.com>
 <CAK+KT39nPESM3OguwZ-qDK_2FnXrezr=_x1iMTXAXJmb8Q+tRA@mail.gmail.com>
 <CAK+KT39G8m1SruQP5AzP74gr=kUcDQRk0nRMyewefYCNKQG3Fw@mail.gmail.com>
 <CAK+KT3_XqSkODhHkgULBxfPz5Yf_8ojvyXd-XBWQA2v_Q_qc9Q@mail.gmail.com>
Message-ID: <CALXCt0paNn2gmzEzv_4s_qjYOUMtDBiGyBoh2ZK70GrMt0NbWw@mail.gmail.com>

https://freeswitch.org/confluence/display/FREESWITCH/Caller+ID+Privacy#CallerIDPrivacy-SettingCIDMethod



On 17 June 2017 at 11:47, Richard Mace <richard.mace at gmail.com> wrote:

> Hi,
>
> I have a situation where I have users dialling a number on my freeswitch
> system starting with 01794 that then immediately calls a mobile number.
> When the mobile rings, it looks like the 01794 number is calling the
> number.
>
> How can I adjust the dial string so that when I call the mobile from
> freeswitch, it looks like like original number calling the 01794 number is
> ringing the mobile directly?
>
> Thanks
>
> Richard
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org
>



-- 

Sincerely,

Giovanni Maruzzelli
OpenTelecom.IT
cell: +39 347 266 56 18
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170617/29053f41/attachment.html>

From lconroy at insensate.co.uk  Sat Jun 17 11:26:28 2017
From: lconroy at insensate.co.uk (Lawrence Conroy)
Date: Sat, 17 Jun 2017 12:26:28 +0100
Subject: [Freeswitch-users] Showing caller number
In-Reply-To: <CAK+KT3_XqSkODhHkgULBxfPz5Yf_8ojvyXd-XBWQA2v_Q_qc9Q@mail.gmail.com>
References: <CAK+KT3-_O9GAY_mO2=k0QwqgOumc9rv4VgrTTRjNsf=PW9NDRw@mail.gmail.com>
 <CAK+KT3_YruiKUmbRdcXhENOPONhLtEMSum4sG0H3H1NV9T-v8g@mail.gmail.com>
 <CAK+KT39_DsgA0udXCF0WjEFMt_tpd2SHymy04-RMikrXgQpocw@mail.gmail.com>
 <CAK+KT3_FR0zNrsSmTaJ_9_ebVr-Hvn=PsjLOJm9yvgjExfH0sA@mail.gmail.com>
 <CAK+KT3_SnBjTW6hXYH2oi0H3mfc6T-OYtFQ6zuERBA8pXLLpMw@mail.gmail.com>
 <CAK+KT3_AFmjA8AupC+pBBN4uGfQhJK1dRsDsuZ4TN10vpSG7Dw@mail.gmail.com>
 <CAK+KT3_1BWUaH4Bwts9UPeugfaKrec8g3zOcKcLprR=UCFWdpQ@mail.gmail.com>
 <CAK+KT39jxJFrWFH16mXy9KrUuafST-yc_AOJy9X27cbPbgk29Q@mail.gmail.com>
 <CAK+KT3_e8ka1+LiBhh8O7Y3LnZCVBRZt3+mGXKv0oFwD8ZdAQw@mail.gmail.com>
 <CAK+KT3_-bE0_AEUDiDAGqR0nhMW3SiLFY3a6JBJ+HJhs-LQ2mg@mail.gmail.com>
 <CAK+KT39nPESM3OguwZ-qDK_2FnXrezr=_x1iMTXAXJmb8Q+tRA@mail.gmail.com>
 <CAK+KT39G8m1SruQP5AzP74gr=kUcDQRk0nRMyewefYCNKQG3Fw@mail.gmail.com>
 <CAK+KT3_XqSkODhHkgULBxfPz5Yf_8ojvyXd-XBWQA2v_Q_qc9Q@mail.gmail.com>
Message-ID: <6061D25F-98A7-480E-8DCB-36B680213847@insensate.co.uk>

Hi there,
 so, basically, you want to fake the outgoing CLI. ISTM that this is NOT a freeSwitch problem.

freeswitch -could- send a different caller ID when outcalling, but you're using some supplier to call out to the mobile via the PSTN.
Typically, that supplier will either trust that you "own" the calling number you're presenting (customer provided network screened),
or will not want OfCom jumping up & down on their head and so put what they believe is your number in the SS#7 message they send
to the mobile operator in the outcall (network provided calling number).
Note that the supplier is supposed to at least screen whatever number you provide. It's their responsibility.

Given that OfCom tends to be unhappy with people faking calling numbers, this is a hard problem.

=> I suspect that it is not something that freeSwitch can fix with UK based outcalls.

best regards,
 Lawrence Conroy
(who used to have an 01794 833xxx number)

On 17 Jun 2017, at 10:47, Richard Mace <richard.mace at gmail.com> wrote:
> Hi,
> 
> I have a situation where I have users dialling a number on my freeswitch system starting with 01794 that then immediately calls a mobile number.
> When the mobile rings, it looks like the 01794 number is calling the number.
> 
> How can I adjust the dial string so that when I call the mobile from freeswitch, it looks like like original number calling the 01794 number is ringing the mobile directly?
> 
> Thanks
> 
> Richard



From infos at madovsky.org  Sat Jun 17 11:55:07 2017
From: infos at madovsky.org (Madovsky)
Date: Sat, 17 Jun 2017 04:55:07 -0700
Subject: [Freeswitch-users] timer not properly configured
Message-ID: <70e56fd8-4c40-53e1-2de2-bd8b879daad2@madovsky.org>

Hi all,

last today git gives

switch_core_timer.c:117 Timer is not properly configured

everytime a call is hangup.

show timer gives

type,name,ikey
timer,soft,CORE_SOFTTIMER_MODULE

1 total.


Thanks

F



From stefan at fuhrmann.homedns.org  Sat Jun 17 14:55:33 2017
From: stefan at fuhrmann.homedns.org (Stefan Fuhrmann)
Date: Sat, 17 Jun 2017 16:55:33 +0200
Subject: [Freeswitch-users] enable Portal, error 404
Message-ID: <8779966.9K6fzknjRZ@stefan-ubu>

Hello all,

Im new to freeswitch and have to ask, how can I enable the portal?
I installed the debian installation and followed the instruction from wiki to 
enable:
https://wiki.freeswitch.org/wiki/Freeswitch_Portal
It is based on mod_xml_rpc, the module is built by default but not loaded, so 
you just need to load it (un-comment it in conf/autoload_configs/
modules.conf.xml) 

load mod_xml_rpc

When I trying to access
ip:8080/portal/index.html
after login Im getting:
error 404

What Im missing?

Can somone help?

Tia
Stefan


From ksh.sip at gmail.com  Sat Jun 17 16:16:09 2017
From: ksh.sip at gmail.com (Gauri Kshirsagar)
Date: Sat, 17 Jun 2017 21:46:09 +0530
Subject: [Freeswitch-users] Adding video to audio call
Message-ID: <CACdNXfdsjhC5dbZGzNJxBsG4XnU1jtw6CpdTcqcwJmw2QGhnyw@mail.gmail.com>

Hi,

I am using Freeswitch version 1.9.0. I can make video calls. But when I try
to add video to audio call it does not work.

A makes audio call to B. Call is established. A adds video .

ReINVITE sent to freeswitch has  video added in SDP.Freeswitch sends 200 OK
response for this INVITE to A which has video in SDP. But there is no
ReINVITE being sent to B.

I tried enabling renegotiate-codec-on-reinvite in vars.xml and also
internal.xml

Is this supported? If so is some other configuration required.

Regards,
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170617/b1ec67eb/attachment.html>

From anthony.minessale at gmail.com  Sat Jun 17 17:29:31 2017
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 17 Jun 2017 17:29:31 +0000
Subject: [Freeswitch-users] timer not properly configured
In-Reply-To: <70e56fd8-4c40-53e1-2de2-bd8b879daad2@madovsky.org>
References: <70e56fd8-4c40-53e1-2de2-bd8b879daad2@madovsky.org>
Message-ID: <CAKbxfG-MJFhV+wkmd9suzMbvSnHLJ_V8YeX1cn7=KHQJvzx7EA@mail.gmail.com>

Jira jira jira
On Sat, Jun 17, 2017 at 7:55 AM Madovsky <infos at madovsky.org> wrote:

> Hi all,
>
> last today git gives
>
> switch_core_timer.c:117 Timer is not properly configured
>
> everytime a call is hangup.
>
> show timer gives
>
> type,name,ikey
> timer,soft,CORE_SOFTTIMER_MODULE
>
> 1 total.
>
>
> Thanks
>
> F
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org

-- 
Anthony Minessale II       ♬ @anthmfs  ♬ @FreeSWITCH  ♬

☞ http://freeswitch.org/  ☞ http://cluecon.com/  ☞
http://twitter.com/FreeSWITCH
☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+
<http://freeswitch.org/g+>*

ClueCon Weekly Development Call
☎ sip:888 at conference.freeswitch.org  ☎ +19193869900

https://www.youtube.com/watch?v=9XXgW34t40s
https://www.youtube.com/watch?v=NLaDpGQuZDA
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170617/7cc717d7/attachment.html>

From s.safarov at gmail.com  Sat Jun 17 18:39:25 2017
From: s.safarov at gmail.com (Sergey Safarov)
Date: Sat, 17 Jun 2017 18:39:25 +0000
Subject: [Freeswitch-users] Adding video to audio call
In-Reply-To: <CACdNXfdsjhC5dbZGzNJxBsG4XnU1jtw6CpdTcqcwJmw2QGhnyw@mail.gmail.com>
References: <CACdNXfdsjhC5dbZGzNJxBsG4XnU1jtw6CpdTcqcwJmw2QGhnyw@mail.gmail.com>
Message-ID: <CAHtxdDcns0e0oCaG8pHfKYYAeHhmbrihHg5EgbUWuatNpWnL5A@mail.gmail.com>

Requred to enable proxy media.

сб, 17 июня 2017 г., 19:20 Gauri Kshirsagar <ksh.sip at gmail.com>:

> Hi,
>
> I am using Freeswitch version 1.9.0. I can make video calls. But when I
> try to add video to audio call it does not work.
>
> A makes audio call to B. Call is established. A adds video .
>
> ReINVITE sent to freeswitch has  video added in SDP.Freeswitch sends 200
> OK response for this INVITE to A which has video in SDP. But there is no
> ReINVITE being sent to B.
>
> I tried enabling renegotiate-codec-on-reinvite in vars.xml and also
> internal.xml
>
> Is this supported? If so is some other configuration required.
>
> Regards,
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170617/a292be13/attachment.html>

From mateo.felipe05 at gmail.com  Sat Jun 17 15:34:39 2017
From: mateo.felipe05 at gmail.com (Felipe Mateo)
Date: Sat, 17 Jun 2017 11:34:39 -0400
Subject: [Freeswitch-users] Fwd: No video playback with mod_av
In-Reply-To: <CAAaM4gyJQfaG51MO4SYmfHhf5js8MAfo8Wv6vHy-fXJxPgh-Bg@mail.gmail.com>
References: <CAAaM4gyJQfaG51MO4SYmfHhf5js8MAfo8Wv6vHy-fXJxPgh-Bg@mail.gmail.com>
Message-ID: <CAAaM4gx07TWK5waDQo0y8p5ytyqxY8f5AvCFZqdjVu3ZOhzobQ@mail.gmail.com>

Hi all,

I am trying to use mod_av to playback application with video support; but
it does not show video output. It only shows video when dialing sample
conference (canvas and screen). I also searched confluence but there is no
documentation for mod_av


Thanks in advance
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170617/f879d7f0/attachment.html>

From bipin at xbipin.com  Sun Jun 18 06:02:10 2017
From: bipin at xbipin.com (Bipin Patel)
Date: Sun, 18 Jun 2017 10:02:10 +0400
Subject: [Freeswitch-users] Adding video to audio call
In-Reply-To: <CAHtxdDcns0e0oCaG8pHfKYYAeHhmbrihHg5EgbUWuatNpWnL5A@mail.gmail.com>
References: <CACdNXfdsjhC5dbZGzNJxBsG4XnU1jtw6CpdTcqcwJmw2QGhnyw@mail.gmail.com>
 <CAHtxdDcns0e0oCaG8pHfKYYAeHhmbrihHg5EgbUWuatNpWnL5A@mail.gmail.com>
Message-ID: <f01f26ec-d5da-b38c-9ad7-3174178c2685@xbipin.com>

hi,

also i believe there is a bug request open on jira relating to this


Regards,
Bipin


------------------------------------------------------------------------
-------- Original Message --------
Subject: Re: [Freeswitch-users] Adding video to audio call
From: Sergey Safarov <s.safarov at gmail.com>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Date: 6/17/2017, 10:39:25 PM
>
> Requred to enable proxy media.
>
>
> сб, 17 июня 2017 г., 19:20 Gauri Kshirsagar <ksh.sip at gmail.com 
> <mailto:ksh.sip at gmail.com>>:
>
>     Hi,
>
>     I am using Freeswitch version 1.9.0. I can make video calls. But
>     when I try to add video to audio call it does not work.
>
>     A makes audio call to B. Call is established. A adds video .
>
>     ReINVITE sent to freeswitch has  video added in SDP.Freeswitch
>     sends 200 OK response for this INVITE to A which has video in SDP.
>     But there is no ReINVITE being sent to B.
>
>     I tried enabling renegotiate-codec-on-reinvite in vars.xml and
>     also internal.xml
>
>     Is this supported? If so is some other configuration required.
>
>     Regards,
>
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://confluence.freeswitch.org
>     http://www.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
>     http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170618/6fe19ff1/attachment-0001.html>

From lexxua at gmail.com  Sun Jun 18 18:59:06 2017
From: lexxua at gmail.com (Volodymyr Fedorov)
Date: Sun, 18 Jun 2017 20:59:06 +0200
Subject: [Freeswitch-users] Debian Stretch
In-Reply-To: <733A7BE7-6E66-4BD8-8736-B6DDBFE695A4@jerris.com>
References: <CAKDN4FeqnOGn+45=5GXoNM8xqmks66ew8uM0GX4JiFEVrqBd+w@mail.gmail.com>
 <CF850B37-D178-46C8-8AE6-6BB4CF6C4DF6@jerris.com>
 <CAKDN4Ffs7g4cGAqr-YJrO9Cj=p9D4jV0fxrUnmgZ00hcV1bD0w@mail.gmail.com>
 <733A7BE7-6E66-4BD8-8736-B6DDBFE695A4@jerris.com>
Message-ID: <CAO_GGArQoE0gBVHzOFqrvRg6nC6ZDRdf-_Dgp8pUitoNLL7EVA@mail.gmail.com>

Hi Michael,
so from today Stretch is current stable it will be really cool to have
packages from freeswitch repository .

Thanks!

On Tue, Jun 13, 2017 at 4:14 AM, Michael Jerris <mike at jerris.com> wrote:

> announcements will come out when we have real dates.
>
> On Jun 12, 2017, at 9:32 PM, Peter Rex <prex5609 at gmail.com> wrote:
>
> Thanks Michael. Hate to do this to you, but is there an estimate on 1.8
> timeframe? Mailing list shows people were talking about configs and feature
> requests in January, but can't see much else. Maybe I'm not looking in the
> right place.
>
> On Mon, Jun 12, 2017 at 6:44 PM, Michael Jerris <mike at jerris.com> wrote:
>
>> Stretch won’t build yet.  I’ll have some patches over the next few weeks
>> to fix that.  1.8 when released will likely target Stretch as its primary
>> but still a bunch of testing to do.  The patches to fix build for stretch
>> will go back into 1.6 branch, once they are complete and tested.
>>
>>
>> On Jun 12, 2017, at 8:28 PM, Peter Rex <prex5609 at gmail.com> wrote:
>>
>> Stretch is the new stable on Saturday. I've looked through Confluence and
>> the mailing lists but I can't find anything relevant. I see interesting
>> possibilities at http://files.freeswitch.org/repo/deb, but I thought I
>> would ask the mailing list if there's a plan yet to add or move the
>> _production_ build to Stretch.
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org
>



-- 
Best regards,
Volodymyr
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170618/64aa3cbd/attachment.html>

From joelists at tm.net.uk  Sun Jun 18 22:52:08 2017
From: joelists at tm.net.uk (Joseph Waite)
Date: Sun, 18 Jun 2017 23:52:08 +0100
Subject: [Freeswitch-users] Call Dropping
Message-ID: <222F1F16-373C-484A-B290-200CB13F2CAD@tm.net.uk>

Hi Guys

Using FreeSwitch with Radius linked to JeraSoft VCS billing system. 

I am sending a Call from a SIPP originator, through the FreeSwitch box and back out to another SIPP terminator scenario.
The call goes through ok, everything happens as it should, however the call immediately drops, I have done egrep’s of both sides of the call and the BYE is defiantly coming from Freeswitch for some reason but I cannot work out why. Anyone any ideas?
I am attaching the FreeSwitch logs plus the egrep’s
If I register zipper on my laptop to FS and make a call works fine.

2017-06-18 22:26:34.199519 [NOTICE] switch_channel.c:1104 New Channel sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 [15021010-8f64-439f-8dbb-1afe090c44a5]
2017-06-18 22:26:34.199519 [DEBUG] switch_core_state_machine.c:584 (sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060) Running State Change CS_NEW (Cur 1 Tot 67)
2017-06-18 22:26:34.199519 [DEBUG] sofia.c:9837 sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 receiving invite from 185.35.228.51:5060 version: 1.6.18 git 6e79667 2017-06-12 21:14:49Z 64bit
2017-06-18 22:26:34.219492 [DEBUG] sofia.c:7048 Channel sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 entering state [received][100]
2017-06-18 22:26:34.219492 [DEBUG] sofia.c:7058 Remote SDP:
v=0
o=user1 53655765 2353687637 IN IP4 185.35.228.51
s=-
c=IN IP4 185.35.228.51
t=0 0
m=audio 6000 RTP/AVP 0
a=rtpmap:0 PCMU/8000

2017-06-18 22:26:34.219492 [DEBUG] sofia.c:7450 (sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060) State Change CS_NEW -> CS_INIT
2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:603 (sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060) State NEW
2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:584 (sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060) Running State Change CS_INIT (Cur 1 Tot 67)
2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:627 (sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060) State INIT
2017-06-18 22:26:34.219492 [DEBUG] mod_sofia.c:90 sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 SOFIA INIT
2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:40 sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 Standard INIT
2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:48 (sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060) State Change CS_INIT -> CS_ROUTING
2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:627 (sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060) State INIT going to sleep
2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:584 (sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060) Running State Change CS_ROUTING (Cur 1 Tot 67)
2017-06-18 22:26:34.219492 [DEBUG] switch_channel.c:2249 (sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060) Callstate Change DOWN -> RINGING
2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:643 (sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060) State ROUTING
2017-06-18 22:26:34.219492 [DEBUG] mod_sofia.c:143 sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 SOFIA ROUTING
2017-06-18 22:26:34.219492 [ERR] mod_xml_radius.c:933 Result of true match: 185.35.228.40 == ^185\.35\.229\.30 
2017-06-18 22:26:34.219492 [INFO] mod_xml_radius.c:986 mod_xml_radius:  Accounting Start success
2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:236 sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 Standard ROUTING
2017-06-18 22:26:34.219492 [INFO] mod_dialplan_xml.c:637 Processing sipp <sipp>->441554555666 in context public
Dialplan: sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 parsing [public->unloop] continue=false
Dialplan: sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false
Dialplan: sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false
Dialplan: sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 parsing [public->outside_call] continue=true
Dialplan: sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 Absolute Condition [outside_call]
Dialplan: sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 Action set(outside_call=true) 
Dialplan: sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 Action export(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) 
Dialplan: sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 parsing [public->call_debug] continue=true
Dialplan: sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 Regex (FAIL) [call_debug] ${call_debug}(false) =~ /^true$/ break=never
Dialplan: sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 parsing [public->rejections] continue=false
Dialplan: sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 Regex (FAIL) [rejections] ${radius_auth_result}() =~ /2/ break=on-false
Dialplan: sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 parsing [public->timedouts] continue=false
Dialplan: sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 Regex (FAIL) [timedouts] ${radius_auth_result}() =~ /1/ break=on-false
Dialplan: sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 parsing [public->JeraSoft VCS Routing] continue=false
Dialplan: sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 Regex (PASS) [JeraSoft VCS Routing] destination_number(441554555666) =~ /^(.+)$/ break=on-false
Dialplan: sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 Action export(nolocal:h323-call-origin=originate) 
Dialplan: sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 Action set(sip_h_X-accountcode=${accountcode}) 
Dialplan: sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 Action set(call_direction=outbound) 
Dialplan: sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 Action set(hangup_after_bridge=true) 
Dialplan: sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 Action set(continue_on_fail=true) 
Dialplan: sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 Action set(inherit_codec=true) 
Dialplan: sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 Action set(call_timeout=20) 
Dialplan: sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 Action set(fail_on_single_reject=USER_BUSY) 
Dialplan: sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 Action set(origination_caller_id_name=${sip_req_user}) 
Dialplan: sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 Action set(origination_caller_id_number=${sip_from_user}) 
Dialplan: sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 Action set(execute_on_answer=sched_hangup +${h323-credit-time} alloted_timeout) 
Dialplan: sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 Action bridge({sip_invite_from_uri=sip:${sip_from_user}@${sip_network_ip}}sofia/external/${destination_number}@185.35.229.30:5060)
Dialplan: sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 Action hangup(${bridge_hangup_cause}) 
2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:286 (sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060) State Change CS_ROUTING -> CS_EXECUTE
2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:643 (sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060) State ROUTING going to sleep
2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:584 (sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060) Running State Change CS_EXECUTE (Cur 1 Tot 67)
2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:650 (sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060) State EXECUTE
2017-06-18 22:26:34.219492 [DEBUG] mod_sofia.c:198 sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 SOFIA EXECUTE
2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:328 sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 Standard EXECUTE
EXECUTE sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 set(outside_call=true)
2017-06-18 22:26:34.219492 [DEBUG] mod_dptools.c:1530 SET sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 [outside_call]=[true]
EXECUTE sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 export(RFC2822_DATE=Sun, 18 Jun 2017 22:26:34 +0100)
2017-06-18 22:26:34.219492 [DEBUG] switch_channel.c:1296 EXPORT (export_vars) [RFC2822_DATE]=[Sun, 18 Jun 2017 22:26:34 +0100]
EXECUTE sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 export(nolocal:h323-call-origin=originate)
2017-06-18 22:26:34.219492 [DEBUG] switch_channel.c:1296 EXPORT (export_vars) (REMOTE ONLY) [h323-call-origin]=[originate]
EXECUTE sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 set(sip_h_X-accountcode=)
2017-06-18 22:26:34.219492 [DEBUG] mod_dptools.c:1530 SET sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 [sip_h_X-accountcode]=[UNDEF]
EXECUTE sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 set(call_direction=outbound)
2017-06-18 22:26:34.219492 [DEBUG] mod_dptools.c:1530 SET sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 [call_direction]=[outbound]
EXECUTE sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 set(hangup_after_bridge=true)
2017-06-18 22:26:34.219492 [DEBUG] mod_dptools.c:1530 SET sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 [hangup_after_bridge]=[true]
EXECUTE sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 set(continue_on_fail=true)
2017-06-18 22:26:34.219492 [DEBUG] mod_dptools.c:1530 SET sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 [continue_on_fail]=[true]
EXECUTE sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 set(inherit_codec=true)
2017-06-18 22:26:34.219492 [DEBUG] mod_dptools.c:1530 SET sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 [inherit_codec]=[true]
EXECUTE sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 set(call_timeout=20)
2017-06-18 22:26:34.219492 [DEBUG] mod_dptools.c:1530 SET sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 [call_timeout]=[20]
EXECUTE sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 set(fail_on_single_reject=USER_BUSY)
2017-06-18 22:26:34.219492 [DEBUG] mod_dptools.c:1530 SET sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 [fail_on_single_reject]=[USER_BUSY]
EXECUTE sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 set(origination_caller_id_name=441554555666)
2017-06-18 22:26:34.219492 [DEBUG] mod_dptools.c:1530 SET sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 [origination_caller_id_name]=[441554555666]
EXECUTE sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 set(origination_caller_id_number=sipp)
2017-06-18 22:26:34.219492 [DEBUG] mod_dptools.c:1530 SET sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 [origination_caller_id_number]=[sipp]
EXECUTE sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 set(execute_on_answer=sched_hangup + alloted_timeout)
2017-06-18 22:26:34.219492 [DEBUG] mod_dptools.c:1530 SET sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 [execute_on_answer]=[sched_hangup + alloted_timeout]
EXECUTE sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 bridge({sip_invite_from_uri=sip:sipp at 185.35.228.51}sofia/external/441554555666 at 185.35.229.30 <mailto:sofia/external/441554555666 at 185.35.229.30>:5060)
2017-06-18 22:26:34.219492 [DEBUG] switch_channel.c:1250 sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 EXPORTING[export_vars] [RFC2822_DATE]=[Sun, 18 Jun 2017 22:26:34 +0100] to event
2017-06-18 22:26:34.219492 [DEBUG] switch_channel.c:1250 sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 EXPORTING[export_vars] [h323-call-origin]=[originate] to event
2017-06-18 22:26:34.219492 [DEBUG] switch_ivr_originate.c:2142 Parsing global variables
2017-06-18 22:26:34.219492 [NOTICE] switch_channel.c:1104 New Channel sofia/external/441554555666 at 185.35.229.30 <mailto:sofia/external/441554555666 at 185.35.229.30>:5060 [96c1a021-5195-41ce-b903-08b98816d70d]
2017-06-18 22:26:34.219492 [DEBUG] mod_sofia.c:4819 (sofia/external/441554555666 at 185.35.229.30 <mailto:sofia/external/441554555666 at 185.35.229.30>:5060) State Change CS_NEW -> CS_INIT
2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:584 (sofia/external/441554555666 at 185.35.229.30 <mailto:sofia/external/441554555666 at 185.35.229.30>:5060) Running State Change CS_INIT (Cur 2 Tot 68)
2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:627 (sofia/external/441554555666 at 185.35.229.30 <mailto:sofia/external/441554555666 at 185.35.229.30>:5060) State INIT
2017-06-18 22:26:34.219492 [DEBUG] mod_sofia.c:90 sofia/external/441554555666 at 185.35.229.30 <mailto:sofia/external/441554555666 at 185.35.229.30>:5060 SOFIA INIT
2017-06-18 22:26:34.219492 [DEBUG] sofia_glue.c:1295 sofia/external/441554555666 at 185.35.229.30 <mailto:sofia/external/441554555666 at 185.35.229.30>:5060 sending invite version: 1.6.18 git 6e79667 2017-06-12 21:14:49Z 64bit
Local SDP:
v=0
o=FreeSWITCH 1497789362 1497789363 IN IP4 185.35.228.40
s=FreeSWITCH
c=IN IP4 185.35.228.40
t=0 0
m=audio 31832 RTP/AVP 0 101 13
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:13 CN/8000
a=ptime:20
a=sendrecv

2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:40 sofia/external/441554555666 at 185.35.229.30 <mailto:sofia/external/441554555666 at 185.35.229.30>:5060 Standard INIT
2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:48 (sofia/external/441554555666 at 185.35.229.30 <mailto:sofia/external/441554555666 at 185.35.229.30>:5060) State Change CS_INIT -> CS_ROUTING
2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:627 (sofia/external/441554555666 at 185.35.229.30 <mailto:sofia/external/441554555666 at 185.35.229.30>:5060) State INIT going to sleep
2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:584 (sofia/external/441554555666 at 185.35.229.30 <mailto:sofia/external/441554555666 at 185.35.229.30>:5060) Running State Change CS_ROUTING (Cur 2 Tot 68)
2017-06-18 22:26:34.219492 [DEBUG] sofia.c:7048 Channel sofia/external/441554555666 at 185.35.229.30 <mailto:sofia/external/441554555666 at 185.35.229.30>:5060 entering state [calling][0]
2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:643 (sofia/external/441554555666 at 185.35.229.30 <mailto:sofia/external/441554555666 at 185.35.229.30>:5060) State ROUTING
2017-06-18 22:26:34.219492 [DEBUG] mod_sofia.c:143 sofia/external/441554555666 at 185.35.229.30 <mailto:sofia/external/441554555666 at 185.35.229.30>:5060 SOFIA ROUTING
2017-06-18 22:26:34.219492 [DEBUG] switch_ivr_originate.c:67 (sofia/external/441554555666 at 185.35.229.30 <mailto:sofia/external/441554555666 at 185.35.229.30>:5060) State Change CS_ROUTING -> CS_CONSUME_MEDIA
2017-06-18 22:26:34.219492 [ERR] mod_xml_radius.c:930 Didn't match: 185.35.229.30:5060 == ^185\.35\.229\.30 
2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:643 (sofia/external/441554555666 at 185.35.229.30 <mailto:sofia/external/441554555666 at 185.35.229.30>:5060) State ROUTING going to sleep
2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:584 (sofia/external/441554555666 at 185.35.229.30 <mailto:sofia/external/441554555666 at 185.35.229.30>:5060) Running State Change CS_CONSUME_MEDIA (Cur 2 Tot 68)
2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:662 (sofia/external/441554555666 at 185.35.229.30 <mailto:sofia/external/441554555666 at 185.35.229.30>:5060) State CONSUME_MEDIA
2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:662 (sofia/external/441554555666 at 185.35.229.30 <mailto:sofia/external/441554555666 at 185.35.229.30>:5060) State CONSUME_MEDIA going to sleep
2017-06-18 22:26:34.259480 [DEBUG] sofia_glue.c:1295 sofia/external/441554555666 at 185.35.229.30 <mailto:sofia/external/441554555666 at 185.35.229.30>:5060 sending invite version: 1.6.18 git 6e79667 2017-06-12 21:14:49Z 64bit
Local SDP:
v=0
o=FreeSWITCH 1497789362 1497789364 IN IP4 185.35.228.40
s=FreeSWITCH
c=IN IP4 185.35.228.40
t=0 0
m=audio 31832 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

2017-06-18 22:26:34.259480 [DEBUG] sofia.c:7048 Channel sofia/external/441554555666 at 185.35.229.30 <mailto:sofia/external/441554555666 at 185.35.229.30>:5060 entering state [calling][0]
2017-06-18 22:26:34.259480 [DEBUG] sofia.c:7048 Channel sofia/external/441554555666 at 185.35.229.30 <mailto:sofia/external/441554555666 at 185.35.229.30>:5060 entering state [proceeding][180]
2017-06-18 22:26:34.259480 [NOTICE] sofia.c:7156 Ring-Ready sofia/external/441554555666 at 185.35.229.30 <mailto:sofia/external/441554555666 at 185.35.229.30>:5060!
2017-06-18 22:26:34.259480 [DEBUG] switch_channel.c:3346 (sofia/external/441554555666 at 185.35.229.30 <mailto:sofia/external/441554555666 at 185.35.229.30>:5060) Callstate Change DOWN -> RINGING
2017-06-18 22:26:34.259480 [DEBUG] sofia.c:7048 Channel sofia/external/441554555666 at 185.35.229.30 <mailto:sofia/external/441554555666 at 185.35.229.30>:5060 entering state [completing][200]
2017-06-18 22:26:34.259480 [DEBUG] sofia.c:7058 Remote SDP:
v=0
o=user1 53655765 2353687637 IN IP4 185.35.228.48
s=-
c=IN IP4 185.35.228.48
t=0 0
m=audio 6000 RTP/AVP 0
a=rtpmap:0 PCMU/8000

2017-06-18 22:26:34.259480 [DEBUG] sofia.c:7048 Channel sofia/external/441554555666 at 185.35.229.30 <mailto:sofia/external/441554555666 at 185.35.229.30>:5060 entering state [ready][200]
2017-06-18 22:26:34.259480 [DEBUG] switch_core_media.c:4445 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]
2017-06-18 22:26:34.259480 [DEBUG] switch_core_media.c:4500 Audio Codec Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match
2017-06-18 22:26:34.259480 [DEBUG] switch_core_media.c:3057 Set Codec sofia/external/441554555666 at 185.35.229.30 <mailto:sofia/external/441554555666 at 185.35.229.30>:5060 PCMU/8000 20 ms 160 samples 64000 bits 1 channels
2017-06-18 22:26:34.259480 [DEBUG] switch_core_codec.c:111 sofia/external/441554555666 at 185.35.229.30 <mailto:sofia/external/441554555666 at 185.35.229.30>:5060 Original read codec set to PCMU:0
2017-06-18 22:26:34.259480 [DEBUG] switch_core_media.c:4770 No 2833 in SDP.  Disable 2833 dtmf and switch to INFO
2017-06-18 22:26:34.259480 [DEBUG] switch_core_media.c:6874 AUDIO RTP [sofia/external/441554555666 at 185.35.229.30 <mailto:sofia/external/441554555666 at 185.35.229.30>:5060] 185.35.228.40 port 31832 -> 185.35.228.48 port 6000 codec: 0 ms: 20
2017-06-18 22:26:34.259480 [DEBUG] switch_rtp.c:4108 Starting timer [soft] 160 bytes per 20ms
2017-06-18 22:26:34.279530 [DEBUG] switch_core_media.c:7205 sofia/external/441554555666 at 185.35.229.30 <mailto:sofia/external/441554555666 at 185.35.229.30>:5060 Set rtp dtmf delay to 40
2017-06-18 22:26:34.279530 [NOTICE] sofia.c:8182 Channel [sofia/external/441554555666 at 185.35.229.30 <mailto:sofia/external/441554555666 at 185.35.229.30>:5060] has been answered
2017-06-18 22:26:34.279530 [DEBUG] switch_channel.c:3773 (sofia/external/441554555666 at 185.35.229.30 <mailto:sofia/external/441554555666 at 185.35.229.30>:5060) Callstate Change RINGING -> ACTIVE
2017-06-18 22:26:34.279530 [NOTICE] mod_sofia.c:2273 Ring-Ready sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060!
2017-06-18 22:26:34.279530 [DEBUG] sofia.c:7048 Channel sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 entering state [early][180]
2017-06-18 22:26:34.279530 [NOTICE] switch_ivr_originate.c:525 Ring Ready sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060!
2017-06-18 22:26:34.279530 [DEBUG] switch_ivr_originate.c:410 Setting codec string on sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 to PCMU at 8000h@20i
2017-06-18 22:26:34.279530 [DEBUG] switch_core_media.c:4445 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]
2017-06-18 22:26:34.279530 [DEBUG] switch_core_media.c:4500 Audio Codec Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match
2017-06-18 22:26:34.279530 [DEBUG] switch_core_media.c:3057 Set Codec sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 PCMU/8000 20 ms 160 samples 64000 bits 1 channels
2017-06-18 22:26:34.279530 [DEBUG] switch_core_codec.c:111 sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 Original read codec set to PCMU:0
2017-06-18 22:26:34.279530 [DEBUG] switch_core_media.c:4770 No 2833 in SDP.  Disable 2833 dtmf and switch to INFO
2017-06-18 22:26:34.279530 [DEBUG] switch_core_media.c:6874 AUDIO RTP [sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060] 185.35.228.40 port 23728 -> 185.35.228.51 port 6000 codec: 0 ms: 20
2017-06-18 22:26:34.279530 [DEBUG] switch_rtp.c:4108 Starting timer [soft] 160 bytes per 20ms
2017-06-18 22:26:34.279530 [DEBUG] switch_core_media.c:7205 sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 Set rtp dtmf delay to 40
2017-06-18 22:26:34.279530 [NOTICE] sofia_media.c:92 Pre-Answer sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060!
2017-06-18 22:26:34.279530 [DEBUG] switch_channel.c:3474 (sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060) Callstate Change RINGING -> EARLY
2017-06-18 22:26:34.279530 [DEBUG] switch_core_media.c:6857 Audio params are unchanged for sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060.
2017-06-18 22:26:34.279530 [DEBUG] mod_sofia.c:850 Local SDP sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060:
v=0
o=FreeSWITCH 1497797466 1497797467 IN IP4 185.35.228.40
s=FreeSWITCH
c=IN IP4 185.35.228.40
t=0 0
m=audio 23728 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=ptime:20
a=sendrecv

2017-06-18 22:26:34.279530 [NOTICE] switch_ivr_originate.c:3647 Channel [sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060] has been answered
EXECUTE sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 sched_hangup(+ alloted_timeout)
2017-06-18 22:26:34.279530 [DEBUG] sofia.c:7048 Channel sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 entering state [completed][200]
2017-06-18 22:26:34.279530 [DEBUG] sofia.c:7048 Channel sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 entering state [ready][200]
2017-06-18 22:26:34.279530 [NOTICE] mod_dptools.c:1188 Hangup sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 [CS_EXECUTE] [NORMAL_CLEARING]
2017-06-18 22:26:34.279530 [DEBUG] switch_core_session.c:2814 sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
2017-06-18 22:26:34.279530 [DEBUG] switch_channel.c:3773 (sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060) Callstate Change EARLY -> ACTIVE
2017-06-18 22:26:34.279530 [DEBUG] switch_ivr_originate.c:3647 sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 skip receive message [ANSWER_EVENT] (channel is hungup already)
2017-06-18 22:26:34.319525 [DEBUG] switch_ivr_originate.c:3848 Originate Resulted in Error Cause: 487 [ORIGINATOR_CANCEL]
2017-06-18 22:26:34.319525 [NOTICE] switch_ivr_originate.c:3938 Hangup sofia/external/441554555666 at 185.35.229.30 <mailto:sofia/external/441554555666 at 185.35.229.30>:5060 [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL]
2017-06-18 22:26:34.319525 [DEBUG] switch_core_state_machine.c:584 (sofia/external/441554555666 at 185.35.229.30 <mailto:sofia/external/441554555666 at 185.35.229.30>:5060) Running State Change CS_HANGUP (Cur 2 Tot 68)
2017-06-18 22:26:34.319525 [DEBUG] switch_core_state_machine.c:850 (sofia/external/441554555666 at 185.35.229.30 <mailto:sofia/external/441554555666 at 185.35.229.30>:5060) Callstate Change ACTIVE -> HANGUP
2017-06-18 22:26:34.319525 [DEBUG] switch_core_state_machine.c:852 (sofia/external/441554555666 at 185.35.229.30 <mailto:sofia/external/441554555666 at 185.35.229.30>:5060) State HANGUP
2017-06-18 22:26:34.319525 [DEBUG] mod_sofia.c:438 Channel sofia/external/441554555666 at 185.35.229.30 <mailto:sofia/external/441554555666 at 185.35.229.30>:5060 hanging up, cause: ORIGINATOR_CANCEL
2017-06-18 22:26:34.319525 [INFO] mod_dptools.c:3418 Originate Failed.  Cause: ORIGINATOR_CANCEL
2017-06-18 22:26:34.319525 [DEBUG] mod_sofia.c:491 Sending BYE to sofia/external/441554555666 at 185.35.229.30 <mailto:sofia/external/441554555666 at 185.35.229.30>:5060
2017-06-18 22:26:34.319525 [DEBUG] switch_core_state_machine.c:60 sofia/external/441554555666 at 185.35.229.30 <mailto:sofia/external/441554555666 at 185.35.229.30>:5060 Standard HANGUP, cause: ORIGINATOR_CANCEL
2017-06-18 22:26:34.319525 [DEBUG] switch_core_state_machine.c:852 (sofia/external/441554555666 at 185.35.229.30 <mailto:sofia/external/441554555666 at 185.35.229.30>:5060) State HANGUP going to sleep
2017-06-18 22:26:34.319525 [DEBUG] switch_core_state_machine.c:619 (sofia/external/441554555666 at 185.35.229.30 <mailto:sofia/external/441554555666 at 185.35.229.30>:5060) State Change CS_HANGUP -> CS_REPORTING
2017-06-18 22:26:34.319525 [DEBUG] switch_core_session.c:2814 sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
2017-06-18 22:26:34.319525 [DEBUG] switch_core_state_machine.c:584 (sofia/external/441554555666 at 185.35.229.30 <mailto:sofia/external/441554555666 at 185.35.229.30>:5060) Running State Change CS_REPORTING (Cur 2 Tot 68)
2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:650 (sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060) State EXECUTE going to sleep
2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:584 (sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060) Running State Change CS_HANGUP (Cur 2 Tot 68)
2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:938 (sofia/external/441554555666 at 185.35.229.30 <mailto:sofia/external/441554555666 at 185.35.229.30>:5060) State REPORTING
2017-06-18 22:26:34.339569 [ERR] mod_xml_radius.c:930 Didn't match: 185.35.229.30 == ^185\.35\.229\.30 
2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:850 (sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060) Callstate Change ACTIVE -> HANGUP
2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:174 sofia/external/441554555666 at 185.35.229.30 <mailto:sofia/external/441554555666 at 185.35.229.30>:5060 Standard REPORTING, cause: ORIGINATOR_CANCEL
2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:938 (sofia/external/441554555666 at 185.35.229.30 <mailto:sofia/external/441554555666 at 185.35.229.30>:5060) State REPORTING going to sleep
2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:852 (sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060) State HANGUP
2017-06-18 22:26:34.339569 [DEBUG] mod_sofia.c:438 Channel sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 hanging up, cause: NORMAL_CLEARING
2017-06-18 22:26:34.339569 [DEBUG] mod_sofia.c:491 Sending BYE to sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060
2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:60 sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 Standard HANGUP, cause: NORMAL_CLEARING
2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:852 (sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060) State HANGUP going to sleep
2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:619 (sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060) State Change CS_HANGUP -> CS_REPORTING
2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:584 (sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060) Running State Change CS_REPORTING (Cur 2 Tot 68)
2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:938 (sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060) State REPORTING
2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:610 (sofia/external/441554555666 at 185.35.229.30 <mailto:sofia/external/441554555666 at 185.35.229.30>:5060) State Change CS_REPORTING -> CS_DESTROY
2017-06-18 22:26:34.339569 [DEBUG] switch_core_session.c:1664 Session 68 (sofia/external/441554555666 at 185.35.229.30 <mailto:sofia/external/441554555666 at 185.35.229.30>:5060) Locked, Waiting on external entities
2017-06-18 22:26:34.339569 [ERR] mod_xml_radius.c:933 Result of true match: 185.35.228.40 == ^185\.35\.229\.30 
2017-06-18 22:26:34.339569 [NOTICE] switch_core_session.c:1682 Session 68 (sofia/external/441554555666 at 185.35.229.30 <mailto:sofia/external/441554555666 at 185.35.229.30>:5060) Ended
2017-06-18 22:26:34.339569 [NOTICE] switch_core_session.c:1686 Close Channel sofia/external/441554555666 at 185.35.229.30 <mailto:sofia/external/441554555666 at 185.35.229.30>:5060 [CS_DESTROY]
2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:741 (sofia/external/441554555666 at 185.35.229.30 <mailto:sofia/external/441554555666 at 185.35.229.30>:5060) Running State Change CS_DESTROY (Cur 1 Tot 68)
2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:751 (sofia/external/441554555666 at 185.35.229.30 <mailto:sofia/external/441554555666 at 185.35.229.30>:5060) State DESTROY
2017-06-18 22:26:34.339569 [DEBUG] mod_sofia.c:343 sofia/external/441554555666 at 185.35.229.30 <mailto:sofia/external/441554555666 at 185.35.229.30>:5060 SOFIA DESTROY
2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:181 sofia/external/441554555666 at 185.35.229.30 <mailto:sofia/external/441554555666 at 185.35.229.30>:5060 Standard DESTROY
2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:751 (sofia/external/441554555666 at 185.35.229.30 <mailto:sofia/external/441554555666 at 185.35.229.30>:5060) State DESTROY going to sleep
2017-06-18 22:26:34.339569 [INFO] mod_xml_radius.c:1044 mod_xml_radius:  Accounting Stop success
2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:174 sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 Standard REPORTING, cause: NORMAL_CLEARING
2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:938 (sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060) State REPORTING going to sleep
2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:610 (sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060) State Change CS_REPORTING -> CS_DESTROY
2017-06-18 22:26:34.339569 [DEBUG] switch_core_session.c:1664 Session 67 (sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060) Locked, Waiting on external entiti
2017-06-18 22:26:34.339569 [NOTICE] switch_core_session.c:1682 Session 67 (sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060) Ended
2017-06-18 22:26:34.339569 [NOTICE] switch_core_session.c:1686 Close Channel sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 [CS_DESTROY]
2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:741 (sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060) Running State Change CS_DESTROY (Cur 0 Tot 68)
2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:751 (sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060) State DESTROY
2017-06-18 22:26:34.339569 [DEBUG] mod_sofia.c:343 sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 SOFIA DESTROY
2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:181 sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 Standard DESTROY
2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:751 (sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060) State DESTROY going to sleep


U 185.35.228.51:5060 -> 185.35.228.40:5080
INVITE sip:441554555666 at 185.35.228.40:5080 <sip:441554555666 at 185.35.228.40:5080> SIP/2.0.
Via: SIP/2.0/UDP 185.35.228.51:5060;branch=z9hG4bK-27036-1-0.
From: sipp <sip:sipp at 185.35.228.51:5060 <sip:sipp at 185.35.228.51:5060>>;tag=27036SIPpTag001.
To: 441554555666 <sip:441554555666 at 185.35.228.40:5080 <sip:441554555666 at 185.35.228.40:5080>>.
Call-ID: 1-27036 at 185.35.228.51 <mailto:1-27036 at 185.35.228.51>.
CSeq: 1 INVITE.
Contact: sip:sipp at 185.35.228.51:5060 <sip:sipp at 185.35.228.51:5060>.
Max-Forwards: 70.
Subject: Performance Test.
Content-Type: application/sdp.
Content-Length:   137.
.
v=0.
o=user1 53655765 2353687637 IN IP4 185.35.228.51.
s=-.
c=IN IP4 185.35.228.51.
t=0 0.
m=audio 6000 RTP/AVP 0.
a=rtpmap:0 PCMU/8000.

#
U 185.35.228.40:5080 -> 185.35.228.51:5060
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP 185.35.228.51:5060;branch=z9hG4bK-27036-1-0.
From: sipp <sip:sipp at 185.35.228.51:5060 <sip:sipp at 185.35.228.51:5060>>;tag=27036SIPpTag001.
To: 441554555666 <sip:441554555666 at 185.35.228.40:5080 <sip:441554555666 at 185.35.228.40:5080>>.
Call-ID: 1-27036 at 185.35.228.51 <mailto:1-27036 at 185.35.228.51>.
CSeq: 1 INVITE.
User-Agent: FreeSWITCH-mod_sofia/1.6.18+git~20170612T211449Z~6e79667c0a~64bit.
Content-Length: 0.
.

#
U 185.35.228.40:5080 -> 185.35.228.51:5060
SIP/2.0 180 Ringing.
Via: SIP/2.0/UDP 185.35.228.51:5060;branch=z9hG4bK-27036-1-0.
From: sipp <sip:sipp at 185.35.228.51:5060 <sip:sipp at 185.35.228.51:5060>>;tag=27036SIPpTag001.
To: 441554555666 <sip:441554555666 at 185.35.228.40:5080 <sip:441554555666 at 185.35.228.40:5080>>;tag=91vp8601aS4Qp.
Call-ID: 1-27036 at 185.35.228.51 <mailto:1-27036 at 185.35.228.51>.
CSeq: 1 INVITE.
Contact: <sip:441554555666 at 185.35.228.40:5080;transport=udp <sip:441554555666 at 185.35.228.40:5080;transport=udp>>.
User-Agent: FreeSWITCH-mod_sofia/1.6.18+git~20170612T211449Z~6e79667c0a~64bit.
Accept: application/sdp.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY.
Supported: timer, path, replaces.
Allow-Events: talk, hold, conference, refer.
Content-Length: 0.
Remote-Party-ID: "Outbound Call" <sip:441554555666 at 185.35.228.40 <sip:441554555666 at 185.35.228.40>>;party=calling;privacy=off;screen=no.
.

#
U 185.35.228.40:5080 -> 185.35.228.51:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 185.35.228.51:5060;branch=z9hG4bK-27036-1-0.
From: sipp <sip:sipp at 185.35.228.51:5060 <sip:sipp at 185.35.228.51:5060>>;tag=27036SIPpTag001.
To: 441554555666 <sip:441554555666 at 185.35.228.40:5080 <sip:441554555666 at 185.35.228.40:5080>>;tag=91vp8601aS4Qp.
Call-ID: 1-27036 at 185.35.228.51 <mailto:1-27036 at 185.35.228.51>.
CSeq: 1 INVITE.
Contact: <sip:441554555666 at 185.35.228.40:5080;transport=udp <sip:441554555666 at 185.35.228.40:5080;transport=udp>>.
User-Agent: FreeSWITCH-mod_sofia/1.6.18+git~20170612T211449Z~6e79667c0a~64bit.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY.
Supported: timer, path, replaces.
Allow-Events: talk, hold, conference, refer.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 166.
Remote-Party-ID: "Outbound Call" <sip:441554555666 at 185.35.228.40 <sip:441554555666 at 185.35.228.40>>;party=calling;privacy=off;screen=no.
.
v=0.
o=FreeSWITCH 1497797499 1497797500 IN IP4 185.35.228.40.
s=FreeSWITCH.
c=IN IP4 185.35.228.40.
t=0 0.
m=audio 25252 RTP/AVP 0.
a=rtpmap:0 PCMU/8000.
a=ptime:20.

#
U 185.35.228.51:5060 -> 185.35.228.40:5080
ACK sip:441554555666 at 185.35.228.40:5080 <sip:441554555666 at 185.35.228.40:5080> SIP/2.0.
Via: SIP/2.0/UDP 185.35.228.51:5060;branch=z9hG4bK-27036-1-5.
From: sipp <sip:sipp at 185.35.228.51:5060 <sip:sipp at 185.35.228.51:5060>>;tag=27036SIPpTag001.
To: 441554555666 <sip:441554555666 at 185.35.228.40:5080 <sip:441554555666 at 185.35.228.40:5080>>;tag=91vp8601aS4Qp.
Call-ID: 1-27036 at 185.35.228.51 <mailto:1-27036 at 185.35.228.51>.
CSeq: 1 ACK.
Contact: sip:sipp at 185.35.228.51:5060 <sip:sipp at 185.35.228.51:5060>.
Max-Forwards: 70.
Subject: Performance Test.
Content-Length: 0.
.

#
U 185.35.228.40:5080 -> 185.35.228.51:5060
BYE sip:sipp at 185.35.228.51:5060 <sip:sipp at 185.35.228.51:5060> SIP/2.0.
Via: SIP/2.0/UDP 185.35.228.40:5080;rport;branch=z9hG4bKUBe0HvDjHF1XN.
Max-Forwards: 70.
From: 441554555666 <sip:441554555666 at 185.35.228.40:5080 <sip:441554555666 at 185.35.228.40:5080>>;tag=91vp8601aS4Qp.
To: sipp <sip:sipp at 185.35.228.51:5060 <sip:sipp at 185.35.228.51:5060>>;tag=27036SIPpTag001.
Call-ID: 1-27036 at 185.35.228.51 <mailto:1-27036 at 185.35.228.51>.
CSeq: 108575311 BYE.
User-Agent: FreeSWITCH-mod_sofia/1.6.18+git~20170612T211449Z~6e79667c0a~64bit.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY.
Supported: timer, path, replaces.
Reason: Q.850;cause=16;text="NORMAL_CLEARING".
Content-Length: 0.
.

#
U 185.35.228.51:5060 -> 185.35.228.40:5080
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 185.35.228.40:5080;rport;branch=z9hG4bKUBe0HvDjHF1XN.
From: 441554555666 <sip:441554555666 at 185.35.228.40:5080 <sip:441554555666 at 185.35.228.40:5080>>;tag=91vp8601aS4Qp.
To: sipp <sip:sipp at 185.35.228.51:5060 <sip:sipp at 185.35.228.51:5060>>;tag=27036SIPpTag001.
Call-ID: 1-27036 at 185.35.228.51 <mailto:1-27036 at 185.35.228.51>.
CSeq: 108575311 BYE.
Contact: <sip:185.35.228.51:5060;transport=UDP <sip:185.35.228.51:5060;transport=UDP>>.
Content-Length: 0.



NGREP of SIP messages from FS to terminator


U 185.35.228.40:5080 -> 185.35.228.48:5060
INVITE sip:441554555666 at 185.35.228.48:5060 SIP/2.0.
Via: SIP/2.0/UDP 185.35.228.40:5080;rport;branch=z9hG4bKXX0HNjFSB1D3c.
Max-Forwards: 69.
From: "sipp" <sip:sipp at 185.35.228.51>;tag=eFUjHeNKv5KNg.
To: <sip:441554555666 at 185.35.229.30:5060>.
Call-ID: c755e7c8-cf13-1235-25a9-363165383663.
CSeq: 108575420 INVITE.
Contact: <sip:mod_sofia at 185.35.228.40:5080>.
User-Agent: FreeSWITCH-mod_sofia/1.6.18+git~20170612T211449Z~6e79667c0a~64bit.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY.
Supported: timer, path, replaces.
Allow-Events: talk, hold, conference, refer.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 222.
X-FS-Support: update_display,send_info.
Remote-Party-ID: "sipp" <sip:sipp at 185.35.228.40>;party=calling;screen=yes;privacy=off.
.
v=0.
o=FreeSWITCH 1497801741 1497801743 IN IP4 185.35.228.40.
s=FreeSWITCH.
c=IN IP4 185.35.228.40.
t=0 0.
m=audio 21228 RTP/AVP 0 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.

#
U 185.35.228.48:5060 -> 185.35.228.40:5080
SIP/2.0 180 Ringing.
Via: SIP/2.0/UDP 185.35.228.40:5080;rport;branch=z9hG4bKXX0HNjFSB1D3c.
From: "sipp" <sip:sipp at 185.35.228.51>;tag=eFUjHeNKv5KNg.
To: <sip:441554555666 at 185.35.229.30:5060>;tag=31480SIPpTag018.
Call-ID: c755e7c8-cf13-1235-25a9-363165383663.
CSeq: 108575420 INVITE.
Contact: <sip:185.35.228.48:5060;transport=UDP>.
Content-Length: 0.
.

#
U 185.35.228.48:5060 -> 185.35.228.40:5080
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 185.35.228.40:5080;rport;branch=z9hG4bKXX0HNjFSB1D3c.
From: "sipp" <sip:sipp at 185.35.228.51>;tag=eFUjHeNKv5KNg.
To: <sip:441554555666 at 185.35.229.30:5060>;tag=31480SIPpTag018.
Call-ID: c755e7c8-cf13-1235-25a9-363165383663.
CSeq: 108575420 INVITE.
Contact: <sip:185.35.228.48:5060;transport=UDP>.
Content-Type: application/sdp.
Content-Length:   137.
.
v=0.
o=user1 53655765 2353687637 IN IP4 185.35.228.48.
s=-.
c=IN IP4 185.35.228.48.
t=0 0.
m=audio 6000 RTP/AVP 0.
a=rtpmap:0 PCMU/8000.

#
U 185.35.228.40:5080 -> 185.35.228.48:5060
ACK sip:185.35.228.48:5060;transport=UDP SIP/2.0.
Via: SIP/2.0/UDP 185.35.228.40:5080;rport;branch=z9hG4bKy6SaQD0v893Nr.
Max-Forwards: 70.
From: "sipp" <sip:sipp at 185.35.228.51>;tag=eFUjHeNKv5KNg.
To: <sip:441554555666 at 185.35.229.30:5060>;tag=31480SIPpTag018.
Call-ID: c755e7c8-cf13-1235-25a9-363165383663.
CSeq: 108575420 ACK.
Contact: <sip:mod_sofia at 185.35.228.40:5080>.
Content-Length: 0.
.

#
U 185.35.228.40:5080 -> 185.35.228.48:5060
BYE sip:185.35.228.48:5060;transport=UDP SIP/2.0.
Via: SIP/2.0/UDP 185.35.228.40:5080;rport;branch=z9hG4bKZFK3r8g05jt8K.
Max-Forwards: 70.
From: "sipp" <sip:sipp at 185.35.228.51>;tag=eFUjHeNKv5KNg.
To: <sip:441554555666 at 185.35.229.30:5060>;tag=31480SIPpTag018.
Call-ID: c755e7c8-cf13-1235-25a9-363165383663.
CSeq: 108575421 BYE.
User-Agent: FreeSWITCH-mod_sofia/1.6.18+git~20170612T211449Z~6e79667c0a~64bit.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY.
Supported: timer, path, replaces.
Reason: SIP;cause=487;text="ORIGINATOR_CANCEL".
Content-Length: 0.
.

#
U 185.35.228.48:5060 -> 185.35.228.40:5080
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 185.35.228.40:5080;rport;branch=z9hG4bKZFK3r8g05jt8K.
From: "sipp" <sip:sipp at 185.35.228.51>;tag=eFUjHeNKv5KNg.
To: <sip:441554555666 at 185.35.229.30:5060>;tag=31480SIPpTag018.
Call-ID: c755e7c8-cf13-1235-25a9-363165383663.
CSeq: 108575421 BYE.
Contact: <sip:185.35.228.48:5060;transport=UDP>.
Content-Length: 0.
.

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170618/66834292/attachment-0001.html>

From colin.morelli at gmail.com  Sun Jun 18 23:08:18 2017
From: colin.morelli at gmail.com (Colin Morelli)
Date: Sun, 18 Jun 2017 19:08:18 -0400
Subject: [Freeswitch-users] Call Dropping
In-Reply-To: <222F1F16-373C-484A-B290-200CB13F2CAD@tm.net.uk>
References: <222F1F16-373C-484A-B290-200CB13F2CAD@tm.net.uk>
Message-ID: <CAPtU-Up9YWb2j6u2wZ6KS6WEZST2++CibUqNeXJOwnPY7FfHaA@mail.gmail.com>

You've got an execute_on_answer of sched_hangup(+${h323-credit-time}
alloted_timeout)

Immediately after your call is answered:
2017-06-18 22:26:34.279530 [NOTICE] switch_ivr_originate.c:3647 Channel [
sofia/external/sipp at 185.35.228.51:5060] has been answered
EXECUTE sofia/external/sipp at 185.35.228.51:5060 sched_hangup(+
alloted_timeout)

It would seem that h323-credit-time is not being set, which is causing
sched_hangup to immediately hangup the call on answer.

On Sun, Jun 18, 2017 at 6:52 PM, Joseph Waite <joelists at tm.net.uk> wrote:

> Hi Guys
>
> Using FreeSwitch with Radius linked to JeraSoft VCS billing system.
>
> I am sending a Call from a SIPP originator, through the FreeSwitch box and
> back out to another SIPP terminator scenario.
> The call goes through ok, everything happens as it should, however the
> call immediately drops, I have done egrep’s of both sides of the call and
> the BYE is defiantly coming from Freeswitch for some reason but I cannot
> work out why. Anyone any ideas?
> I am attaching the FreeSwitch logs plus the egrep’s
> If I register zipper on my laptop to FS and make a call works fine.
>
> 2017-06-18 22:26:34.199519 [NOTICE] switch_channel.c:1104 New Channel
> sofia/external/sipp at 185.35.228.51:5060 [15021010-8f64-439f-8dbb-
> 1afe090c44a5]
> 2017-06-18 22:26:34.199519 [DEBUG] switch_core_state_machine.c:584 (
> sofia/external/sipp at 185.35.228.51:5060) Running State Change CS_NEW (Cur
> 1 Tot 67)
> 2017-06-18 22:26:34.199519 [DEBUG] sofia.c:9837 sofia/external/
> sipp at 185.35.228.51:5060 receiving invite from 185.35.228.51:5060 version:
> 1.6.18 git 6e79667 2017-06-12 21:14:49Z 64bit
> 2017-06-18 22:26:34.219492 [DEBUG] sofia.c:7048 Channel
> sofia/external/sipp at 185.35.228.51:5060 entering state [received][100]
> 2017-06-18 22:26:34.219492 [DEBUG] sofia.c:7058 Remote SDP:
> v=0
> o=user1 53655765 2353687637 IN IP4 185.35.228.51
> s=-
> c=IN IP4 185.35.228.51
> t=0 0
> m=audio 6000 RTP/AVP 0
> a=rtpmap:0 PCMU/8000
>
> 2017-06-18 22:26:34.219492 [DEBUG] sofia.c:7450 (
> sofia/external/sipp at 185.35.228.51:5060) State Change CS_NEW -> CS_INIT
> 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:603 (
> sofia/external/sipp at 185.35.228.51:5060) State NEW
> 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:584 (
> sofia/external/sipp at 185.35.228.51:5060) Running State Change CS_INIT (Cur
> 1 Tot 67)
> 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:627 (
> sofia/external/sipp at 185.35.228.51:5060) State INIT
> 2017-06-18 22:26:34.219492 [DEBUG] mod_sofia.c:90 sofia/external/
> sipp at 185.35.228.51:5060 SOFIA INIT
> 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:40
> sofia/external/sipp at 185.35.228.51:5060 Standard INIT
> 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:48 (
> sofia/external/sipp at 185.35.228.51:5060) State Change CS_INIT -> CS_ROUTING
> 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:627 (
> sofia/external/sipp at 185.35.228.51:5060) State INIT going to sleep
> 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:584 (
> sofia/external/sipp at 185.35.228.51:5060) Running State Change CS_ROUTING
> (Cur 1 Tot 67)
> 2017-06-18 22:26:34.219492 [DEBUG] switch_channel.c:2249 (
> sofia/external/sipp at 185.35.228.51:5060) Callstate Change DOWN -> RINGING
> 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:643 (
> sofia/external/sipp at 185.35.228.51:5060) State ROUTING
> 2017-06-18 22:26:34.219492 [DEBUG] mod_sofia.c:143 sofia/
> external/sipp at 185.35.228.51:5060 SOFIA ROUTING
> 2017-06-18 22:26:34.219492 [ERR] mod_xml_radius.c:933 Result of true
> match: 185.35.228.40 == ^185\.35\.229\.30
> 2017-06-18 22:26:34.219492 [INFO] mod_xml_radius.c:986 mod_xml_radius:
> Accounting Start success
> 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:236
> sofia/external/sipp at 185.35.228.51:5060 Standard ROUTING
> 2017-06-18 22:26:34.219492 [INFO] mod_dialplan_xml.c:637 Processing sipp
> <sipp>->441554555666 in context public
> Dialplan: sofia/external/sipp at 185.35.228.51:5060 parsing [public->unloop]
> continue=false
> Dialplan: sofia/external/sipp at 185.35.228.51:5060 Regex (PASS) [unloop]
> ${unroll_loops}(true) =~ /^true$/ break=on-false
> Dialplan: sofia/external/sipp at 185.35.228.51:5060 Regex (FAIL) [unloop]
> ${sip_looped_call}() =~ /^true$/ break=on-false
> Dialplan: sofia/external/sipp at 185.35.228.51:5060 parsing
> [public->outside_call] continue=true
> Dialplan: sofia/external/sipp at 185.35.228.51:5060 Absolute Condition
> [outside_call]
> Dialplan: sofia/external/sipp at 185.35.228.51:5060 Action
> set(outside_call=true)
> Dialplan: sofia/external/sipp at 185.35.228.51:5060 Action
> export(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)})
> Dialplan: sofia/external/sipp at 185.35.228.51:5060 parsing
> [public->call_debug] continue=true
> Dialplan: sofia/external/sipp at 185.35.228.51:5060 Regex (FAIL)
> [call_debug] ${call_debug}(false) =~ /^true$/ break=never
> Dialplan: sofia/external/sipp at 185.35.228.51:5060 parsing
> [public->rejections] continue=false
> Dialplan: sofia/external/sipp at 185.35.228.51:5060 Regex (FAIL)
> [rejections] ${radius_auth_result}() =~ /2/ break=on-false
> Dialplan: sofia/external/sipp at 185.35.228.51:5060 parsing
> [public->timedouts] continue=false
> Dialplan: sofia/external/sipp at 185.35.228.51:5060 Regex (FAIL) [timedouts]
> ${radius_auth_result}() =~ /1/ break=on-false
> Dialplan: sofia/external/sipp at 185.35.228.51:5060 parsing
> [public->JeraSoft VCS Routing] continue=false
> Dialplan: sofia/external/sipp at 185.35.228.51:5060 Regex (PASS) [JeraSoft
> VCS Routing] destination_number(441554555666) =~ /^(.+)$/ break=on-false
> Dialplan: sofia/external/sipp at 185.35.228.51:5060 Action
> export(nolocal:h323-call-origin=originate)
> Dialplan: sofia/external/sipp at 185.35.228.51:5060 Action
> set(sip_h_X-accountcode=${accountcode})
> Dialplan: sofia/external/sipp at 185.35.228.51:5060 Action
> set(call_direction=outbound)
> Dialplan: sofia/external/sipp at 185.35.228.51:5060 Action
> set(hangup_after_bridge=true)
> Dialplan: sofia/external/sipp at 185.35.228.51:5060 Action
> set(continue_on_fail=true)
> Dialplan: sofia/external/sipp at 185.35.228.51:5060 Action
> set(inherit_codec=true)
> Dialplan: sofia/external/sipp at 185.35.228.51:5060 Action
> set(call_timeout=20)
> Dialplan: sofia/external/sipp at 185.35.228.51:5060 Action
> set(fail_on_single_reject=USER_BUSY)
> Dialplan: sofia/external/sipp at 185.35.228.51:5060 Action
> set(origination_caller_id_name=${sip_req_user})
> Dialplan: sofia/external/sipp at 185.35.228.51:5060 Action
> set(origination_caller_id_number=${sip_from_user})
> Dialplan: sofia/external/sipp at 185.35.228.51:5060 Action
> set(execute_on_answer=sched_hangup +${h323-credit-time} alloted_timeout)
> Dialplan: sofia/external/sipp at 185.35.228.51:5060 Action
> bridge({sip_invite_from_uri=sip:${sip_from_user}@${sip_
> network_ip}}sofia/external/${destination_number}@185.35.229.30:5060
> <http://destination_number%7D at 185.35.229.30:5060>)
> Dialplan: sofia/external/sipp at 185.35.228.51:5060 Action
> hangup(${bridge_hangup_cause})
> 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:286 (
> sofia/external/sipp at 185.35.228.51:5060) State Change CS_ROUTING ->
> CS_EXECUTE
> 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:643 (
> sofia/external/sipp at 185.35.228.51:5060) State ROUTING going to sleep
> 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:584 (
> sofia/external/sipp at 185.35.228.51:5060) Running State Change CS_EXECUTE
> (Cur 1 Tot 67)
> 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:650 (
> sofia/external/sipp at 185.35.228.51:5060) State EXECUTE
> 2017-06-18 22:26:34.219492 [DEBUG] mod_sofia.c:198 sofia/
> external/sipp at 185.35.228.51:5060 SOFIA EXECUTE
> 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:328
> sofia/external/sipp at 185.35.228.51:5060 Standard EXECUTE
> EXECUTE sofia/external/sipp at 185.35.228.51:5060 set(outside_call=true)
> 2017-06-18 22:26:34.219492 [DEBUG] mod_dptools.c:1530 SET
> sofia/external/sipp at 185.35.228.51:5060 [outside_call]=[true]
> EXECUTE sofia/external/sipp at 185.35.228.51:5060 export(RFC2822_DATE=Sun,
> 18 Jun 2017 22:26:34 +0100)
> 2017-06-18 22:26:34.219492 [DEBUG] switch_channel.c:1296 EXPORT
> (export_vars) [RFC2822_DATE]=[Sun, 18 Jun 2017 22:26:34 +0100]
> EXECUTE sofia/external/sipp at 185.35.228.51:5060 export(nolocal:h323-call-
> origin=originate)
> 2017-06-18 22:26:34.219492 [DEBUG] switch_channel.c:1296 EXPORT
> (export_vars) (REMOTE ONLY) [h323-call-origin]=[originate]
> EXECUTE sofia/external/sipp at 185.35.228.51:5060 set(sip_h_X-accountcode=)
> 2017-06-18 22:26:34.219492 [DEBUG] mod_dptools.c:1530 SET
> sofia/external/sipp at 185.35.228.51:5060 [sip_h_X-accountcode]=[UNDEF]
> EXECUTE sofia/external/sipp at 185.35.228.51:5060
> set(call_direction=outbound)
> 2017-06-18 22:26:34.219492 [DEBUG] mod_dptools.c:1530 SET
> sofia/external/sipp at 185.35.228.51:5060 [call_direction]=[outbound]
> EXECUTE sofia/external/sipp at 185.35.228.51:5060
> set(hangup_after_bridge=true)
> 2017-06-18 22:26:34.219492 [DEBUG] mod_dptools.c:1530 SET
> sofia/external/sipp at 185.35.228.51:5060 [hangup_after_bridge]=[true]
> EXECUTE sofia/external/sipp at 185.35.228.51:5060 set(continue_on_fail=true)
> 2017-06-18 22:26:34.219492 [DEBUG] mod_dptools.c:1530 SET
> sofia/external/sipp at 185.35.228.51:5060 [continue_on_fail]=[true]
> EXECUTE sofia/external/sipp at 185.35.228.51:5060 set(inherit_codec=true)
> 2017-06-18 22:26:34.219492 [DEBUG] mod_dptools.c:1530 SET
> sofia/external/sipp at 185.35.228.51:5060 [inherit_codec]=[true]
> EXECUTE sofia/external/sipp at 185.35.228.51:5060 set(call_timeout=20)
> 2017-06-18 22:26:34.219492 [DEBUG] mod_dptools.c:1530 SET
> sofia/external/sipp at 185.35.228.51:5060 [call_timeout]=[20]
> EXECUTE sofia/external/sipp at 185.35.228.51:5060 set(fail_on_single_reject=
> USER_BUSY)
> 2017-06-18 22:26:34.219492 [DEBUG] mod_dptools.c:1530 SET
> sofia/external/sipp at 185.35.228.51:5060 [fail_on_single_reject]=[USER_BUSY]
> EXECUTE sofia/external/sipp at 185.35.228.51:5060 set(origination_caller_id_
> name=441554555666)
> 2017-06-18 22:26:34.219492 [DEBUG] mod_dptools.c:1530 SET
> sofia/external/sipp at 185.35.228.51:5060 [origination_caller_id_name]=[
> 441554555666]
> EXECUTE sofia/external/sipp at 185.35.228.51:5060 set(origination_caller_id_
> number=sipp)
> 2017-06-18 22:26:34.219492 [DEBUG] mod_dptools.c:1530 SET
> sofia/external/sipp at 185.35.228.51:5060 [origination_caller_id_number]
> =[sipp]
> EXECUTE sofia/external/sipp at 185.35.228.51:5060
> set(execute_on_answer=sched_hangup + alloted_timeout)
> 2017-06-18 22:26:34.219492 [DEBUG] mod_dptools.c:1530 SET
> sofia/external/sipp at 185.35.228.51:5060 [execute_on_answer]=[sched_hangup
> + alloted_timeout]
> EXECUTE sofia/external/sipp at 185.35.228.51:5060
> bridge({sip_invite_from_uri=sip:sipp at 185.35.228.51}sofia/
> external/441554555666 at 185.35.229.30:5060)
> 2017-06-18 22:26:34.219492 [DEBUG] switch_channel.c:1250 sofia/
> external/sipp at 185.35.228.51:5060 EXPORTING[export_vars]
> [RFC2822_DATE]=[Sun, 18 Jun 2017 22:26:34 +0100] to event
> 2017-06-18 22:26:34.219492 [DEBUG] switch_channel.c:1250 sofia/
> external/sipp at 185.35.228.51:5060 EXPORTING[export_vars]
> [h323-call-origin]=[originate] to event
> 2017-06-18 22:26:34.219492 [DEBUG] switch_ivr_originate.c:2142 Parsing
> global variables
> 2017-06-18 22:26:34.219492 [NOTICE] switch_channel.c:1104 New Channel
> sofia/external/441554555666 at 185.35.229.30:5060 [96c1a021-5195-41ce-b903-
> 08b98816d70d]
> 2017-06-18 22:26:34.219492 [DEBUG] mod_sofia.c:4819 (
> sofia/external/441554555666 at 185.35.229.30:5060) State Change CS_NEW ->
> CS_INIT
> 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:584 (
> sofia/external/441554555666 at 185.35.229.30:5060) Running State Change
> CS_INIT (Cur 2 Tot 68)
> 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:627 (
> sofia/external/441554555666 at 185.35.229.30:5060) State INIT
> 2017-06-18 22:26:34.219492 [DEBUG] mod_sofia.c:90 sofia/external/
> 441554555666 at 185.35.229.30:5060 SOFIA INIT
> 2017-06-18 22:26:34.219492 [DEBUG] sofia_glue.c:1295 sofia/
> external/441554555666 at 185.35.229.30:5060 sending invite version: 1.6.18
> git 6e79667 2017-06-12 21:14:49Z 64bit
> Local SDP:
> v=0
> o=FreeSWITCH 1497789362 1497789363 IN IP4 185.35.228.40
> s=FreeSWITCH
> c=IN IP4 185.35.228.40
> t=0 0
> m=audio 31832 RTP/AVP 0 101 13
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=rtpmap:13 CN/8000
> a=ptime:20
> a=sendrecv
>
> 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:40
> sofia/external/441554555666 at 185.35.229.30:5060 Standard INIT
> 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:48 (
> sofia/external/441554555666 at 185.35.229.30:5060) State Change CS_INIT ->
> CS_ROUTING
> 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:627 (
> sofia/external/441554555666 at 185.35.229.30:5060) State INIT going to sleep
> 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:584 (
> sofia/external/441554555666 at 185.35.229.30:5060) Running State Change
> CS_ROUTING (Cur 2 Tot 68)
> 2017-06-18 22:26:34.219492 [DEBUG] sofia.c:7048 Channel sofia/external/
> 441554555666 at 185.35.229.30:5060 entering state [calling][0]
> 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:643 (
> sofia/external/441554555666 at 185.35.229.30:5060) State ROUTING
> 2017-06-18 22:26:34.219492 [DEBUG] mod_sofia.c:143 sofia/
> external/441554555666 at 185.35.229.30:5060 SOFIA ROUTING
> 2017-06-18 22:26:34.219492 [DEBUG] switch_ivr_originate.c:67 (
> sofia/external/441554555666 at 185.35.229.30:5060) State Change CS_ROUTING
> -> CS_CONSUME_MEDIA
> 2017-06-18 22:26:34.219492 [ERR] mod_xml_radius.c:930 Didn't match:
> 185.35.229.30:5060 == ^185\.35\.229\.30
> 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:643 (
> sofia/external/441554555666 at 185.35.229.30:5060) State ROUTING going to
> sleep
> 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:584 (
> sofia/external/441554555666 at 185.35.229.30:5060) Running State Change
> CS_CONSUME_MEDIA (Cur 2 Tot 68)
> 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:662 (
> sofia/external/441554555666 at 185.35.229.30:5060) State CONSUME_MEDIA
> 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:662 (
> sofia/external/441554555666 at 185.35.229.30:5060) State CONSUME_MEDIA going
> to sleep
> 2017-06-18 22:26:34.259480 [DEBUG] sofia_glue.c:1295 sofia/
> external/441554555666 at 185.35.229.30:5060 sending invite version: 1.6.18
> git 6e79667 2017-06-12 21:14:49Z 64bit
> Local SDP:
> v=0
> o=FreeSWITCH 1497789362 1497789364 IN IP4 185.35.228.40
> s=FreeSWITCH
> c=IN IP4 185.35.228.40
> t=0 0
> m=audio 31832 RTP/AVP 0 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
> a=sendrecv
>
> 2017-06-18 22:26:34.259480 [DEBUG] sofia.c:7048 Channel sofia/external/
> 441554555666 at 185.35.229.30:5060 entering state [calling][0]
> 2017-06-18 22:26:34.259480 [DEBUG] sofia.c:7048 Channel sofia/external/
> 441554555666 at 185.35.229.30:5060 entering state [proceeding][180]
> 2017-06-18 22:26:34.259480 [NOTICE] sofia.c:7156 Ring-Ready
> sofia/external/441554555666 at 185.35.229.30:5060!
> 2017-06-18 22:26:34.259480 [DEBUG] switch_channel.c:3346 (
> sofia/external/441554555666 at 185.35.229.30:5060) Callstate Change DOWN ->
> RINGING
> 2017-06-18 22:26:34.259480 [DEBUG] sofia.c:7048 Channel sofia/external/
> 441554555666 at 185.35.229.30:5060 entering state [completing][200]
> 2017-06-18 22:26:34.259480 [DEBUG] sofia.c:7058 Remote SDP:
> v=0
> o=user1 53655765 2353687637 IN IP4 185.35.228.48
> s=-
> c=IN IP4 185.35.228.48
> t=0 0
> m=audio 6000 RTP/AVP 0
> a=rtpmap:0 PCMU/8000
>
> 2017-06-18 22:26:34.259480 [DEBUG] sofia.c:7048 Channel sofia/external/
> 441554555666 at 185.35.229.30:5060 entering state [ready][200]
> 2017-06-18 22:26:34.259480 [DEBUG] switch_core_media.c:4445 Audio Codec
> Compare [PCMU:0:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]
> 2017-06-18 22:26:34.259480 [DEBUG] switch_core_media.c:4500 Audio Codec
> Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match
> 2017-06-18 22:26:34.259480 [DEBUG] switch_core_media.c:3057 Set Codec
> sofia/external/441554555666 at 185.35.229.30:5060 PCMU/8000 20 ms 160
> samples 64000 bits 1 channels
> 2017-06-18 22:26:34.259480 [DEBUG] switch_core_codec.c:111 sofia/
> external/441554555666 at 185.35.229.30:5060 Original read codec set to PCMU:0
> 2017-06-18 22:26:34.259480 [DEBUG] switch_core_media.c:4770 No 2833 in
> SDP.  Disable 2833 dtmf and switch to INFO
> 2017-06-18 22:26:34.259480 [DEBUG] switch_core_media.c:6874 AUDIO RTP [
> sofia/external/441554555666 at 185.35.229.30:5060] 185.35.228.40 port 31832
> -> 185.35.228.48 port 6000 codec: 0 ms: 20
> 2017-06-18 22:26:34.259480 [DEBUG] switch_rtp.c:4108 Starting timer [soft]
> 160 bytes per 20ms
> 2017-06-18 22:26:34.279530 [DEBUG] switch_core_media.c:7205 sofia
> /external/441554555666 at 185.35.229.30:5060 Set rtp dtmf delay to 40
> 2017-06-18 22:26:34.279530 [NOTICE] sofia.c:8182 Channel [
> sofia/external/441554555666 at 185.35.229.30:5060] has been answered
> 2017-06-18 22:26:34.279530 [DEBUG] switch_channel.c:3773 (
> sofia/external/441554555666 at 185.35.229.30:5060) Callstate Change RINGING
> -> ACTIVE
> 2017-06-18 22:26:34.279530 [NOTICE] mod_sofia.c:2273 Ring-Ready
> sofia/external/sipp at 185.35.228.51:5060!
> 2017-06-18 22:26:34.279530 [DEBUG] sofia.c:7048 Channel
> sofia/external/sipp at 185.35.228.51:5060 entering state [early][180]
> 2017-06-18 22:26:34.279530 [NOTICE] switch_ivr_originate.c:525 Ring Ready
> sofia/external/sipp at 185.35.228.51:5060!
> 2017-06-18 22:26:34.279530 [DEBUG] switch_ivr_originate.c:410 Setting
> codec string on sofia/external/sipp at 185.35.228.51:5060 to PCMU at 8000h@20i
> 2017-06-18 22:26:34.279530 [DEBUG] switch_core_media.c:4445 Audio Codec
> Compare [PCMU:0:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]
> 2017-06-18 22:26:34.279530 [DEBUG] switch_core_media.c:4500 Audio Codec
> Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match
> 2017-06-18 22:26:34.279530 [DEBUG] switch_core_media.c:3057 Set Codec
> sofia/external/sipp at 185.35.228.51:5060 PCMU/8000 20 ms 160 samples 64000
> bits 1 channels
> 2017-06-18 22:26:34.279530 [DEBUG] switch_core_codec.c:111 sofia/
> external/sipp at 185.35.228.51:5060 Original read codec set to PCMU:0
> 2017-06-18 22:26:34.279530 [DEBUG] switch_core_media.c:4770 No 2833 in
> SDP.  Disable 2833 dtmf and switch to INFO
> 2017-06-18 22:26:34.279530 [DEBUG] switch_core_media.c:6874 AUDIO RTP [
> sofia/external/sipp at 185.35.228.51:5060] 185.35.228.40 port 23728 ->
> 185.35.228.51 port 6000 codec: 0 ms: 20
> 2017-06-18 22:26:34.279530 [DEBUG] switch_rtp.c:4108 Starting timer [soft]
> 160 bytes per 20ms
> 2017-06-18 22:26:34.279530 [DEBUG] switch_core_media.c:7205 sofia
> /external/sipp at 185.35.228.51:5060 Set rtp dtmf delay to 40
> 2017-06-18 22:26:34.279530 [NOTICE] sofia_media.c:92 Pre-Answer
> sofia/external/sipp at 185.35.228.51:5060!
> 2017-06-18 22:26:34.279530 [DEBUG] switch_channel.c:3474 (
> sofia/external/sipp at 185.35.228.51:5060) Callstate Change RINGING -> EARLY
> 2017-06-18 22:26:34.279530 [DEBUG] switch_core_media.c:6857 Audio params
> are unchanged for sofia/external/sipp at 185.35.228.51:5060.
> 2017-06-18 22:26:34.279530 [DEBUG] mod_sofia.c:850 Local SDP
> sofia/external/sipp at 185.35.228.51:5060:
> v=0
> o=FreeSWITCH 1497797466 1497797467 IN IP4 185.35.228.40
> s=FreeSWITCH
> c=IN IP4 185.35.228.40
> t=0 0
> m=audio 23728 RTP/AVP 0
> a=rtpmap:0 PCMU/8000
> a=ptime:20
> a=sendrecv
>
> 2017-06-18 22:26:34.279530 [NOTICE] switch_ivr_originate.c:3647 Channel [
> sofia/external/sipp at 185.35.228.51:5060] has been answered
> EXECUTE sofia/external/sipp at 185.35.228.51:5060 sched_hangup(+
> alloted_timeout)
> 2017-06-18 22:26:34.279530 [DEBUG] sofia.c:7048 Channel
> sofia/external/sipp at 185.35.228.51:5060 entering state [completed][200]
> 2017-06-18 22:26:34.279530 [DEBUG] sofia.c:7048 Channel
> sofia/external/sipp at 185.35.228.51:5060 entering state [ready][200]
> 2017-06-18 22:26:34.279530 [NOTICE] mod_dptools.c:1188 Hangup
> sofia/external/sipp at 185.35.228.51:5060 [CS_EXECUTE] [NORMAL_CLEARING]
> 2017-06-18 22:26:34.279530 [DEBUG] switch_core_session.c:2814 sof
> ia/external/sipp at 185.35.228.51:5060 skip receive message
> [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
> 2017-06-18 22:26:34.279530 [DEBUG] switch_channel.c:3773 (
> sofia/external/sipp at 185.35.228.51:5060) Callstate Change EARLY -> ACTIVE
> 2017-06-18 22:26:34.279530 [DEBUG] switch_ivr_originate.c:3647 so
> fia/external/sipp at 185.35.228.51:5060 skip receive message [ANSWER_EVENT]
> (channel is hungup already)
> 2017-06-18 22:26:34.319525 [DEBUG] switch_ivr_originate.c:3848 Originate
> Resulted in Error Cause: 487 [ORIGINATOR_CANCEL]
> 2017-06-18 22:26:34.319525 [NOTICE] switch_ivr_originate.c:3938 Hangup
> sofia/external/441554555666 at 185.35.229.30:5060 [CS_CONSUME_MEDIA]
> [ORIGINATOR_CANCEL]
> 2017-06-18 22:26:34.319525 [DEBUG] switch_core_state_machine.c:584 (
> sofia/external/441554555666 at 185.35.229.30:5060) Running State Change
> CS_HANGUP (Cur 2 Tot 68)
> 2017-06-18 22:26:34.319525 [DEBUG] switch_core_state_machine.c:850 (
> sofia/external/441554555666 at 185.35.229.30:5060) Callstate Change ACTIVE
> -> HANGUP
> 2017-06-18 22:26:34.319525 [DEBUG] switch_core_state_machine.c:852 (
> sofia/external/441554555666 at 185.35.229.30:5060) State HANGUP
> 2017-06-18 22:26:34.319525 [DEBUG] mod_sofia.c:438 Channel sofia/external/
> 441554555666 at 185.35.229.30:5060 hanging up, cause: ORIGINATOR_CANCEL
> 2017-06-18 22:26:34.319525 [INFO] mod_dptools.c:3418 Originate Failed.
> Cause: ORIGINATOR_CANCEL
> 2017-06-18 22:26:34.319525 [DEBUG] mod_sofia.c:491 Sending BYE to
> sofia/external/441554555666 at 185.35.229.30:5060
> 2017-06-18 22:26:34.319525 [DEBUG] switch_core_state_machine.c:60
> sofia/external/441554555666 at 185.35.229.30:5060 Standard HANGUP, cause:
> ORIGINATOR_CANCEL
> 2017-06-18 22:26:34.319525 [DEBUG] switch_core_state_machine.c:852 (
> sofia/external/441554555666 at 185.35.229.30:5060) State HANGUP going to
> sleep
> 2017-06-18 22:26:34.319525 [DEBUG] switch_core_state_machine.c:619 (
> sofia/external/441554555666 at 185.35.229.30:5060) State Change CS_HANGUP ->
> CS_REPORTING
> 2017-06-18 22:26:34.319525 [DEBUG] switch_core_session.c:2814 sof
> ia/external/sipp at 185.35.228.51:5060 skip receive message
> [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
> 2017-06-18 22:26:34.319525 [DEBUG] switch_core_state_machine.c:584 (
> sofia/external/441554555666 at 185.35.229.30:5060) Running State Change
> CS_REPORTING (Cur 2 Tot 68)
> 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:650 (
> sofia/external/sipp at 185.35.228.51:5060) State EXECUTE going to sleep
> 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:584 (
> sofia/external/sipp at 185.35.228.51:5060) Running State Change CS_HANGUP
> (Cur 2 Tot 68)
> 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:938 (
> sofia/external/441554555666 at 185.35.229.30:5060) State REPORTING
> 2017-06-18 22:26:34.339569 [ERR] mod_xml_radius.c:930 Didn't match:
> 185.35.229.30 == ^185\.35\.229\.30
> 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:850 (
> sofia/external/sipp at 185.35.228.51:5060) Callstate Change ACTIVE -> HANGUP
> 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:174
> sofia/external/441554555666 at 185.35.229.30:5060 Standard REPORTING, cause:
> ORIGINATOR_CANCEL
> 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:938 (
> sofia/external/441554555666 at 185.35.229.30:5060) State REPORTING going to
> sleep
> 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:852 (
> sofia/external/sipp at 185.35.228.51:5060) State HANGUP
> 2017-06-18 22:26:34.339569 [DEBUG] mod_sofia.c:438 Channel
> sofia/external/sipp at 185.35.228.51:5060 hanging up, cause: NORMAL_CLEARING
> 2017-06-18 22:26:34.339569 [DEBUG] mod_sofia.c:491 Sending BYE to
> sofia/external/sipp at 185.35.228.51:5060
> 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:60
> sofia/external/sipp at 185.35.228.51:5060 Standard HANGUP, cause:
> NORMAL_CLEARING
> 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:852 (
> sofia/external/sipp at 185.35.228.51:5060) State HANGUP going to sleep
> 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:619 (
> sofia/external/sipp at 185.35.228.51:5060) State Change CS_HANGUP ->
> CS_REPORTING
> 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:584 (
> sofia/external/sipp at 185.35.228.51:5060) Running State Change CS_REPORTING
> (Cur 2 Tot 68)
> 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:938 (
> sofia/external/sipp at 185.35.228.51:5060) State REPORTING
> 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:610 (
> sofia/external/441554555666 at 185.35.229.30:5060) State Change CS_REPORTING
> -> CS_DESTROY
> 2017-06-18 22:26:34.339569 [DEBUG] switch_core_session.c:1664 Session 68 (
> sofia/external/441554555666 at 185.35.229.30:5060) Locked, Waiting on
> external entities
> 2017-06-18 22:26:34.339569 [ERR] mod_xml_radius.c:933 Result of true
> match: 185.35.228.40 == ^185\.35\.229\.30
> 2017-06-18 22:26:34.339569 [NOTICE] switch_core_session.c:1682 Session 68 (
> sofia/external/441554555666 at 185.35.229.30:5060) Ended
> 2017-06-18 22:26:34.339569 [NOTICE] switch_core_session.c:1686 Close
> Channel sofia/external/441554555666 at 185.35.229.30:5060 [CS_DESTROY]
> 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:741 (
> sofia/external/441554555666 at 185.35.229.30:5060) Running State Change
> CS_DESTROY (Cur 1 Tot 68)
> 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:751 (
> sofia/external/441554555666 at 185.35.229.30:5060) State DESTROY
> 2017-06-18 22:26:34.339569 [DEBUG] mod_sofia.c:343 sofia/
> external/441554555666 at 185.35.229.30:5060 SOFIA DESTROY
> 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:181
> sofia/external/441554555666 at 185.35.229.30:5060 Standard DESTROY
> 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:751 (
> sofia/external/441554555666 at 185.35.229.30:5060) State DESTROY going to
> sleep
> 2017-06-18 22:26:34.339569 [INFO] mod_xml_radius.c:1044 mod_xml_radius:
> Accounting Stop success
> 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:174
> sofia/external/sipp at 185.35.228.51:5060 Standard REPORTING, cause:
> NORMAL_CLEARING
> 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:938 (
> sofia/external/sipp at 185.35.228.51:5060) State REPORTING going to sleep
> 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:610 (
> sofia/external/sipp at 185.35.228.51:5060) State Change CS_REPORTING ->
> CS_DESTROY
> 2017-06-18 22:26:34.339569 [DEBUG] switch_core_session.c:1664 Session 67 (
> sofia/external/sipp at 185.35.228.51:5060) Locked, Waiting on external entiti
> 2017-06-18 22:26:34.339569 [NOTICE] switch_core_session.c:1682 Session 67 (
> sofia/external/sipp at 185.35.228.51:5060) Ended
> 2017-06-18 22:26:34.339569 [NOTICE] switch_core_session.c:1686 Close
> Channel sofia/external/sipp at 185.35.228.51:5060 [CS_DESTROY]
> 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:741 (
> sofia/external/sipp at 185.35.228.51:5060) Running State Change CS_DESTROY
> (Cur 0 Tot 68)
> 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:751 (
> sofia/external/sipp at 185.35.228.51:5060) State DESTROY
> 2017-06-18 22:26:34.339569 [DEBUG] mod_sofia.c:343 sofia/
> external/sipp at 185.35.228.51:5060 SOFIA DESTROY
> 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:181
> sofia/external/sipp at 185.35.228.51:5060 Standard DESTROY
> 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:751 (
> sofia/external/sipp at 185.35.228.51:5060) State DESTROY going to sleep
>
>
> U 185.35.228.51:5060 -> 185.35.228.40:5080
> INVITE sip:441554555666 at 185.35.228.40:5080 SIP/2.0.
> Via: SIP/2.0/UDP 185.35.228.51:5060;branch=z9hG4bK-27036-1-0.
> From: sipp <sip:sipp at 185.35.228.51:5060>;tag=27036SIPpTag001.
> To: 441554555666 <sip:441554555666 at 185.35.228.40:5080>.
> Call-ID: 1-27036 at 185.35.228.51.
> CSeq: 1 INVITE.
> Contact: sip:sipp at 185.35.228.51:5060.
> Max-Forwards: 70.
> Subject: Performance Test.
> Content-Type: application/sdp.
> Content-Length:   137.
> .
> v=0.
> o=user1 53655765 2353687637 IN IP4 185.35.228.51.
> s=-.
> c=IN IP4 185.35.228.51.
> t=0 0.
> m=audio 6000 RTP/AVP 0.
> a=rtpmap:0 PCMU/8000.
>
> #
> U 185.35.228.40:5080 -> 185.35.228.51:5060
> SIP/2.0 100 Trying.
> Via: SIP/2.0/UDP 185.35.228.51:5060;branch=z9hG4bK-27036-1-0.
> From: sipp <sip:sipp at 185.35.228.51:5060>;tag=27036SIPpTag001.
> To: 441554555666 <sip:441554555666 at 185.35.228.40:5080>.
> Call-ID: 1-27036 at 185.35.228.51.
> CSeq: 1 INVITE.
> User-Agent: FreeSWITCH-mod_sofia/1.6.18+git~20170612T211449Z~
> 6e79667c0a~64bit.
> Content-Length: 0.
> .
>
> #
> U 185.35.228.40:5080 -> 185.35.228.51:5060
> SIP/2.0 180 Ringing.
> Via: SIP/2.0/UDP 185.35.228.51:5060;branch=z9hG4bK-27036-1-0.
> From: sipp <sip:sipp at 185.35.228.51:5060>;tag=27036SIPpTag001.
> To: 441554555666 <sip:441554555666 at 185.35.228.40:5080>;tag=91vp8601aS4Qp.
> Call-ID: 1-27036 at 185.35.228.51.
> CSeq: 1 INVITE.
> Contact: <sip:441554555666 at 185.35.228.40:5080;transport=udp>.
> User-Agent: FreeSWITCH-mod_sofia/1.6.18+git~20170612T211449Z~
> 6e79667c0a~64bit.
> Accept: application/sdp.
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
> REFER, NOTIFY.
> Supported: timer, path, replaces.
> Allow-Events: talk, hold, conference, refer.
> Content-Length: 0.
> Remote-Party-ID: "Outbound Call" <sip:441554555666 at 185.35.228.40
> >;party=calling;privacy=off;screen=no.
> .
>
> #
> U 185.35.228.40:5080 -> 185.35.228.51:5060
> SIP/2.0 200 OK.
> Via: SIP/2.0/UDP 185.35.228.51:5060;branch=z9hG4bK-27036-1-0.
> From: sipp <sip:sipp at 185.35.228.51:5060>;tag=27036SIPpTag001.
> To: 441554555666 <sip:441554555666 at 185.35.228.40:5080>;tag=91vp8601aS4Qp.
> Call-ID: 1-27036 at 185.35.228.51.
> CSeq: 1 INVITE.
> Contact: <sip:441554555666 at 185.35.228.40:5080;transport=udp>.
> User-Agent: FreeSWITCH-mod_sofia/1.6.18+git~20170612T211449Z~
> 6e79667c0a~64bit.
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
> REFER, NOTIFY.
> Supported: timer, path, replaces.
> Allow-Events: talk, hold, conference, refer.
> Content-Type: application/sdp.
> Content-Disposition: session.
> Content-Length: 166.
> Remote-Party-ID: "Outbound Call" <sip:441554555666 at 185.35.228.40
> >;party=calling;privacy=off;screen=no.
> .
> v=0.
> o=FreeSWITCH 1497797499 1497797500 IN IP4 185.35.228.40.
> s=FreeSWITCH.
> c=IN IP4 185.35.228.40.
> t=0 0.
> m=audio 25252 RTP/AVP 0.
> a=rtpmap:0 PCMU/8000.
> a=ptime:20.
>
> #
> U 185.35.228.51:5060 -> 185.35.228.40:5080
> ACK sip:441554555666 at 185.35.228.40:5080 SIP/2.0.
> Via: SIP/2.0/UDP 185.35.228.51:5060;branch=z9hG4bK-27036-1-5.
> From: sipp <sip:sipp at 185.35.228.51:5060>;tag=27036SIPpTag001.
> To: 441554555666 <sip:441554555666 at 185.35.228.40:5080>;tag=91vp8601aS4Qp.
> Call-ID: 1-27036 at 185.35.228.51.
> CSeq: 1 ACK.
> Contact: sip:sipp at 185.35.228.51:5060.
> Max-Forwards: 70.
> Subject: Performance Test.
> Content-Length: 0.
> .
>
> #
> U 185.35.228.40:5080 -> 185.35.228.51:5060
> BYE sip:sipp at 185.35.228.51:5060 SIP/2.0.
> Via: SIP/2.0/UDP 185.35.228.40:5080;rport;branch=z9hG4bKUBe0HvDjHF1XN.
> Max-Forwards: 70.
> From: 441554555666 <sip:441554555666 at 185.35.228.40:5080
> >;tag=91vp8601aS4Qp.
> To: sipp <sip:sipp at 185.35.228.51:5060>;tag=27036SIPpTag001.
> Call-ID: 1-27036 at 185.35.228.51.
> CSeq: 108575311 BYE.
> User-Agent: FreeSWITCH-mod_sofia/1.6.18+git~20170612T211449Z~
> 6e79667c0a~64bit.
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
> REFER, NOTIFY.
> Supported: timer, path, replaces.
> Reason: Q.850;cause=16;text="NORMAL_CLEARING".
> Content-Length: 0.
> .
>
> #
> U 185.35.228.51:5060 -> 185.35.228.40:5080
> SIP/2.0 200 OK.
> Via: SIP/2.0/UDP 185.35.228.40:5080;rport;branch=z9hG4bKUBe0HvDjHF1XN.
> From: 441554555666 <sip:441554555666 at 185.35.228.40:5080
> >;tag=91vp8601aS4Qp.
> To: sipp <sip:sipp at 185.35.228.51:5060>;tag=27036SIPpTag001.
> Call-ID: 1-27036 at 185.35.228.51.
> CSeq: 108575311 BYE.
> Contact: <sip:185.35.228.51:5060;transport=UDP>.
> Content-Length: 0.
>
>
>
> NGREP of SIP messages from FS to terminator
>
>
> U 185.35.228.40:5080 -> 185.35.228.48:5060
> INVITE sip:441554555666 at 185.35.228.48:5060 SIP/2.0.
> Via: SIP/2.0/UDP 185.35.228.40:5080;rport;branch=z9hG4bKXX0HNjFSB1D3c.
> Max-Forwards: 69.
> From: "sipp" <sip:sipp at 185.35.228.51>;tag=eFUjHeNKv5KNg.
> To: <sip:441554555666 at 185.35.229.30:5060>.
> Call-ID: c755e7c8-cf13-1235-25a9-363165383663.
> CSeq: 108575420 INVITE.
> Contact: <sip:mod_sofia at 185.35.228.40:5080>.
> User-Agent: FreeSWITCH-mod_sofia/1.6.18+git~20170612T211449Z~
> 6e79667c0a~64bit.
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
> REFER, NOTIFY.
> Supported: timer, path, replaces.
> Allow-Events: talk, hold, conference, refer.
> Content-Type: application/sdp.
> Content-Disposition: session.
> Content-Length: 222.
> X-FS-Support: update_display,send_info.
> Remote-Party-ID: "sipp" <sip:sipp at 185.35.228.40>;party=calling;screen=yes;
> privacy=off.
> .
> v=0.
> o=FreeSWITCH 1497801741 1497801743 IN IP4 185.35.228.40.
> s=FreeSWITCH.
> c=IN IP4 185.35.228.40.
> t=0 0.
> m=audio 21228 RTP/AVP 0 101.
> a=rtpmap:0 PCMU/8000.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-16.
> a=ptime:20.
>
> #
> U 185.35.228.48:5060 -> 185.35.228.40:5080
> SIP/2.0 180 Ringing.
> Via: SIP/2.0/UDP 185.35.228.40:5080;rport;branch=z9hG4bKXX0HNjFSB1D3c.
> From: "sipp" <sip:sipp at 185.35.228.51>;tag=eFUjHeNKv5KNg.
> To: <sip:441554555666 at 185.35.229.30:5060>;tag=31480SIPpTag018.
> Call-ID: c755e7c8-cf13-1235-25a9-363165383663.
> CSeq: 108575420 INVITE.
> Contact: <sip:185.35.228.48:5060;transport=UDP>.
> Content-Length: 0.
> .
>
> #
> U 185.35.228.48:5060 -> 185.35.228.40:5080
> SIP/2.0 200 OK.
> Via: SIP/2.0/UDP 185.35.228.40:5080;rport;branch=z9hG4bKXX0HNjFSB1D3c.
> From: "sipp" <sip:sipp at 185.35.228.51>;tag=eFUjHeNKv5KNg.
> To: <sip:441554555666 at 185.35.229.30:5060>;tag=31480SIPpTag018.
> Call-ID: c755e7c8-cf13-1235-25a9-363165383663.
> CSeq: 108575420 INVITE.
> Contact: <sip:185.35.228.48:5060;transport=UDP>.
> Content-Type: application/sdp.
> Content-Length:   137.
> .
> v=0.
> o=user1 53655765 2353687637 IN IP4 185.35.228.48.
> s=-.
> c=IN IP4 185.35.228.48.
> t=0 0.
> m=audio 6000 RTP/AVP 0.
> a=rtpmap:0 PCMU/8000.
>
> #
> U 185.35.228.40:5080 -> 185.35.228.48:5060
> ACK sip:185.35.228.48:5060;transport=UDP SIP/2.0.
> Via: SIP/2.0/UDP 185.35.228.40:5080;rport;branch=z9hG4bKy6SaQD0v893Nr.
> Max-Forwards: 70.
> From: "sipp" <sip:sipp at 185.35.228.51>;tag=eFUjHeNKv5KNg.
> To: <sip:441554555666 at 185.35.229.30:5060>;tag=31480SIPpTag018.
> Call-ID: c755e7c8-cf13-1235-25a9-363165383663.
> CSeq: 108575420 ACK.
> Contact: <sip:mod_sofia at 185.35.228.40:5080>.
> Content-Length: 0.
> .
>
> #
> U 185.35.228.40:5080 -> 185.35.228.48:5060
> BYE sip:185.35.228.48:5060;transport=UDP SIP/2.0.
> Via: SIP/2.0/UDP 185.35.228.40:5080;rport;branch=z9hG4bKZFK3r8g05jt8K.
> Max-Forwards: 70.
> From: "sipp" <sip:sipp at 185.35.228.51>;tag=eFUjHeNKv5KNg.
> To: <sip:441554555666 at 185.35.229.30:5060>;tag=31480SIPpTag018.
> Call-ID: c755e7c8-cf13-1235-25a9-363165383663.
> CSeq: 108575421 BYE.
> User-Agent: FreeSWITCH-mod_sofia/1.6.18+git~20170612T211449Z~
> 6e79667c0a~64bit.
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
> REFER, NOTIFY.
> Supported: timer, path, replaces.
> Reason: SIP;cause=487;text="ORIGINATOR_CANCEL".
> Content-Length: 0.
> .
>
> #
> U 185.35.228.48:5060 -> 185.35.228.40:5080
> SIP/2.0 200 OK.
> Via: SIP/2.0/UDP 185.35.228.40:5080;rport;branch=z9hG4bKZFK3r8g05jt8K.
> From: "sipp" <sip:sipp at 185.35.228.51>;tag=eFUjHeNKv5KNg.
> To: <sip:441554555666 at 185.35.229.30:5060>;tag=31480SIPpTag018.
> Call-ID: c755e7c8-cf13-1235-25a9-363165383663.
> CSeq: 108575421 BYE.
> Contact: <sip:185.35.228.48:5060;transport=UDP>.
> Content-Length: 0.
> .
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170618/c1b5b9b7/attachment-0001.html>

From ahmed at netelsat.net  Sun Jun 18 23:33:23 2017
From: ahmed at netelsat.net (Ahmed Sboor)
Date: Mon, 19 Jun 2017 04:33:23 +0500
Subject: [Freeswitch-users] Call Dropping
In-Reply-To: <CAPtU-Up9YWb2j6u2wZ6KS6WEZST2++CibUqNeXJOwnPY7FfHaA@mail.gmail.com>
References: <222F1F16-373C-484A-B290-200CB13F2CAD@tm.net.uk>
 <CAPtU-Up9YWb2j6u2wZ6KS6WEZST2++CibUqNeXJOwnPY7FfHaA@mail.gmail.com>
Message-ID: <CALFFxt0BWRGT+gWqWL27M-Y1WDYuLUMj+zyic72LvBNDZWjCuw@mail.gmail.com>

if on VCS , Rate table is set and balance or credit limit is also positive
and rate exist in rate table , then h323-credit-limit is also set.
you should also post mod radius debug logs.


On Mon, Jun 19, 2017 at 4:08 AM, Colin Morelli <colin.morelli at gmail.com>
wrote:

> You've got an execute_on_answer of sched_hangup(+${h323-credit-time}
> alloted_timeout)
>
> Immediately after your call is answered:
> 2017-06-18 22:26:34.279530 [NOTICE] switch_ivr_originate.c:3647 Channel [
> sofia/external/sipp at 185.35.228.51:5060] has been answered
> EXECUTE sofia/external/sipp at 185.35.228.51:5060 sched_hangup(+
> alloted_timeout)
>
> It would seem that h323-credit-time is not being set, which is causing
> sched_hangup to immediately hangup the call on answer.
>
> On Sun, Jun 18, 2017 at 6:52 PM, Joseph Waite <joelists at tm.net.uk> wrote:
>
>> Hi Guys
>>
>> Using FreeSwitch with Radius linked to JeraSoft VCS billing system.
>>
>> I am sending a Call from a SIPP originator, through the FreeSwitch box
>> and back out to another SIPP terminator scenario.
>> The call goes through ok, everything happens as it should, however the
>> call immediately drops, I have done egrep’s of both sides of the call and
>> the BYE is defiantly coming from Freeswitch for some reason but I cannot
>> work out why. Anyone any ideas?
>> I am attaching the FreeSwitch logs plus the egrep’s
>> If I register zipper on my laptop to FS and make a call works fine.
>>
>> 2017-06-18 22:26:34.199519 [NOTICE] switch_channel.c:1104 New Channel
>> sofia/external/sipp at 185.35.228.51:5060 [15021010-8f64-439f-8dbb-1afe0
>> 90c44a5]
>> 2017-06-18 22:26:34.199519 [DEBUG] switch_core_state_machine.c:584 (
>> sofia/external/sipp at 185.35.228.51:5060) Running State Change CS_NEW (Cur
>> 1 Tot 67)
>> 2017-06-18 22:26:34.199519 [DEBUG] sofia.c:9837 sofia/external/si
>> pp at 185.35.228.51:5060 receiving invite from 185.35.228.51:5060 version:
>> 1.6.18 git 6e79667 2017-06-12 21:14:49Z 64bit
>> 2017-06-18 22:26:34.219492 [DEBUG] sofia.c:7048 Channel
>> sofia/external/sipp at 185.35.228.51:5060 entering state [received][100]
>> 2017-06-18 22:26:34.219492 [DEBUG] sofia.c:7058 Remote SDP:
>> v=0
>> o=user1 53655765 2353687637 IN IP4 185.35.228.51
>> s=-
>> c=IN IP4 185.35.228.51
>> t=0 0
>> m=audio 6000 RTP/AVP 0
>> a=rtpmap:0 PCMU/8000
>>
>> 2017-06-18 22:26:34.219492 [DEBUG] sofia.c:7450 (
>> sofia/external/sipp at 185.35.228.51:5060) State Change CS_NEW -> CS_INIT
>> 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:603 (
>> sofia/external/sipp at 185.35.228.51:5060) State NEW
>> 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:584 (
>> sofia/external/sipp at 185.35.228.51:5060) Running State Change CS_INIT
>> (Cur 1 Tot 67)
>> 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:627 (
>> sofia/external/sipp at 185.35.228.51:5060) State INIT
>> 2017-06-18 22:26:34.219492 [DEBUG] mod_sofia.c:90 sofia/external/
>> sipp at 185.35.228.51:5060 SOFIA INIT
>> 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:40
>> sofia/external/sipp at 185.35.228.51:5060 Standard INIT
>> 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:48 (
>> sofia/external/sipp at 185.35.228.51:5060) State Change CS_INIT ->
>> CS_ROUTING
>> 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:627 (
>> sofia/external/sipp at 185.35.228.51:5060) State INIT going to sleep
>> 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:584 (
>> sofia/external/sipp at 185.35.228.51:5060) Running State Change CS_ROUTING
>> (Cur 1 Tot 67)
>> 2017-06-18 22:26:34.219492 [DEBUG] switch_channel.c:2249 (
>> sofia/external/sipp at 185.35.228.51:5060) Callstate Change DOWN -> RINGING
>> 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:643 (
>> sofia/external/sipp at 185.35.228.51:5060) State ROUTING
>> 2017-06-18 22:26:34.219492 [DEBUG] mod_sofia.c:143 sofia/external
>> /sipp at 185.35.228.51:5060 SOFIA ROUTING
>> 2017-06-18 22:26:34.219492 [ERR] mod_xml_radius.c:933 Result of true
>> match: 185.35.228.40 == ^185\.35\.229\.30
>> 2017-06-18 22:26:34.219492 [INFO] mod_xml_radius.c:986 mod_xml_radius:
>> Accounting Start success
>> 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:236
>> sofia/external/sipp at 185.35.228.51:5060 Standard ROUTING
>> 2017-06-18 22:26:34.219492 [INFO] mod_dialplan_xml.c:637 Processing sipp
>> <sipp>->441554555666 in context public
>> Dialplan: sofia/external/sipp at 185.35.228.51:5060 parsing
>> [public->unloop] continue=false
>> Dialplan: sofia/external/sipp at 185.35.228.51:5060 Regex (PASS) [unloop]
>> ${unroll_loops}(true) =~ /^true$/ break=on-false
>> Dialplan: sofia/external/sipp at 185.35.228.51:5060 Regex (FAIL) [unloop]
>> ${sip_looped_call}() =~ /^true$/ break=on-false
>> Dialplan: sofia/external/sipp at 185.35.228.51:5060 parsing
>> [public->outside_call] continue=true
>> Dialplan: sofia/external/sipp at 185.35.228.51:5060 Absolute Condition
>> [outside_call]
>> Dialplan: sofia/external/sipp at 185.35.228.51:5060 Action
>> set(outside_call=true)
>> Dialplan: sofia/external/sipp at 185.35.228.51:5060 Action
>> export(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)})
>> Dialplan: sofia/external/sipp at 185.35.228.51:5060 parsing
>> [public->call_debug] continue=true
>> Dialplan: sofia/external/sipp at 185.35.228.51:5060 Regex (FAIL)
>> [call_debug] ${call_debug}(false) =~ /^true$/ break=never
>> Dialplan: sofia/external/sipp at 185.35.228.51:5060 parsing
>> [public->rejections] continue=false
>> Dialplan: sofia/external/sipp at 185.35.228.51:5060 Regex (FAIL)
>> [rejections] ${radius_auth_result}() =~ /2/ break=on-false
>> Dialplan: sofia/external/sipp at 185.35.228.51:5060 parsing
>> [public->timedouts] continue=false
>> Dialplan: sofia/external/sipp at 185.35.228.51:5060 Regex (FAIL)
>> [timedouts] ${radius_auth_result}() =~ /1/ break=on-false
>> Dialplan: sofia/external/sipp at 185.35.228.51:5060 parsing
>> [public->JeraSoft VCS Routing] continue=false
>> Dialplan: sofia/external/sipp at 185.35.228.51:5060 Regex (PASS) [JeraSoft
>> VCS Routing] destination_number(441554555666) =~ /^(.+)$/ break=on-false
>> Dialplan: sofia/external/sipp at 185.35.228.51:5060 Action
>> export(nolocal:h323-call-origin=originate)
>> Dialplan: sofia/external/sipp at 185.35.228.51:5060 Action
>> set(sip_h_X-accountcode=${accountcode})
>> Dialplan: sofia/external/sipp at 185.35.228.51:5060 Action
>> set(call_direction=outbound)
>> Dialplan: sofia/external/sipp at 185.35.228.51:5060 Action
>> set(hangup_after_bridge=true)
>> Dialplan: sofia/external/sipp at 185.35.228.51:5060 Action
>> set(continue_on_fail=true)
>> Dialplan: sofia/external/sipp at 185.35.228.51:5060 Action
>> set(inherit_codec=true)
>> Dialplan: sofia/external/sipp at 185.35.228.51:5060 Action
>> set(call_timeout=20)
>> Dialplan: sofia/external/sipp at 185.35.228.51:5060 Action
>> set(fail_on_single_reject=USER_BUSY)
>> Dialplan: sofia/external/sipp at 185.35.228.51:5060 Action
>> set(origination_caller_id_name=${sip_req_user})
>> Dialplan: sofia/external/sipp at 185.35.228.51:5060 Action
>> set(origination_caller_id_number=${sip_from_user})
>> Dialplan: sofia/external/sipp at 185.35.228.51:5060 Action
>> set(execute_on_answer=sched_hangup +${h323-credit-time} alloted_timeout)
>> Dialplan: sofia/external/sipp at 185.35.228.51:5060 Action
>> bridge({sip_invite_from_uri=sip:${sip_from_user}@${sip_netwo
>> rk_ip}}sofia/external/${destination_number}@185.35.229.30:5060
>> <http://destination_number%7D at 185.35.229.30:5060>)
>> Dialplan: sofia/external/sipp at 185.35.228.51:5060 Action
>> hangup(${bridge_hangup_cause})
>> 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:286 (
>> sofia/external/sipp at 185.35.228.51:5060) State Change CS_ROUTING ->
>> CS_EXECUTE
>> 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:643 (
>> sofia/external/sipp at 185.35.228.51:5060) State ROUTING going to sleep
>> 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:584 (
>> sofia/external/sipp at 185.35.228.51:5060) Running State Change CS_EXECUTE
>> (Cur 1 Tot 67)
>> 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:650 (
>> sofia/external/sipp at 185.35.228.51:5060) State EXECUTE
>> 2017-06-18 22:26:34.219492 [DEBUG] mod_sofia.c:198 sofia/external
>> /sipp at 185.35.228.51:5060 SOFIA EXECUTE
>> 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:328
>> sofia/external/sipp at 185.35.228.51:5060 Standard EXECUTE
>> EXECUTE sofia/external/sipp at 185.35.228.51:5060 set(outside_call=true)
>> 2017-06-18 22:26:34.219492 [DEBUG] mod_dptools.c:1530 SET
>> sofia/external/sipp at 185.35.228.51:5060 [outside_call]=[true]
>> EXECUTE sofia/external/sipp at 185.35.228.51:5060 export(RFC2822_DATE=Sun,
>> 18 Jun 2017 22:26:34 +0100)
>> 2017-06-18 22:26:34.219492 [DEBUG] switch_channel.c:1296 EXPORT
>> (export_vars) [RFC2822_DATE]=[Sun, 18 Jun 2017 22:26:34 +0100]
>> EXECUTE sofia/external/sipp at 185.35.228.51:5060
>> export(nolocal:h323-call-origin=originate)
>> 2017-06-18 22:26:34.219492 [DEBUG] switch_channel.c:1296 EXPORT
>> (export_vars) (REMOTE ONLY) [h323-call-origin]=[originate]
>> EXECUTE sofia/external/sipp at 185.35.228.51:5060 set(sip_h_X-accountcode=)
>> 2017-06-18 22:26:34.219492 [DEBUG] mod_dptools.c:1530 SET
>> sofia/external/sipp at 185.35.228.51:5060 [sip_h_X-accountcode]=[UNDEF]
>> EXECUTE sofia/external/sipp at 185.35.228.51:5060
>> set(call_direction=outbound)
>> 2017-06-18 22:26:34.219492 [DEBUG] mod_dptools.c:1530 SET
>> sofia/external/sipp at 185.35.228.51:5060 [call_direction]=[outbound]
>> EXECUTE sofia/external/sipp at 185.35.228.51:5060
>> set(hangup_after_bridge=true)
>> 2017-06-18 22:26:34.219492 [DEBUG] mod_dptools.c:1530 SET
>> sofia/external/sipp at 185.35.228.51:5060 [hangup_after_bridge]=[true]
>> EXECUTE sofia/external/sipp at 185.35.228.51:5060 set(continue_on_fail=true)
>> 2017-06-18 22:26:34.219492 [DEBUG] mod_dptools.c:1530 SET
>> sofia/external/sipp at 185.35.228.51:5060 [continue_on_fail]=[true]
>> EXECUTE sofia/external/sipp at 185.35.228.51:5060 set(inherit_codec=true)
>> 2017-06-18 22:26:34.219492 [DEBUG] mod_dptools.c:1530 SET
>> sofia/external/sipp at 185.35.228.51:5060 [inherit_codec]=[true]
>> EXECUTE sofia/external/sipp at 185.35.228.51:5060 set(call_timeout=20)
>> 2017-06-18 22:26:34.219492 [DEBUG] mod_dptools.c:1530 SET
>> sofia/external/sipp at 185.35.228.51:5060 [call_timeout]=[20]
>> EXECUTE sofia/external/sipp at 185.35.228.51:5060
>> set(fail_on_single_reject=USER_BUSY)
>> 2017-06-18 22:26:34.219492 [DEBUG] mod_dptools.c:1530 SET
>> sofia/external/sipp at 185.35.228.51:5060 [fail_on_single_reject]=[USER_
>> BUSY]
>> EXECUTE sofia/external/sipp at 185.35.228.51:5060
>> set(origination_caller_id_name=441554555666)
>> 2017-06-18 22:26:34.219492 [DEBUG] mod_dptools.c:1530 SET
>> sofia/external/sipp at 185.35.228.51:5060 [origination_caller_id_name]=[
>> 441554555666]
>> EXECUTE sofia/external/sipp at 185.35.228.51:5060
>> set(origination_caller_id_number=sipp)
>> 2017-06-18 22:26:34.219492 [DEBUG] mod_dptools.c:1530 SET
>> sofia/external/sipp at 185.35.228.51:5060 [origination_caller_id_number]
>> =[sipp]
>> EXECUTE sofia/external/sipp at 185.35.228.51:5060
>> set(execute_on_answer=sched_hangup + alloted_timeout)
>> 2017-06-18 22:26:34.219492 [DEBUG] mod_dptools.c:1530 SET
>> sofia/external/sipp at 185.35.228.51:5060 [execute_on_answer]=[sched_hangup
>> + alloted_timeout]
>> EXECUTE sofia/external/sipp at 185.35.228.51:5060
>> bridge({sip_invite_from_uri=sip:sipp at 185.35.228.51}sofia/ext
>> ernal/441554555666 at 185.35.229.30:5060)
>> 2017-06-18 22:26:34.219492 [DEBUG] switch_channel.c:1250 sofia/ex
>> ternal/sipp at 185.35.228.51:5060 EXPORTING[export_vars]
>> [RFC2822_DATE]=[Sun, 18 Jun 2017 22:26:34 +0100] to event
>> 2017-06-18 22:26:34.219492 [DEBUG] switch_channel.c:1250 sofia/ex
>> ternal/sipp at 185.35.228.51:5060 EXPORTING[export_vars]
>> [h323-call-origin]=[originate] to event
>> 2017-06-18 22:26:34.219492 [DEBUG] switch_ivr_originate.c:2142 Parsing
>> global variables
>> 2017-06-18 22:26:34.219492 [NOTICE] switch_channel.c:1104 New Channel
>> sofia/external/441554555666 at 185.35.229.30:5060
>> [96c1a021-5195-41ce-b903-08b98816d70d]
>> 2017-06-18 22:26:34.219492 [DEBUG] mod_sofia.c:4819 (
>> sofia/external/441554555666 at 185.35.229.30:5060) State Change CS_NEW ->
>> CS_INIT
>> 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:584 (
>> sofia/external/441554555666 at 185.35.229.30:5060) Running State Change
>> CS_INIT (Cur 2 Tot 68)
>> 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:627 (
>> sofia/external/441554555666 at 185.35.229.30:5060) State INIT
>> 2017-06-18 22:26:34.219492 [DEBUG] mod_sofia.c:90 sofia/external/
>> 441554555666 at 185.35.229.30:5060 SOFIA INIT
>> 2017-06-18 22:26:34.219492 [DEBUG] sofia_glue.c:1295 sofia/extern
>> al/441554555666 at 185.35.229.30:5060 sending invite version: 1.6.18 git
>> 6e79667 2017-06-12 21:14:49Z 64bit
>> Local SDP:
>> v=0
>> o=FreeSWITCH 1497789362 1497789363 IN IP4 185.35.228.40
>> s=FreeSWITCH
>> c=IN IP4 185.35.228.40
>> t=0 0
>> m=audio 31832 RTP/AVP 0 101 13
>> a=rtpmap:0 PCMU/8000
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>> a=rtpmap:13 CN/8000
>> a=ptime:20
>> a=sendrecv
>>
>> 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:40
>> sofia/external/441554555666 at 185.35.229.30:5060 Standard INIT
>> 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:48 (
>> sofia/external/441554555666 at 185.35.229.30:5060) State Change CS_INIT ->
>> CS_ROUTING
>> 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:627 (
>> sofia/external/441554555666 at 185.35.229.30:5060) State INIT going to sleep
>> 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:584 (
>> sofia/external/441554555666 at 185.35.229.30:5060) Running State Change
>> CS_ROUTING (Cur 2 Tot 68)
>> 2017-06-18 22:26:34.219492 [DEBUG] sofia.c:7048 Channel
>> sofia/external/441554555666 at 185.35.229.30:5060 entering state
>> [calling][0]
>> 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:643 (
>> sofia/external/441554555666 at 185.35.229.30:5060) State ROUTING
>> 2017-06-18 22:26:34.219492 [DEBUG] mod_sofia.c:143 sofia/external
>> /441554555666 at 185.35.229.30:5060 SOFIA ROUTING
>> 2017-06-18 22:26:34.219492 [DEBUG] switch_ivr_originate.c:67 (
>> sofia/external/441554555666 at 185.35.229.30:5060) State Change CS_ROUTING
>> -> CS_CONSUME_MEDIA
>> 2017-06-18 22:26:34.219492 [ERR] mod_xml_radius.c:930 Didn't match:
>> 185.35.229.30:5060 == ^185\.35\.229\.30
>> 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:643 (
>> sofia/external/441554555666 at 185.35.229.30:5060) State ROUTING going to
>> sleep
>> 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:584 (
>> sofia/external/441554555666 at 185.35.229.30:5060) Running State Change
>> CS_CONSUME_MEDIA (Cur 2 Tot 68)
>> 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:662 (
>> sofia/external/441554555666 at 185.35.229.30:5060) State CONSUME_MEDIA
>> 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:662 (
>> sofia/external/441554555666 at 185.35.229.30:5060) State CONSUME_MEDIA
>> going to sleep
>> 2017-06-18 22:26:34.259480 [DEBUG] sofia_glue.c:1295 sofia/extern
>> al/441554555666 at 185.35.229.30:5060 sending invite version: 1.6.18 git
>> 6e79667 2017-06-12 21:14:49Z 64bit
>> Local SDP:
>> v=0
>> o=FreeSWITCH 1497789362 1497789364 IN IP4 185.35.228.40
>> s=FreeSWITCH
>> c=IN IP4 185.35.228.40
>> t=0 0
>> m=audio 31832 RTP/AVP 0 101
>> a=rtpmap:0 PCMU/8000
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>> a=ptime:20
>> a=sendrecv
>>
>> 2017-06-18 22:26:34.259480 [DEBUG] sofia.c:7048 Channel
>> sofia/external/441554555666 at 185.35.229.30:5060 entering state
>> [calling][0]
>> 2017-06-18 22:26:34.259480 [DEBUG] sofia.c:7048 Channel
>> sofia/external/441554555666 at 185.35.229.30:5060 entering state
>> [proceeding][180]
>> 2017-06-18 22:26:34.259480 [NOTICE] sofia.c:7156 Ring-Ready
>> sofia/external/441554555666 at 185.35.229.30:5060!
>> 2017-06-18 22:26:34.259480 [DEBUG] switch_channel.c:3346 (
>> sofia/external/441554555666 at 185.35.229.30:5060) Callstate Change DOWN ->
>> RINGING
>> 2017-06-18 22:26:34.259480 [DEBUG] sofia.c:7048 Channel
>> sofia/external/441554555666 at 185.35.229.30:5060 entering state
>> [completing][200]
>> 2017-06-18 22:26:34.259480 [DEBUG] sofia.c:7058 Remote SDP:
>> v=0
>> o=user1 53655765 2353687637 IN IP4 185.35.228.48
>> s=-
>> c=IN IP4 185.35.228.48
>> t=0 0
>> m=audio 6000 RTP/AVP 0
>> a=rtpmap:0 PCMU/8000
>>
>> 2017-06-18 22:26:34.259480 [DEBUG] sofia.c:7048 Channel
>> sofia/external/441554555666 at 185.35.229.30:5060 entering state
>> [ready][200]
>> 2017-06-18 22:26:34.259480 [DEBUG] switch_core_media.c:4445 Audio Codec
>> Compare [PCMU:0:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]
>> 2017-06-18 22:26:34.259480 [DEBUG] switch_core_media.c:4500 Audio Codec
>> Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match
>> 2017-06-18 22:26:34.259480 [DEBUG] switch_core_media.c:3057 Set Codec
>> sofia/external/441554555666 at 185.35.229.30:5060 PCMU/8000 20 ms 160
>> samples 64000 bits 1 channels
>> 2017-06-18 22:26:34.259480 [DEBUG] switch_core_codec.c:111 sofia/
>> external/441554555666 at 185.35.229.30:5060 Original read codec set to
>> PCMU:0
>> 2017-06-18 22:26:34.259480 [DEBUG] switch_core_media.c:4770 No 2833 in
>> SDP.  Disable 2833 dtmf and switch to INFO
>> 2017-06-18 22:26:34.259480 [DEBUG] switch_core_media.c:6874 AUDIO RTP [
>> sofia/external/441554555666 at 185.35.229.30:5060] 185.35.228.40 port 31832
>> -> 185.35.228.48 port 6000 codec: 0 ms: 20
>> 2017-06-18 22:26:34.259480 [DEBUG] switch_rtp.c:4108 Starting timer
>> [soft] 160 bytes per 20ms
>> 2017-06-18 22:26:34.279530 [DEBUG] switch_core_media.c:7205 sofia
>> /external/441554555666 at 185.35.229.30:5060 Set rtp dtmf delay to 40
>> 2017-06-18 22:26:34.279530 [NOTICE] sofia.c:8182 Channel [
>> sofia/external/441554555666 at 185.35.229.30:5060] has been answered
>> 2017-06-18 22:26:34.279530 [DEBUG] switch_channel.c:3773 (
>> sofia/external/441554555666 at 185.35.229.30:5060) Callstate Change RINGING
>> -> ACTIVE
>> 2017-06-18 22:26:34.279530 [NOTICE] mod_sofia.c:2273 Ring-Ready
>> sofia/external/sipp at 185.35.228.51:5060!
>> 2017-06-18 22:26:34.279530 [DEBUG] sofia.c:7048 Channel
>> sofia/external/sipp at 185.35.228.51:5060 entering state [early][180]
>> 2017-06-18 22:26:34.279530 [NOTICE] switch_ivr_originate.c:525 Ring Ready
>> sofia/external/sipp at 185.35.228.51:5060!
>> 2017-06-18 22:26:34.279530 [DEBUG] switch_ivr_originate.c:410 Setting
>> codec string on sofia/external/sipp at 185.35.228.51:5060 to PCMU at 8000h@20i
>> 2017-06-18 22:26:34.279530 [DEBUG] switch_core_media.c:4445 Audio Codec
>> Compare [PCMU:0:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]
>> 2017-06-18 22:26:34.279530 [DEBUG] switch_core_media.c:4500 Audio Codec
>> Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match
>> 2017-06-18 22:26:34.279530 [DEBUG] switch_core_media.c:3057 Set Codec
>> sofia/external/sipp at 185.35.228.51:5060 PCMU/8000 20 ms 160 samples 64000
>> bits 1 channels
>> 2017-06-18 22:26:34.279530 [DEBUG] switch_core_codec.c:111 sofia/
>> external/sipp at 185.35.228.51:5060 Original read codec set to PCMU:0
>> 2017-06-18 22:26:34.279530 [DEBUG] switch_core_media.c:4770 No 2833 in
>> SDP.  Disable 2833 dtmf and switch to INFO
>> 2017-06-18 22:26:34.279530 [DEBUG] switch_core_media.c:6874 AUDIO RTP [
>> sofia/external/sipp at 185.35.228.51:5060] 185.35.228.40 port 23728 ->
>> 185.35.228.51 port 6000 codec: 0 ms: 20
>> 2017-06-18 22:26:34.279530 [DEBUG] switch_rtp.c:4108 Starting timer
>> [soft] 160 bytes per 20ms
>> 2017-06-18 22:26:34.279530 [DEBUG] switch_core_media.c:7205 sofia
>> /external/sipp at 185.35.228.51:5060 Set rtp dtmf delay to 40
>> 2017-06-18 22:26:34.279530 [NOTICE] sofia_media.c:92 Pre-Answer
>> sofia/external/sipp at 185.35.228.51:5060!
>> 2017-06-18 22:26:34.279530 [DEBUG] switch_channel.c:3474 (
>> sofia/external/sipp at 185.35.228.51:5060) Callstate Change RINGING -> EARLY
>> 2017-06-18 22:26:34.279530 [DEBUG] switch_core_media.c:6857 Audio params
>> are unchanged for sofia/external/sipp at 185.35.228.51:5060.
>> 2017-06-18 22:26:34.279530 [DEBUG] mod_sofia.c:850 Local SDP
>> sofia/external/sipp at 185.35.228.51:5060:
>> v=0
>> o=FreeSWITCH 1497797466 1497797467 IN IP4 185.35.228.40
>> s=FreeSWITCH
>> c=IN IP4 185.35.228.40
>> t=0 0
>> m=audio 23728 RTP/AVP 0
>> a=rtpmap:0 PCMU/8000
>> a=ptime:20
>> a=sendrecv
>>
>> 2017-06-18 22:26:34.279530 [NOTICE] switch_ivr_originate.c:3647 Channel [
>> sofia/external/sipp at 185.35.228.51:5060] has been answered
>> EXECUTE sofia/external/sipp at 185.35.228.51:5060 sched_hangup(+
>> alloted_timeout)
>> 2017-06-18 22:26:34.279530 [DEBUG] sofia.c:7048 Channel
>> sofia/external/sipp at 185.35.228.51:5060 entering state [completed][200]
>> 2017-06-18 22:26:34.279530 [DEBUG] sofia.c:7048 Channel
>> sofia/external/sipp at 185.35.228.51:5060 entering state [ready][200]
>> 2017-06-18 22:26:34.279530 [NOTICE] mod_dptools.c:1188 Hangup
>> sofia/external/sipp at 185.35.228.51:5060 [CS_EXECUTE] [NORMAL_CLEARING]
>> 2017-06-18 22:26:34.279530 [DEBUG] switch_core_session.c:2814 sof
>> ia/external/sipp at 185.35.228.51:5060 skip receive message
>> [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
>> 2017-06-18 22:26:34.279530 [DEBUG] switch_channel.c:3773 (
>> sofia/external/sipp at 185.35.228.51:5060) Callstate Change EARLY -> ACTIVE
>> 2017-06-18 22:26:34.279530 [DEBUG] switch_ivr_originate.c:3647 so
>> fia/external/sipp at 185.35.228.51:5060 skip receive message [ANSWER_EVENT]
>> (channel is hungup already)
>> 2017-06-18 22:26:34.319525 [DEBUG] switch_ivr_originate.c:3848 Originate
>> Resulted in Error Cause: 487 [ORIGINATOR_CANCEL]
>> 2017-06-18 22:26:34.319525 [NOTICE] switch_ivr_originate.c:3938 Hangup
>> sofia/external/441554555666 at 185.35.229.30:5060 [CS_CONSUME_MEDIA]
>> [ORIGINATOR_CANCEL]
>> 2017-06-18 22:26:34.319525 [DEBUG] switch_core_state_machine.c:584 (
>> sofia/external/441554555666 at 185.35.229.30:5060) Running State Change
>> CS_HANGUP (Cur 2 Tot 68)
>> 2017-06-18 22:26:34.319525 [DEBUG] switch_core_state_machine.c:850 (
>> sofia/external/441554555666 at 185.35.229.30:5060) Callstate Change ACTIVE
>> -> HANGUP
>> 2017-06-18 22:26:34.319525 [DEBUG] switch_core_state_machine.c:852 (
>> sofia/external/441554555666 at 185.35.229.30:5060) State HANGUP
>> 2017-06-18 22:26:34.319525 [DEBUG] mod_sofia.c:438 Channel
>> sofia/external/441554555666 at 185.35.229.30:5060 hanging up, cause:
>> ORIGINATOR_CANCEL
>> 2017-06-18 22:26:34.319525 [INFO] mod_dptools.c:3418 Originate Failed.
>> Cause: ORIGINATOR_CANCEL
>> 2017-06-18 22:26:34.319525 [DEBUG] mod_sofia.c:491 Sending BYE to
>> sofia/external/441554555666 at 185.35.229.30:5060
>> 2017-06-18 22:26:34.319525 [DEBUG] switch_core_state_machine.c:60
>> sofia/external/441554555666 at 185.35.229.30:5060 Standard HANGUP, cause:
>> ORIGINATOR_CANCEL
>> 2017-06-18 22:26:34.319525 [DEBUG] switch_core_state_machine.c:852 (
>> sofia/external/441554555666 at 185.35.229.30:5060) State HANGUP going to
>> sleep
>> 2017-06-18 22:26:34.319525 [DEBUG] switch_core_state_machine.c:619 (
>> sofia/external/441554555666 at 185.35.229.30:5060) State Change CS_HANGUP
>> -> CS_REPORTING
>> 2017-06-18 22:26:34.319525 [DEBUG] switch_core_session.c:2814 sof
>> ia/external/sipp at 185.35.228.51:5060 skip receive message
>> [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
>> 2017-06-18 22:26:34.319525 [DEBUG] switch_core_state_machine.c:584 (
>> sofia/external/441554555666 at 185.35.229.30:5060) Running State Change
>> CS_REPORTING (Cur 2 Tot 68)
>> 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:650 (
>> sofia/external/sipp at 185.35.228.51:5060) State EXECUTE going to sleep
>> 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:584 (
>> sofia/external/sipp at 185.35.228.51:5060) Running State Change CS_HANGUP
>> (Cur 2 Tot 68)
>> 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:938 (
>> sofia/external/441554555666 at 185.35.229.30:5060) State REPORTING
>> 2017-06-18 22:26:34.339569 [ERR] mod_xml_radius.c:930 Didn't match:
>> 185.35.229.30 == ^185\.35\.229\.30
>> 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:850 (
>> sofia/external/sipp at 185.35.228.51:5060) Callstate Change ACTIVE -> HANGUP
>> 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:174
>> sofia/external/441554555666 at 185.35.229.30:5060 Standard REPORTING,
>> cause: ORIGINATOR_CANCEL
>> 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:938 (
>> sofia/external/441554555666 at 185.35.229.30:5060) State REPORTING going to
>> sleep
>> 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:852 (
>> sofia/external/sipp at 185.35.228.51:5060) State HANGUP
>> 2017-06-18 22:26:34.339569 [DEBUG] mod_sofia.c:438 Channel
>> sofia/external/sipp at 185.35.228.51:5060 hanging up, cause: NORMAL_CLEARING
>> 2017-06-18 22:26:34.339569 [DEBUG] mod_sofia.c:491 Sending BYE to
>> sofia/external/sipp at 185.35.228.51:5060
>> 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:60
>> sofia/external/sipp at 185.35.228.51:5060 Standard HANGUP, cause:
>> NORMAL_CLEARING
>> 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:852 (
>> sofia/external/sipp at 185.35.228.51:5060) State HANGUP going to sleep
>> 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:619 (
>> sofia/external/sipp at 185.35.228.51:5060) State Change CS_HANGUP ->
>> CS_REPORTING
>> 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:584 (
>> sofia/external/sipp at 185.35.228.51:5060) Running State Change
>> CS_REPORTING (Cur 2 Tot 68)
>> 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:938 (
>> sofia/external/sipp at 185.35.228.51:5060) State REPORTING
>> 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:610 (
>> sofia/external/441554555666 at 185.35.229.30:5060) State Change
>> CS_REPORTING -> CS_DESTROY
>> 2017-06-18 22:26:34.339569 [DEBUG] switch_core_session.c:1664 Session 68 (
>> sofia/external/441554555666 at 185.35.229.30:5060) Locked, Waiting on
>> external entities
>> 2017-06-18 22:26:34.339569 [ERR] mod_xml_radius.c:933 Result of true
>> match: 185.35.228.40 == ^185\.35\.229\.30
>> 2017-06-18 22:26:34.339569 [NOTICE] switch_core_session.c:1682 Session 68
>> (sofia/external/441554555666 at 185.35.229.30:5060) Ended
>> 2017-06-18 22:26:34.339569 [NOTICE] switch_core_session.c:1686 Close
>> Channel sofia/external/441554555666 at 185.35.229.30:5060 [CS_DESTROY]
>> 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:741 (
>> sofia/external/441554555666 at 185.35.229.30:5060) Running State Change
>> CS_DESTROY (Cur 1 Tot 68)
>> 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:751 (
>> sofia/external/441554555666 at 185.35.229.30:5060) State DESTROY
>> 2017-06-18 22:26:34.339569 [DEBUG] mod_sofia.c:343 sofia/external
>> /441554555666 at 185.35.229.30:5060 SOFIA DESTROY
>> 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:181
>> sofia/external/441554555666 at 185.35.229.30:5060 Standard DESTROY
>> 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:751 (
>> sofia/external/441554555666 at 185.35.229.30:5060) State DESTROY going to
>> sleep
>> 2017-06-18 22:26:34.339569 [INFO] mod_xml_radius.c:1044 mod_xml_radius:
>> Accounting Stop success
>> 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:174
>> sofia/external/sipp at 185.35.228.51:5060 Standard REPORTING, cause:
>> NORMAL_CLEARING
>> 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:938 (
>> sofia/external/sipp at 185.35.228.51:5060) State REPORTING going to sleep
>> 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:610 (
>> sofia/external/sipp at 185.35.228.51:5060) State Change CS_REPORTING ->
>> CS_DESTROY
>> 2017-06-18 22:26:34.339569 [DEBUG] switch_core_session.c:1664 Session 67 (
>> sofia/external/sipp at 185.35.228.51:5060) Locked, Waiting on external
>> entiti
>> 2017-06-18 22:26:34.339569 [NOTICE] switch_core_session.c:1682 Session 67
>> (sofia/external/sipp at 185.35.228.51:5060) Ended
>> 2017-06-18 22:26:34.339569 [NOTICE] switch_core_session.c:1686 Close
>> Channel sofia/external/sipp at 185.35.228.51:5060 [CS_DESTROY]
>> 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:741 (
>> sofia/external/sipp at 185.35.228.51:5060) Running State Change CS_DESTROY
>> (Cur 0 Tot 68)
>> 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:751 (
>> sofia/external/sipp at 185.35.228.51:5060) State DESTROY
>> 2017-06-18 22:26:34.339569 [DEBUG] mod_sofia.c:343 sofia/external
>> /sipp at 185.35.228.51:5060 SOFIA DESTROY
>> 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:181
>> sofia/external/sipp at 185.35.228.51:5060 Standard DESTROY
>> 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:751 (
>> sofia/external/sipp at 185.35.228.51:5060) State DESTROY going to sleep
>>
>>
>> U 185.35.228.51:5060 -> 185.35.228.40:5080
>> INVITE sip:441554555666 at 185.35.228.40:5080 SIP/2.0.
>> Via: SIP/2.0/UDP 185.35.228.51:5060;branch=z9hG4bK-27036-1-0.
>> From: sipp <sip:sipp at 185.35.228.51:5060>;tag=27036SIPpTag001.
>> To: 441554555666 <sip:441554555666 at 185.35.228.40:5080>.
>> Call-ID: 1-27036 at 185.35.228.51.
>> CSeq: 1 INVITE.
>> Contact: sip:sipp at 185.35.228.51:5060.
>> Max-Forwards: 70.
>> Subject: Performance Test.
>> Content-Type: application/sdp.
>> Content-Length:   137.
>> .
>> v=0.
>> o=user1 53655765 2353687637 IN IP4 185.35.228.51.
>> s=-.
>> c=IN IP4 185.35.228.51.
>> t=0 0.
>> m=audio 6000 RTP/AVP 0.
>> a=rtpmap:0 PCMU/8000.
>>
>> #
>> U 185.35.228.40:5080 -> 185.35.228.51:5060
>> SIP/2.0 100 Trying.
>> Via: SIP/2.0/UDP 185.35.228.51:5060;branch=z9hG4bK-27036-1-0.
>> From: sipp <sip:sipp at 185.35.228.51:5060>;tag=27036SIPpTag001.
>> To: 441554555666 <sip:441554555666 at 185.35.228.40:5080>.
>> Call-ID: 1-27036 at 185.35.228.51.
>> CSeq: 1 INVITE.
>> User-Agent: FreeSWITCH-mod_sofia/1.6.18+git~20170612T211449Z~6e79667c0a~
>> 64bit.
>> Content-Length: 0.
>> .
>>
>> #
>> U 185.35.228.40:5080 -> 185.35.228.51:5060
>> SIP/2.0 180 Ringing.
>> Via: SIP/2.0/UDP 185.35.228.51:5060;branch=z9hG4bK-27036-1-0.
>> From: sipp <sip:sipp at 185.35.228.51:5060>;tag=27036SIPpTag001.
>> To: 441554555666 <sip:441554555666 at 185.35.228.40:5080>;tag=91vp8601aS4Qp.
>> Call-ID: 1-27036 at 185.35.228.51.
>> CSeq: 1 INVITE.
>> Contact: <sip:441554555666 at 185.35.228.40:5080;transport=udp>.
>> User-Agent: FreeSWITCH-mod_sofia/1.6.18+git~20170612T211449Z~6e79667c0a~
>> 64bit.
>> Accept: application/sdp.
>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
>> REGISTER, REFER, NOTIFY.
>> Supported: timer, path, replaces.
>> Allow-Events: talk, hold, conference, refer.
>> Content-Length: 0.
>> Remote-Party-ID: "Outbound Call" <sip:441554555666 at 185.35.228.40
>> >;party=calling;privacy=off;screen=no.
>> .
>>
>> #
>> U 185.35.228.40:5080 -> 185.35.228.51:5060
>> SIP/2.0 200 OK.
>> Via: SIP/2.0/UDP 185.35.228.51:5060;branch=z9hG4bK-27036-1-0.
>> From: sipp <sip:sipp at 185.35.228.51:5060>;tag=27036SIPpTag001.
>> To: 441554555666 <sip:441554555666 at 185.35.228.40:5080>;tag=91vp8601aS4Qp.
>> Call-ID: 1-27036 at 185.35.228.51.
>> CSeq: 1 INVITE.
>> Contact: <sip:441554555666 at 185.35.228.40:5080;transport=udp>.
>> User-Agent: FreeSWITCH-mod_sofia/1.6.18+git~20170612T211449Z~6e79667c0a~
>> 64bit.
>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
>> REGISTER, REFER, NOTIFY.
>> Supported: timer, path, replaces.
>> Allow-Events: talk, hold, conference, refer.
>> Content-Type: application/sdp.
>> Content-Disposition: session.
>> Content-Length: 166.
>> Remote-Party-ID: "Outbound Call" <sip:441554555666 at 185.35.228.40
>> >;party=calling;privacy=off;screen=no.
>> .
>> v=0.
>> o=FreeSWITCH 1497797499 1497797500 IN IP4 185.35.228.40.
>> s=FreeSWITCH.
>> c=IN IP4 185.35.228.40.
>> t=0 0.
>> m=audio 25252 RTP/AVP 0.
>> a=rtpmap:0 PCMU/8000.
>> a=ptime:20.
>>
>> #
>> U 185.35.228.51:5060 -> 185.35.228.40:5080
>> ACK sip:441554555666 at 185.35.228.40:5080 SIP/2.0.
>> Via: SIP/2.0/UDP 185.35.228.51:5060;branch=z9hG4bK-27036-1-5.
>> From: sipp <sip:sipp at 185.35.228.51:5060>;tag=27036SIPpTag001.
>> To: 441554555666 <sip:441554555666 at 185.35.228.40:5080>;tag=91vp8601aS4Qp.
>> Call-ID: 1-27036 at 185.35.228.51.
>> CSeq: 1 ACK.
>> Contact: sip:sipp at 185.35.228.51:5060.
>> Max-Forwards: 70.
>> Subject: Performance Test.
>> Content-Length: 0.
>> .
>>
>> #
>> U 185.35.228.40:5080 -> 185.35.228.51:5060
>> BYE sip:sipp at 185.35.228.51:5060 SIP/2.0.
>> Via: SIP/2.0/UDP 185.35.228.40:5080;rport;branch=z9hG4bKUBe0HvDjHF1XN.
>> Max-Forwards: 70.
>> From: 441554555666 <sip:441554555666 at 185.35.228.40:5080
>> >;tag=91vp8601aS4Qp.
>> To: sipp <sip:sipp at 185.35.228.51:5060>;tag=27036SIPpTag001.
>> Call-ID: 1-27036 at 185.35.228.51.
>> CSeq: 108575311 BYE.
>> User-Agent: FreeSWITCH-mod_sofia/1.6.18+git~20170612T211449Z~6e79667c0a~
>> 64bit.
>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
>> REGISTER, REFER, NOTIFY.
>> Supported: timer, path, replaces.
>> Reason: Q.850;cause=16;text="NORMAL_CLEARING".
>> Content-Length: 0.
>> .
>>
>> #
>> U 185.35.228.51:5060 -> 185.35.228.40:5080
>> SIP/2.0 200 OK.
>> Via: SIP/2.0/UDP 185.35.228.40:5080;rport;branch=z9hG4bKUBe0HvDjHF1XN.
>> From: 441554555666 <sip:441554555666 at 185.35.228.40:5080
>> >;tag=91vp8601aS4Qp.
>> To: sipp <sip:sipp at 185.35.228.51:5060>;tag=27036SIPpTag001.
>> Call-ID: 1-27036 at 185.35.228.51.
>> CSeq: 108575311 BYE.
>> Contact: <sip:185.35.228.51:5060;transport=UDP>.
>> Content-Length: 0.
>>
>>
>>
>> NGREP of SIP messages from FS to terminator
>>
>>
>> U 185.35.228.40:5080 -> 185.35.228.48:5060
>> INVITE sip:441554555666 at 185.35.228.48:5060 SIP/2.0.
>> Via: SIP/2.0/UDP 185.35.228.40:5080;rport;branch=z9hG4bKXX0HNjFSB1D3c.
>> Max-Forwards: 69.
>> From: "sipp" <sip:sipp at 185.35.228.51>;tag=eFUjHeNKv5KNg.
>> To: <sip:441554555666 at 185.35.229.30:5060>.
>> Call-ID: c755e7c8-cf13-1235-25a9-363165383663.
>> CSeq: 108575420 INVITE.
>> Contact: <sip:mod_sofia at 185.35.228.40:5080>.
>> User-Agent: FreeSWITCH-mod_sofia/1.6.18+git~20170612T211449Z~6e79667c0a~
>> 64bit.
>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
>> REGISTER, REFER, NOTIFY.
>> Supported: timer, path, replaces.
>> Allow-Events: talk, hold, conference, refer.
>> Content-Type: application/sdp.
>> Content-Disposition: session.
>> Content-Length: 222.
>> X-FS-Support: update_display,send_info.
>> Remote-Party-ID: "sipp" <sip:sipp at 185.35.228.40>;party
>> =calling;screen=yes;privacy=off.
>> .
>> v=0.
>> o=FreeSWITCH 1497801741 1497801743 IN IP4 185.35.228.40.
>> s=FreeSWITCH.
>> c=IN IP4 185.35.228.40.
>> t=0 0.
>> m=audio 21228 RTP/AVP 0 101.
>> a=rtpmap:0 PCMU/8000.
>> a=rtpmap:101 telephone-event/8000.
>> a=fmtp:101 0-16.
>> a=ptime:20.
>>
>> #
>> U 185.35.228.48:5060 -> 185.35.228.40:5080
>> SIP/2.0 180 Ringing.
>> Via: SIP/2.0/UDP 185.35.228.40:5080;rport;branch=z9hG4bKXX0HNjFSB1D3c.
>> From: "sipp" <sip:sipp at 185.35.228.51>;tag=eFUjHeNKv5KNg.
>> To: <sip:441554555666 at 185.35.229.30:5060>;tag=31480SIPpTag018.
>> Call-ID: c755e7c8-cf13-1235-25a9-363165383663.
>> CSeq: 108575420 INVITE.
>> Contact: <sip:185.35.228.48:5060;transport=UDP>.
>> Content-Length: 0.
>> .
>>
>> #
>> U 185.35.228.48:5060 -> 185.35.228.40:5080
>> SIP/2.0 200 OK.
>> Via: SIP/2.0/UDP 185.35.228.40:5080;rport;branch=z9hG4bKXX0HNjFSB1D3c.
>> From: "sipp" <sip:sipp at 185.35.228.51>;tag=eFUjHeNKv5KNg.
>> To: <sip:441554555666 at 185.35.229.30:5060>;tag=31480SIPpTag018.
>> Call-ID: c755e7c8-cf13-1235-25a9-363165383663.
>> CSeq: 108575420 INVITE.
>> Contact: <sip:185.35.228.48:5060;transport=UDP>.
>> Content-Type: application/sdp.
>> Content-Length:   137.
>> .
>> v=0.
>> o=user1 53655765 2353687637 IN IP4 185.35.228.48.
>> s=-.
>> c=IN IP4 185.35.228.48.
>> t=0 0.
>> m=audio 6000 RTP/AVP 0.
>> a=rtpmap:0 PCMU/8000.
>>
>> #
>> U 185.35.228.40:5080 -> 185.35.228.48:5060
>> ACK sip:185.35.228.48:5060;transport=UDP SIP/2.0.
>> Via: SIP/2.0/UDP 185.35.228.40:5080;rport;branch=z9hG4bKy6SaQD0v893Nr.
>> Max-Forwards: 70.
>> From: "sipp" <sip:sipp at 185.35.228.51>;tag=eFUjHeNKv5KNg.
>> To: <sip:441554555666 at 185.35.229.30:5060>;tag=31480SIPpTag018.
>> Call-ID: c755e7c8-cf13-1235-25a9-363165383663.
>> CSeq: 108575420 ACK.
>> Contact: <sip:mod_sofia at 185.35.228.40:5080>.
>> Content-Length: 0.
>> .
>>
>> #
>> U 185.35.228.40:5080 -> 185.35.228.48:5060
>> BYE sip:185.35.228.48:5060;transport=UDP SIP/2.0.
>> Via: SIP/2.0/UDP 185.35.228.40:5080;rport;branch=z9hG4bKZFK3r8g05jt8K.
>> Max-Forwards: 70.
>> From: "sipp" <sip:sipp at 185.35.228.51>;tag=eFUjHeNKv5KNg.
>> To: <sip:441554555666 at 185.35.229.30:5060>;tag=31480SIPpTag018.
>> Call-ID: c755e7c8-cf13-1235-25a9-363165383663.
>> CSeq: 108575421 BYE.
>> User-Agent: FreeSWITCH-mod_sofia/1.6.18+git~20170612T211449Z~6e79667c0a~
>> 64bit.
>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
>> REGISTER, REFER, NOTIFY.
>> Supported: timer, path, replaces.
>> Reason: SIP;cause=487;text="ORIGINATOR_CANCEL".
>> Content-Length: 0.
>> .
>>
>> #
>> U 185.35.228.48:5060 -> 185.35.228.40:5080
>> SIP/2.0 200 OK.
>> Via: SIP/2.0/UDP 185.35.228.40:5080;rport;branch=z9hG4bKZFK3r8g05jt8K.
>> From: "sipp" <sip:sipp at 185.35.228.51>;tag=eFUjHeNKv5KNg.
>> To: <sip:441554555666 at 185.35.229.30:5060>;tag=31480SIPpTag018.
>> Call-ID: c755e7c8-cf13-1235-25a9-363165383663.
>> CSeq: 108575421 BYE.
>> Contact: <sip:185.35.228.48:5060;transport=UDP>.
>> Content-Length: 0.
>> .
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.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
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170619/3f081f25/attachment-0001.html>

From joelists at tm.net.uk  Mon Jun 19 07:42:55 2017
From: joelists at tm.net.uk (Joseph Waite)
Date: Mon, 19 Jun 2017 08:42:55 +0100
Subject: [Freeswitch-users] Call Dropping
In-Reply-To: <CALFFxt0BWRGT+gWqWL27M-Y1WDYuLUMj+zyic72LvBNDZWjCuw@mail.gmail.com>
References: <222F1F16-373C-484A-B290-200CB13F2CAD@tm.net.uk>
 <CAPtU-Up9YWb2j6u2wZ6KS6WEZST2++CibUqNeXJOwnPY7FfHaA@mail.gmail.com>
 <CALFFxt0BWRGT+gWqWL27M-Y1WDYuLUMj+zyic72LvBNDZWjCuw@mail.gmail.com>
Message-ID: <4CE385F8-7E96-4565-AB5D-94C8459EDAD7@tm.net.uk>

Ok, I had set auth_calls to false in my sip profile, however because it was not sending the Radius auth it was not getting a value for h323-credit-time.

Trouble is if I set auth_calls to true then it sends a “SIP/2.0 407 Proxy Authentication Required” in response to an INVITE which I don’t want on my IP authenticated calls port. How do I get FS to not send this but still send the Radius Auth packet based simply on the IP address?

Regards
> On 19 Jun 2017, at 00:33, Ahmed Sboor <ahmed at netelsat.net> wrote:
> 
> if on VCS , Rate table is set and balance or credit limit is also positive and rate exist in rate table , then h323-credit-limit is also set.
> you should also post mod radius debug logs.
> 
> 
> On Mon, Jun 19, 2017 at 4:08 AM, Colin Morelli <colin.morelli at gmail.com <mailto:colin.morelli at gmail.com>> wrote:
> You've got an execute_on_answer of sched_hangup(+${h323-credit-time} alloted_timeout)
> 
> Immediately after your call is answered:
> 2017-06-18 22:26:34.279530 [NOTICE] switch_ivr_originate.c:3647 Channel [sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060] has been answered
> EXECUTE sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 sched_hangup(+ alloted_timeout)
> 
> It would seem that h323-credit-time is not being set, which is causing sched_hangup to immediately hangup the call on answer.
> 
> On Sun, Jun 18, 2017 at 6:52 PM, Joseph Waite <joelists at tm.net.uk <mailto:joelists at tm.net.uk>> wrote:
> Hi Guys
> 
> Using FreeSwitch with Radius linked to JeraSoft VCS billing system. 
> 
> I am sending a Call from a SIPP originator, through the FreeSwitch box and back out to another SIPP terminator scenario.
> The call goes through ok, everything happens as it should, however the call immediately drops, I have done egrep’s of both sides of the call and the BYE is defiantly coming from Freeswitch for some reason but I cannot work out why. Anyone any ideas?
> I am attaching the FreeSwitch logs plus the egrep’s
> If I register zipper on my laptop to FS and make a call works fine.
> 
> 2017-06-18 22:26:34.199519 [NOTICE] switch_channel.c:1104 New Channel sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 [15021010-8f64-439f-8dbb-1afe090c44a5]
> 2017-06-18 22:26:34.199519 [DEBUG] switch_core_state_machine.c:584 (sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060) Running State Change CS_NEW (Cur 1 Tot 67)
> 2017-06-18 22:26:34.199519 [DEBUG] sofia.c:9837 sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 receiving invite from 185.35.228.51:5060 <http://185.35.228.51:5060/> version: 1.6.18 git 6e79667 2017-06-12 21:14:49Z 64bit
> 2017-06-18 22:26:34.219492 [DEBUG] sofia.c:7048 Channel sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 entering state [received][100]
> 2017-06-18 22:26:34.219492 [DEBUG] sofia.c:7058 Remote SDP:
> v=0
> o=user1 53655765 2353687637 IN IP4 185.35.228.51
> s=-
> c=IN IP4 185.35.228.51
> t=0 0
> m=audio 6000 RTP/AVP 0
> a=rtpmap:0 PCMU/8000
> 
> 2017-06-18 22:26:34.219492 [DEBUG] sofia.c:7450 (sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060) State Change CS_NEW -> CS_INIT
> 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:603 (sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060) State NEW
> 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:584 (sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060) Running State Change CS_INIT (Cur 1 Tot 67)
> 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:627 (sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060) State INIT
> 2017-06-18 22:26:34.219492 [DEBUG] mod_sofia.c:90 sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 SOFIA INIT
> 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:40 sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 Standard INIT
> 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:48 (sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060) State Change CS_INIT -> CS_ROUTING
> 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:627 (sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060) State INIT going to sleep
> 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:584 (sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060) Running State Change CS_ROUTING (Cur 1 Tot 67)
> 2017-06-18 22:26:34.219492 [DEBUG] switch_channel.c:2249 (sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060) Callstate Change DOWN -> RINGING
> 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:643 (sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060) State ROUTING
> 2017-06-18 22:26:34.219492 [DEBUG] mod_sofia.c:143 sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 SOFIA ROUTING
> 2017-06-18 22:26:34.219492 [ERR] mod_xml_radius.c:933 Result of true match: 185.35.228.40 == ^185\.35\.229\.30 
> 2017-06-18 22:26:34.219492 [INFO] mod_xml_radius.c:986 mod_xml_radius:  Accounting Start success
> 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:236 sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 Standard ROUTING
> 2017-06-18 22:26:34.219492 [INFO] mod_dialplan_xml.c:637 Processing sipp <sipp>->441554555666 in context public
> Dialplan: sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 parsing [public->unloop] continue=false
> Dialplan: sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false
> Dialplan: sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false
> Dialplan: sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 parsing [public->outside_call] continue=true
> Dialplan: sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 Absolute Condition [outside_call]
> Dialplan: sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 Action set(outside_call=true) 
> Dialplan: sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 Action export(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) 
> Dialplan: sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 parsing [public->call_debug] continue=true
> Dialplan: sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 Regex (FAIL) [call_debug] ${call_debug}(false) =~ /^true$/ break=never
> Dialplan: sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 parsing [public->rejections] continue=false
> Dialplan: sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 Regex (FAIL) [rejections] ${radius_auth_result}() =~ /2/ break=on-false
> Dialplan: sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 parsing [public->timedouts] continue=false
> Dialplan: sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 Regex (FAIL) [timedouts] ${radius_auth_result}() =~ /1/ break=on-false
> Dialplan: sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 parsing [public->JeraSoft VCS Routing] continue=false
> Dialplan: sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 Regex (PASS) [JeraSoft VCS Routing] destination_number(441554555666) =~ /^(.+)$/ break=on-false
> Dialplan: sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 Action export(nolocal:h323-call-origin=originate) 
> Dialplan: sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 Action set(sip_h_X-accountcode=${accountcode}) 
> Dialplan: sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 Action set(call_direction=outbound) 
> Dialplan: sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 Action set(hangup_after_bridge=true) 
> Dialplan: sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 Action set(continue_on_fail=true) 
> Dialplan: sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 Action set(inherit_codec=true) 
> Dialplan: sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 Action set(call_timeout=20) 
> Dialplan: sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 Action set(fail_on_single_reject=USER_BUSY) 
> Dialplan: sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 Action set(origination_caller_id_name=${sip_req_user}) 
> Dialplan: sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 Action set(origination_caller_id_number=${sip_from_user}) 
> Dialplan: sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 Action set(execute_on_answer=sched_hangup +${h323-credit-time} alloted_timeout) 
> Dialplan: sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 Action bridge({sip_invite_from_uri=sip:${sip_from_user}@${sip_network_ip}}sofia/external/${destination_number}@185.35.229.30:5060 <http://destination_number%7D at 185.35.229.30:5060/>) 
> Dialplan: sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 Action hangup(${bridge_hangup_cause}) 
> 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:286 (sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060) State Change CS_ROUTING -> CS_EXECUTE
> 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:643 (sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060) State ROUTING going to sleep
> 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:584 (sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060) Running State Change CS_EXECUTE (Cur 1 Tot 67)
> 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:650 (sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060) State EXECUTE
> 2017-06-18 22:26:34.219492 [DEBUG] mod_sofia.c:198 sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 SOFIA EXECUTE
> 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:328 sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 Standard EXECUTE
> EXECUTE sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 set(outside_call=true)
> 2017-06-18 22:26:34.219492 [DEBUG] mod_dptools.c:1530 SET sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 [outside_call]=[true]
> EXECUTE sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 export(RFC2822_DATE=Sun, 18 Jun 2017 22:26:34 +0100)
> 2017-06-18 22:26:34.219492 [DEBUG] switch_channel.c:1296 EXPORT (export_vars) [RFC2822_DATE]=[Sun, 18 Jun 2017 22:26:34 +0100]
> EXECUTE sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 export(nolocal:h323-call-origin=originate)
> 2017-06-18 22:26:34.219492 [DEBUG] switch_channel.c:1296 EXPORT (export_vars) (REMOTE ONLY) [h323-call-origin]=[originate]
> EXECUTE sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 set(sip_h_X-accountcode=)
> 2017-06-18 22:26:34.219492 [DEBUG] mod_dptools.c:1530 SET sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 [sip_h_X-accountcode]=[UNDEF]
> EXECUTE sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 set(call_direction=outbound)
> 2017-06-18 22:26:34.219492 [DEBUG] mod_dptools.c:1530 SET sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 [call_direction]=[outbound]
> EXECUTE sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 set(hangup_after_bridge=true)
> 2017-06-18 22:26:34.219492 [DEBUG] mod_dptools.c:1530 SET sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 [hangup_after_bridge]=[true]
> EXECUTE sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 set(continue_on_fail=true)
> 2017-06-18 22:26:34.219492 [DEBUG] mod_dptools.c:1530 SET sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 [continue_on_fail]=[true]
> EXECUTE sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 set(inherit_codec=true)
> 2017-06-18 22:26:34.219492 [DEBUG] mod_dptools.c:1530 SET sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 [inherit_codec]=[true]
> EXECUTE sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 set(call_timeout=20)
> 2017-06-18 22:26:34.219492 [DEBUG] mod_dptools.c:1530 SET sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 [call_timeout]=[20]
> EXECUTE sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 set(fail_on_single_reject=USER_BUSY)
> 2017-06-18 22:26:34.219492 [DEBUG] mod_dptools.c:1530 SET sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 [fail_on_single_reject]=[USER_BUSY]
> EXECUTE sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 set(origination_caller_id_name=441554555666)
> 2017-06-18 22:26:34.219492 [DEBUG] mod_dptools.c:1530 SET sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 [origination_caller_id_name]=[441554555666]
> EXECUTE sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 set(origination_caller_id_number=sipp)
> 2017-06-18 22:26:34.219492 [DEBUG] mod_dptools.c:1530 SET sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 [origination_caller_id_number]=[sipp]
> EXECUTE sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 set(execute_on_answer=sched_hangup + alloted_timeout)
> 2017-06-18 22:26:34.219492 [DEBUG] mod_dptools.c:1530 SET sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 [execute_on_answer]=[sched_hangup + alloted_timeout]
> EXECUTE sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 bridge({sip_invite_from_uri=sip:sipp at 185.35.228.51 <mailto:sip%3Asipp at 185.35.228.51>}sofia/external/441554555666 at 185.35.229.30 <mailto:sofia/external/441554555666 at 185.35.229.30>:5060)
> 2017-06-18 22:26:34.219492 [DEBUG] switch_channel.c:1250 sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 EXPORTING[export_vars] [RFC2822_DATE]=[Sun, 18 Jun 2017 22:26:34 +0100] to event
> 2017-06-18 22:26:34.219492 [DEBUG] switch_channel.c:1250 sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 EXPORTING[export_vars] [h323-call-origin]=[originate] to event
> 2017-06-18 22:26:34.219492 [DEBUG] switch_ivr_originate.c:2142 Parsing global variables
> 2017-06-18 22:26:34.219492 [NOTICE] switch_channel.c:1104 New Channel sofia/external/441554555666 at 185.35.229.30 <mailto:sofia/external/441554555666 at 185.35.229.30>:5060 [96c1a021-5195-41ce-b903-08b98816d70d]
> 2017-06-18 22:26:34.219492 [DEBUG] mod_sofia.c:4819 (sofia/external/441554555666 at 185.35.229.30 <mailto:sofia/external/441554555666 at 185.35.229.30>:5060) State Change CS_NEW -> CS_INIT
> 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:584 (sofia/external/441554555666 at 185.35.229.30 <mailto:sofia/external/441554555666 at 185.35.229.30>:5060) Running State Change CS_INIT (Cur 2 Tot 68)
> 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:627 (sofia/external/441554555666 at 185.35.229.30 <mailto:sofia/external/441554555666 at 185.35.229.30>:5060) State INIT
> 2017-06-18 22:26:34.219492 [DEBUG] mod_sofia.c:90 sofia/external/441554555666 at 185.35.229.30 <mailto:sofia/external/441554555666 at 185.35.229.30>:5060 SOFIA INIT
> 2017-06-18 22:26:34.219492 [DEBUG] sofia_glue.c:1295 sofia/external/441554555666 at 185.35.229.30 <mailto:sofia/external/441554555666 at 185.35.229.30>:5060 sending invite version: 1.6.18 git 6e79667 2017-06-12 21:14:49Z 64bit
> Local SDP:
> v=0
> o=FreeSWITCH 1497789362 1497789363 IN IP4 185.35.228.40
> s=FreeSWITCH
> c=IN IP4 185.35.228.40
> t=0 0
> m=audio 31832 RTP/AVP 0 101 13
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=rtpmap:13 CN/8000
> a=ptime:20
> a=sendrecv
> 
> 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:40 sofia/external/441554555666 at 185.35.229.30 <mailto:sofia/external/441554555666 at 185.35.229.30>:5060 Standard INIT
> 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:48 (sofia/external/441554555666 at 185.35.229.30 <mailto:sofia/external/441554555666 at 185.35.229.30>:5060) State Change CS_INIT -> CS_ROUTING
> 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:627 (sofia/external/441554555666 at 185.35.229.30 <mailto:sofia/external/441554555666 at 185.35.229.30>:5060) State INIT going to sleep
> 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:584 (sofia/external/441554555666 at 185.35.229.30 <mailto:sofia/external/441554555666 at 185.35.229.30>:5060) Running State Change CS_ROUTING (Cur 2 Tot 68)
> 2017-06-18 22:26:34.219492 [DEBUG] sofia.c:7048 Channel sofia/external/441554555666 at 185.35.229.30 <mailto:sofia/external/441554555666 at 185.35.229.30>:5060 entering state [calling][0]
> 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:643 (sofia/external/441554555666 at 185.35.229.30 <mailto:sofia/external/441554555666 at 185.35.229.30>:5060) State ROUTING
> 2017-06-18 22:26:34.219492 [DEBUG] mod_sofia.c:143 sofia/external/441554555666 at 185.35.229.30 <mailto:sofia/external/441554555666 at 185.35.229.30>:5060 SOFIA ROUTING
> 2017-06-18 22:26:34.219492 [DEBUG] switch_ivr_originate.c:67 (sofia/external/441554555666 at 185.35.229.30 <mailto:sofia/external/441554555666 at 185.35.229.30>:5060) State Change CS_ROUTING -> CS_CONSUME_MEDIA
> 2017-06-18 22:26:34.219492 [ERR] mod_xml_radius.c:930 Didn't match: 185.35.229.30:5060 <http://185.35.229.30:5060/> == ^185\.35\.229\.30 
> 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:643 (sofia/external/441554555666 at 185.35.229.30 <mailto:sofia/external/441554555666 at 185.35.229.30>:5060) State ROUTING going to sleep
> 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:584 (sofia/external/441554555666 at 185.35.229.30 <mailto:sofia/external/441554555666 at 185.35.229.30>:5060) Running State Change CS_CONSUME_MEDIA (Cur 2 Tot 68)
> 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:662 (sofia/external/441554555666 at 185.35.229.30 <mailto:sofia/external/441554555666 at 185.35.229.30>:5060) State CONSUME_MEDIA
> 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:662 (sofia/external/441554555666 at 185.35.229.30 <mailto:sofia/external/441554555666 at 185.35.229.30>:5060) State CONSUME_MEDIA going to sleep
> 2017-06-18 22:26:34.259480 [DEBUG] sofia_glue.c:1295 sofia/external/441554555666 at 185.35.229.30 <mailto:sofia/external/441554555666 at 185.35.229.30>:5060 sending invite version: 1.6.18 git 6e79667 2017-06-12 21:14:49Z 64bit
> Local SDP:
> v=0
> o=FreeSWITCH 1497789362 1497789364 IN IP4 185.35.228.40
> s=FreeSWITCH
> c=IN IP4 185.35.228.40
> t=0 0
> m=audio 31832 RTP/AVP 0 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
> a=sendrecv
> 
> 2017-06-18 22:26:34.259480 [DEBUG] sofia.c:7048 Channel sofia/external/441554555666 at 185.35.229.30 <mailto:sofia/external/441554555666 at 185.35.229.30>:5060 entering state [calling][0]
> 2017-06-18 22:26:34.259480 [DEBUG] sofia.c:7048 Channel sofia/external/441554555666 at 185.35.229.30 <mailto:sofia/external/441554555666 at 185.35.229.30>:5060 entering state [proceeding][180]
> 2017-06-18 22:26:34.259480 [NOTICE] sofia.c:7156 Ring-Ready sofia/external/441554555666 at 185.35.229.30 <mailto:sofia/external/441554555666 at 185.35.229.30>:5060!
> 2017-06-18 22:26:34.259480 [DEBUG] switch_channel.c:3346 (sofia/external/441554555666 at 185.35.229.30 <mailto:sofia/external/441554555666 at 185.35.229.30>:5060) Callstate Change DOWN -> RINGING
> 2017-06-18 22:26:34.259480 [DEBUG] sofia.c:7048 Channel sofia/external/441554555666 at 185.35.229.30 <mailto:sofia/external/441554555666 at 185.35.229.30>:5060 entering state [completing][200]
> 2017-06-18 22:26:34.259480 [DEBUG] sofia.c:7058 Remote SDP:
> v=0
> o=user1 53655765 2353687637 IN IP4 185.35.228.48
> s=-
> c=IN IP4 185.35.228.48
> t=0 0
> m=audio 6000 RTP/AVP 0
> a=rtpmap:0 PCMU/8000
> 
> 2017-06-18 22:26:34.259480 [DEBUG] sofia.c:7048 Channel sofia/external/441554555666 at 185.35.229.30 <mailto:sofia/external/441554555666 at 185.35.229.30>:5060 entering state [ready][200]
> 2017-06-18 22:26:34.259480 [DEBUG] switch_core_media.c:4445 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]
> 2017-06-18 22:26:34.259480 [DEBUG] switch_core_media.c:4500 Audio Codec Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match
> 2017-06-18 22:26:34.259480 [DEBUG] switch_core_media.c:3057 Set Codec sofia/external/441554555666 at 185.35.229.30 <mailto:sofia/external/441554555666 at 185.35.229.30>:5060 PCMU/8000 20 ms 160 samples 64000 bits 1 channels
> 2017-06-18 22:26:34.259480 [DEBUG] switch_core_codec.c:111 sofia/external/441554555666 at 185.35.229.30 <mailto:sofia/external/441554555666 at 185.35.229.30>:5060 Original read codec set to PCMU:0
> 2017-06-18 22:26:34.259480 [DEBUG] switch_core_media.c:4770 No 2833 in SDP.  Disable 2833 dtmf and switch to INFO
> 2017-06-18 22:26:34.259480 [DEBUG] switch_core_media.c:6874 AUDIO RTP [sofia/external/441554555666 at 185.35.229.30 <mailto:sofia/external/441554555666 at 185.35.229.30>:5060] 185.35.228.40 port 31832 -> 185.35.228.48 port 6000 codec: 0 ms: 20
> 2017-06-18 22:26:34.259480 [DEBUG] switch_rtp.c:4108 Starting timer [soft] 160 bytes per 20ms
> 2017-06-18 22:26:34.279530 [DEBUG] switch_core_media.c:7205 sofia/external/441554555666 at 185.35.229.30 <mailto:sofia/external/441554555666 at 185.35.229.30>:5060 Set rtp dtmf delay to 40
> 2017-06-18 22:26:34.279530 [NOTICE] sofia.c:8182 Channel [sofia/external/441554555666 at 185.35.229.30 <mailto:sofia/external/441554555666 at 185.35.229.30>:5060] has been answered
> 2017-06-18 22:26:34.279530 [DEBUG] switch_channel.c:3773 (sofia/external/441554555666 at 185.35.229.30 <mailto:sofia/external/441554555666 at 185.35.229.30>:5060) Callstate Change RINGING -> ACTIVE
> 2017-06-18 22:26:34.279530 [NOTICE] mod_sofia.c:2273 Ring-Ready sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060!
> 2017-06-18 22:26:34.279530 [DEBUG] sofia.c:7048 Channel sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 entering state [early][180]
> 2017-06-18 22:26:34.279530 [NOTICE] switch_ivr_originate.c:525 Ring Ready sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060!
> 2017-06-18 22:26:34.279530 [DEBUG] switch_ivr_originate.c:410 Setting codec string on sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 to PCMU at 8000h@20i
> 2017-06-18 22:26:34.279530 [DEBUG] switch_core_media.c:4445 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]
> 2017-06-18 22:26:34.279530 [DEBUG] switch_core_media.c:4500 Audio Codec Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match
> 2017-06-18 22:26:34.279530 [DEBUG] switch_core_media.c:3057 Set Codec sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 PCMU/8000 20 ms 160 samples 64000 bits 1 channels
> 2017-06-18 22:26:34.279530 [DEBUG] switch_core_codec.c:111 sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 Original read codec set to PCMU:0
> 2017-06-18 22:26:34.279530 [DEBUG] switch_core_media.c:4770 No 2833 in SDP.  Disable 2833 dtmf and switch to INFO
> 2017-06-18 22:26:34.279530 [DEBUG] switch_core_media.c:6874 AUDIO RTP [sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060] 185.35.228.40 port 23728 -> 185.35.228.51 port 6000 codec: 0 ms: 20
> 2017-06-18 22:26:34.279530 [DEBUG] switch_rtp.c:4108 Starting timer [soft] 160 bytes per 20ms
> 2017-06-18 22:26:34.279530 [DEBUG] switch_core_media.c:7205 sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 Set rtp dtmf delay to 40
> 2017-06-18 22:26:34.279530 [NOTICE] sofia_media.c:92 Pre-Answer sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060!
> 2017-06-18 22:26:34.279530 [DEBUG] switch_channel.c:3474 (sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060) Callstate Change RINGING -> EARLY
> 2017-06-18 22:26:34.279530 [DEBUG] switch_core_media.c:6857 Audio params are unchanged for sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060.
> 2017-06-18 22:26:34.279530 [DEBUG] mod_sofia.c:850 Local SDP sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060:
> v=0
> o=FreeSWITCH 1497797466 1497797467 IN IP4 185.35.228.40
> s=FreeSWITCH
> c=IN IP4 185.35.228.40
> t=0 0
> m=audio 23728 RTP/AVP 0
> a=rtpmap:0 PCMU/8000
> a=ptime:20
> a=sendrecv
> 
> 2017-06-18 22:26:34.279530 [NOTICE] switch_ivr_originate.c:3647 Channel [sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060] has been answered
> EXECUTE sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 sched_hangup(+ alloted_timeout)
> 2017-06-18 22:26:34.279530 [DEBUG] sofia.c:7048 Channel sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 entering state [completed][200]
> 2017-06-18 22:26:34.279530 [DEBUG] sofia.c:7048 Channel sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 entering state [ready][200]
> 2017-06-18 22:26:34.279530 [NOTICE] mod_dptools.c:1188 Hangup sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 [CS_EXECUTE] [NORMAL_CLEARING]
> 2017-06-18 22:26:34.279530 [DEBUG] switch_core_session.c:2814 sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
> 2017-06-18 22:26:34.279530 [DEBUG] switch_channel.c:3773 (sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060) Callstate Change EARLY -> ACTIVE
> 2017-06-18 22:26:34.279530 [DEBUG] switch_ivr_originate.c:3647 sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 skip receive message [ANSWER_EVENT] (channel is hungup already)
> 2017-06-18 22:26:34.319525 [DEBUG] switch_ivr_originate.c:3848 Originate Resulted in Error Cause: 487 [ORIGINATOR_CANCEL]
> 2017-06-18 22:26:34.319525 [NOTICE] switch_ivr_originate.c:3938 Hangup sofia/external/441554555666 at 185.35.229.30 <mailto:sofia/external/441554555666 at 185.35.229.30>:5060 [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL]
> 2017-06-18 22:26:34.319525 [DEBUG] switch_core_state_machine.c:584 (sofia/external/441554555666 at 185.35.229.30 <mailto:sofia/external/441554555666 at 185.35.229.30>:5060) Running State Change CS_HANGUP (Cur 2 Tot 68)
> 2017-06-18 22:26:34.319525 [DEBUG] switch_core_state_machine.c:850 (sofia/external/441554555666 at 185.35.229.30 <mailto:sofia/external/441554555666 at 185.35.229.30>:5060) Callstate Change ACTIVE -> HANGUP
> 2017-06-18 22:26:34.319525 [DEBUG] switch_core_state_machine.c:852 (sofia/external/441554555666 at 185.35.229.30 <mailto:sofia/external/441554555666 at 185.35.229.30>:5060) State HANGUP
> 2017-06-18 22:26:34.319525 [DEBUG] mod_sofia.c:438 Channel sofia/external/441554555666 at 185.35.229.30 <mailto:sofia/external/441554555666 at 185.35.229.30>:5060 hanging up, cause: ORIGINATOR_CANCEL
> 2017-06-18 22:26:34.319525 [INFO] mod_dptools.c:3418 Originate Failed.  Cause: ORIGINATOR_CANCEL
> 2017-06-18 22:26:34.319525 [DEBUG] mod_sofia.c:491 Sending BYE to sofia/external/441554555666 at 185.35.229.30 <mailto:sofia/external/441554555666 at 185.35.229.30>:5060
> 2017-06-18 22:26:34.319525 [DEBUG] switch_core_state_machine.c:60 sofia/external/441554555666 at 185.35.229.30 <mailto:sofia/external/441554555666 at 185.35.229.30>:5060 Standard HANGUP, cause: ORIGINATOR_CANCEL
> 2017-06-18 22:26:34.319525 [DEBUG] switch_core_state_machine.c:852 (sofia/external/441554555666 at 185.35.229.30 <mailto:sofia/external/441554555666 at 185.35.229.30>:5060) State HANGUP going to sleep
> 2017-06-18 22:26:34.319525 [DEBUG] switch_core_state_machine.c:619 (sofia/external/441554555666 at 185.35.229.30 <mailto:sofia/external/441554555666 at 185.35.229.30>:5060) State Change CS_HANGUP -> CS_REPORTING
> 2017-06-18 22:26:34.319525 [DEBUG] switch_core_session.c:2814 sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
> 2017-06-18 22:26:34.319525 [DEBUG] switch_core_state_machine.c:584 (sofia/external/441554555666 at 185.35.229.30 <mailto:sofia/external/441554555666 at 185.35.229.30>:5060) Running State Change CS_REPORTING (Cur 2 Tot 68)
> 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:650 (sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060) State EXECUTE going to sleep
> 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:584 (sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060) Running State Change CS_HANGUP (Cur 2 Tot 68)
> 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:938 (sofia/external/441554555666 at 185.35.229.30 <mailto:sofia/external/441554555666 at 185.35.229.30>:5060) State REPORTING
> 2017-06-18 22:26:34.339569 [ERR] mod_xml_radius.c:930 Didn't match: 185.35.229.30 == ^185\.35\.229\.30 
> 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:850 (sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060) Callstate Change ACTIVE -> HANGUP
> 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:174 sofia/external/441554555666 at 185.35.229.30 <mailto:sofia/external/441554555666 at 185.35.229.30>:5060 Standard REPORTING, cause: ORIGINATOR_CANCEL
> 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:938 (sofia/external/441554555666 at 185.35.229.30 <mailto:sofia/external/441554555666 at 185.35.229.30>:5060) State REPORTING going to sleep
> 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:852 (sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060) State HANGUP
> 2017-06-18 22:26:34.339569 [DEBUG] mod_sofia.c:438 Channel sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 hanging up, cause: NORMAL_CLEARING
> 2017-06-18 22:26:34.339569 [DEBUG] mod_sofia.c:491 Sending BYE to sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060
> 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:60 sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 Standard HANGUP, cause: NORMAL_CLEARING
> 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:852 (sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060) State HANGUP going to sleep
> 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:619 (sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060) State Change CS_HANGUP -> CS_REPORTING
> 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:584 (sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060) Running State Change CS_REPORTING (Cur 2 Tot 68)
> 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:938 (sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060) State REPORTING
> 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:610 (sofia/external/441554555666 at 185.35.229.30 <mailto:sofia/external/441554555666 at 185.35.229.30>:5060) State Change CS_REPORTING -> CS_DESTROY
> 2017-06-18 22:26:34.339569 [DEBUG] switch_core_session.c:1664 Session 68 (sofia/external/441554555666 at 185.35.229.30 <mailto:sofia/external/441554555666 at 185.35.229.30>:5060) Locked, Waiting on external entities
> 2017-06-18 22:26:34.339569 [ERR] mod_xml_radius.c:933 Result of true match: 185.35.228.40 == ^185\.35\.229\.30 
> 2017-06-18 22:26:34.339569 [NOTICE] switch_core_session.c:1682 Session 68 (sofia/external/441554555666 at 185.35.229.30 <mailto:sofia/external/441554555666 at 185.35.229.30>:5060) Ended
> 2017-06-18 22:26:34.339569 [NOTICE] switch_core_session.c:1686 Close Channel sofia/external/441554555666 at 185.35.229.30 <mailto:sofia/external/441554555666 at 185.35.229.30>:5060 [CS_DESTROY]
> 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:741 (sofia/external/441554555666 at 185.35.229.30 <mailto:sofia/external/441554555666 at 185.35.229.30>:5060) Running State Change CS_DESTROY (Cur 1 Tot 68)
> 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:751 (sofia/external/441554555666 at 185.35.229.30 <mailto:sofia/external/441554555666 at 185.35.229.30>:5060) State DESTROY
> 2017-06-18 22:26:34.339569 [DEBUG] mod_sofia.c:343 sofia/external/441554555666 at 185.35.229.30 <mailto:sofia/external/441554555666 at 185.35.229.30>:5060 SOFIA DESTROY
> 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:181 sofia/external/441554555666 at 185.35.229.30 <mailto:sofia/external/441554555666 at 185.35.229.30>:5060 Standard DESTROY
> 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:751 (sofia/external/441554555666 at 185.35.229.30 <mailto:sofia/external/441554555666 at 185.35.229.30>:5060) State DESTROY going to sleep
> 2017-06-18 22:26:34.339569 [INFO] mod_xml_radius.c:1044 mod_xml_radius:  Accounting Stop success
> 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:174 sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 Standard REPORTING, cause: NORMAL_CLEARING
> 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:938 (sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060) State REPORTING going to sleep
> 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:610 (sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060) State Change CS_REPORTING -> CS_DESTROY
> 2017-06-18 22:26:34.339569 [DEBUG] switch_core_session.c:1664 Session 67 (sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060) Locked, Waiting on external entiti
> 2017-06-18 22:26:34.339569 [NOTICE] switch_core_session.c:1682 Session 67 (sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060) Ended
> 2017-06-18 22:26:34.339569 [NOTICE] switch_core_session.c:1686 Close Channel sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 [CS_DESTROY]
> 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:741 (sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060) Running State Change CS_DESTROY (Cur 0 Tot 68)
> 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:751 (sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060) State DESTROY
> 2017-06-18 22:26:34.339569 [DEBUG] mod_sofia.c:343 sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 SOFIA DESTROY
> 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:181 sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 Standard DESTROY
> 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:751 (sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060) State DESTROY going to sleep
> 
> 
> U 185.35.228.51:5060 <http://185.35.228.51:5060/> -> 185.35.228.40:5080 <http://185.35.228.40:5080/>
> INVITE sip:441554555666 at 185.35.228.40:5080 <> SIP/2.0.
> Via: SIP/2.0/UDP 185.35.228.51:5060;branch=z9hG4bK-27036-1-0.
> From: sipp <sip:sipp at 185.35.228.51:5060 <>>;tag=27036SIPpTag001.
> To: 441554555666 <sip:441554555666 at 185.35.228.40:5080 <>>.
> Call-ID: 1-27036 at 185.35.228.51 <mailto:1-27036 at 185.35.228.51>.
> CSeq: 1 INVITE.
> Contact: sip:sipp at 185.35.228.51:5060 <>.
> Max-Forwards: 70.
> Subject: Performance Test.
> Content-Type: application/sdp.
> Content-Length:   137.
> .
> v=0.
> o=user1 53655765 2353687637 IN IP4 185.35.228.51.
> s=-.
> c=IN IP4 185.35.228.51.
> t=0 0.
> m=audio 6000 RTP/AVP 0.
> a=rtpmap:0 PCMU/8000.
> 
> #
> U 185.35.228.40:5080 <http://185.35.228.40:5080/> -> 185.35.228.51:5060 <http://185.35.228.51:5060/>
> SIP/2.0 100 Trying.
> Via: SIP/2.0/UDP 185.35.228.51:5060;branch=z9hG4bK-27036-1-0.
> From: sipp <sip:sipp at 185.35.228.51:5060 <>>;tag=27036SIPpTag001.
> To: 441554555666 <sip:441554555666 at 185.35.228.40:5080 <>>.
> Call-ID: 1-27036 at 185.35.228.51 <mailto:1-27036 at 185.35.228.51>.
> CSeq: 1 INVITE.
> User-Agent: FreeSWITCH-mod_sofia/1.6.18+git~20170612T211449Z~6e79667c0a~64bit.
> Content-Length: 0.
> .
> 
> #
> U 185.35.228.40:5080 <http://185.35.228.40:5080/> -> 185.35.228.51:5060 <http://185.35.228.51:5060/>
> SIP/2.0 180 Ringing.
> Via: SIP/2.0/UDP 185.35.228.51:5060;branch=z9hG4bK-27036-1-0.
> From: sipp <sip:sipp at 185.35.228.51:5060 <>>;tag=27036SIPpTag001.
> To: 441554555666 <sip:441554555666 at 185.35.228.40:5080 <>>;tag=91vp8601aS4Qp.
> Call-ID: 1-27036 at 185.35.228.51 <mailto:1-27036 at 185.35.228.51>.
> CSeq: 1 INVITE.
> Contact: <sip:441554555666 at 185.35.228.40:5080;transport=udp <>>.
> User-Agent: FreeSWITCH-mod_sofia/1.6.18+git~20170612T211449Z~6e79667c0a~64bit.
> Accept: application/sdp.
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY.
> Supported: timer, path, replaces.
> Allow-Events: talk, hold, conference, refer.
> Content-Length: 0.
> Remote-Party-ID: "Outbound Call" <sip:441554555666 at 185.35.228.40 <>>;party=calling;privacy=off;screen=no.
> .
> 
> #
> U 185.35.228.40:5080 <http://185.35.228.40:5080/> -> 185.35.228.51:5060 <http://185.35.228.51:5060/>
> SIP/2.0 200 OK.
> Via: SIP/2.0/UDP 185.35.228.51:5060;branch=z9hG4bK-27036-1-0.
> From: sipp <sip:sipp at 185.35.228.51:5060 <>>;tag=27036SIPpTag001.
> To: 441554555666 <sip:441554555666 at 185.35.228.40:5080 <>>;tag=91vp8601aS4Qp.
> Call-ID: 1-27036 at 185.35.228.51 <mailto:1-27036 at 185.35.228.51>.
> CSeq: 1 INVITE.
> Contact: <sip:441554555666 at 185.35.228.40:5080;transport=udp <>>.
> User-Agent: FreeSWITCH-mod_sofia/1.6.18+git~20170612T211449Z~6e79667c0a~64bit.
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY.
> Supported: timer, path, replaces.
> Allow-Events: talk, hold, conference, refer.
> Content-Type: application/sdp.
> Content-Disposition: session.
> Content-Length: 166.
> Remote-Party-ID: "Outbound Call" <sip:441554555666 at 185.35.228.40 <>>;party=calling;privacy=off;screen=no.
> .
> v=0.
> o=FreeSWITCH 1497797499 1497797500 IN IP4 185.35.228.40.
> s=FreeSWITCH.
> c=IN IP4 185.35.228.40.
> t=0 0.
> m=audio 25252 RTP/AVP 0.
> a=rtpmap:0 PCMU/8000.
> a=ptime:20.
> 
> #
> U 185.35.228.51:5060 <http://185.35.228.51:5060/> -> 185.35.228.40:5080 <http://185.35.228.40:5080/>
> ACK sip:441554555666 at 185.35.228.40:5080 <> SIP/2.0.
> Via: SIP/2.0/UDP 185.35.228.51:5060;branch=z9hG4bK-27036-1-5.
> From: sipp <sip:sipp at 185.35.228.51:5060 <>>;tag=27036SIPpTag001.
> To: 441554555666 <sip:441554555666 at 185.35.228.40:5080 <>>;tag=91vp8601aS4Qp.
> Call-ID: 1-27036 at 185.35.228.51 <mailto:1-27036 at 185.35.228.51>.
> CSeq: 1 ACK.
> Contact: sip:sipp at 185.35.228.51:5060 <>.
> Max-Forwards: 70.
> Subject: Performance Test.
> Content-Length: 0.
> .
> 
> #
> U 185.35.228.40:5080 <http://185.35.228.40:5080/> -> 185.35.228.51:5060 <http://185.35.228.51:5060/>
> BYE sip:sipp at 185.35.228.51:5060 <> SIP/2.0.
> Via: SIP/2.0/UDP 185.35.228.40:5080;rport;branch=z9hG4bKUBe0HvDjHF1XN.
> Max-Forwards: 70.
> From: 441554555666 <sip:441554555666 at 185.35.228.40:5080 <>>;tag=91vp8601aS4Qp.
> To: sipp <sip:sipp at 185.35.228.51:5060 <>>;tag=27036SIPpTag001.
> Call-ID: 1-27036 at 185.35.228.51 <mailto:1-27036 at 185.35.228.51>.
> CSeq: 108575311 BYE.
> User-Agent: FreeSWITCH-mod_sofia/1.6.18+git~20170612T211449Z~6e79667c0a~64bit.
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY.
> Supported: timer, path, replaces.
> Reason: Q.850;cause=16;text="NORMAL_CLEARING".
> Content-Length: 0.
> .
> 
> #
> U 185.35.228.51:5060 <http://185.35.228.51:5060/> -> 185.35.228.40:5080 <http://185.35.228.40:5080/>
> SIP/2.0 200 OK.
> Via: SIP/2.0/UDP 185.35.228.40:5080;rport;branch=z9hG4bKUBe0HvDjHF1XN.
> From: 441554555666 <sip:441554555666 at 185.35.228.40:5080 <>>;tag=91vp8601aS4Qp.
> To: sipp <sip:sipp at 185.35.228.51:5060 <>>;tag=27036SIPpTag001.
> Call-ID: 1-27036 at 185.35.228.51 <mailto:1-27036 at 185.35.228.51>.
> CSeq: 108575311 BYE.
> Contact: <sip:185.35.228.51:5060;transport=UDP <>>.
> Content-Length: 0.
> 
> 
> 
> NGREP of SIP messages from FS to terminator
> 
> 
> U 185.35.228.40:5080 <http://185.35.228.40:5080/> -> 185.35.228.48:5060 <http://185.35.228.48:5060/>
> INVITE sip:441554555666 at 185.35.228.48:5060 <> SIP/2.0.
> Via: SIP/2.0/UDP 185.35.228.40:5080;rport;branch=z9hG4bKXX0HNjFSB1D3c.
> Max-Forwards: 69.
> From: "sipp" <sip:sipp at 185.35.228.51 <>>;tag=eFUjHeNKv5KNg.
> To: <sip:441554555666 at 185.35.229.30:5060 <>>.
> Call-ID: c755e7c8-cf13-1235-25a9-363165383663.
> CSeq: 108575420 INVITE.
> Contact: <sip:mod_sofia at 185.35.228.40:5080 <>>.
> User-Agent: FreeSWITCH-mod_sofia/1.6.18+git~20170612T211449Z~6e79667c0a~64bit.
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY.
> Supported: timer, path, replaces.
> Allow-Events: talk, hold, conference, refer.
> Content-Type: application/sdp.
> Content-Disposition: session.
> Content-Length: 222.
> X-FS-Support: update_display,send_info.
> Remote-Party-ID: "sipp" <sip:sipp at 185.35.228.40 <>>;party=calling;screen=yes;privacy=off.
> .
> v=0.
> o=FreeSWITCH 1497801741 1497801743 IN IP4 185.35.228.40.
> s=FreeSWITCH.
> c=IN IP4 185.35.228.40.
> t=0 0.
> m=audio 21228 RTP/AVP 0 101.
> a=rtpmap:0 PCMU/8000.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-16.
> a=ptime:20.
> 
> #
> U 185.35.228.48:5060 <http://185.35.228.48:5060/> -> 185.35.228.40:5080 <http://185.35.228.40:5080/>
> SIP/2.0 180 Ringing.
> Via: SIP/2.0/UDP 185.35.228.40:5080;rport;branch=z9hG4bKXX0HNjFSB1D3c.
> From: "sipp" <sip:sipp at 185.35.228.51 <>>;tag=eFUjHeNKv5KNg.
> To: <sip:441554555666 at 185.35.229.30:5060 <>>;tag=31480SIPpTag018.
> Call-ID: c755e7c8-cf13-1235-25a9-363165383663.
> CSeq: 108575420 INVITE.
> Contact: <sip:185.35.228.48:5060;transport=UDP <>>.
> Content-Length: 0.
> .
> 
> #
> U 185.35.228.48:5060 <http://185.35.228.48:5060/> -> 185.35.228.40:5080 <http://185.35.228.40:5080/>
> SIP/2.0 200 OK.
> Via: SIP/2.0/UDP 185.35.228.40:5080;rport;branch=z9hG4bKXX0HNjFSB1D3c.
> From: "sipp" <sip:sipp at 185.35.228.51 <>>;tag=eFUjHeNKv5KNg.
> To: <sip:441554555666 at 185.35.229.30:5060 <>>;tag=31480SIPpTag018.
> Call-ID: c755e7c8-cf13-1235-25a9-363165383663.
> CSeq: 108575420 INVITE.
> Contact: <sip:185.35.228.48:5060;transport=UDP <>>.
> Content-Type: application/sdp.
> Content-Length:   137.
> .
> v=0.
> o=user1 53655765 2353687637 IN IP4 185.35.228.48.
> s=-.
> c=IN IP4 185.35.228.48.
> t=0 0.
> m=audio 6000 RTP/AVP 0.
> a=rtpmap:0 PCMU/8000.
> 
> #
> U 185.35.228.40:5080 <http://185.35.228.40:5080/> -> 185.35.228.48:5060 <http://185.35.228.48:5060/>
> ACK sip:185.35.228.48:5060;transport=UDP <> SIP/2.0.
> Via: SIP/2.0/UDP 185.35.228.40:5080;rport;branch=z9hG4bKy6SaQD0v893Nr.
> Max-Forwards: 70.
> From: "sipp" <sip:sipp at 185.35.228.51 <>>;tag=eFUjHeNKv5KNg.
> To: <sip:441554555666 at 185.35.229.30:5060 <>>;tag=31480SIPpTag018.
> Call-ID: c755e7c8-cf13-1235-25a9-363165383663.
> CSeq: 108575420 ACK.
> Contact: <sip:mod_sofia at 185.35.228.40:5080 <>>.
> Content-Length: 0.
> .
> 
> #
> U 185.35.228.40:5080 <http://185.35.228.40:5080/> -> 185.35.228.48:5060 <http://185.35.228.48:5060/>
> BYE sip:185.35.228.48:5060;transport=UDP <> SIP/2.0.
> Via: SIP/2.0/UDP 185.35.228.40:5080;rport;branch=z9hG4bKZFK3r8g05jt8K.
> Max-Forwards: 70.
> From: "sipp" <sip:sipp at 185.35.228.51 <>>;tag=eFUjHeNKv5KNg.
> To: <sip:441554555666 at 185.35.229.30:5060 <>>;tag=31480SIPpTag018.
> Call-ID: c755e7c8-cf13-1235-25a9-363165383663.
> CSeq: 108575421 BYE.
> User-Agent: FreeSWITCH-mod_sofia/1.6.18+git~20170612T211449Z~6e79667c0a~64bit.
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY.
> Supported: timer, path, replaces.
> Reason: SIP;cause=487;text="ORIGINATOR_CANCEL".
> Content-Length: 0.
> .
> 
> #
> U 185.35.228.48:5060 <http://185.35.228.48:5060/> -> 185.35.228.40:5080 <http://185.35.228.40:5080/>
> SIP/2.0 200 OK.
> Via: SIP/2.0/UDP 185.35.228.40:5080;rport;branch=z9hG4bKZFK3r8g05jt8K.
> From: "sipp" <sip:sipp at 185.35.228.51 <>>;tag=eFUjHeNKv5KNg.
> To: <sip:441554555666 at 185.35.229.30:5060 <>>;tag=31480SIPpTag018.
> Call-ID: c755e7c8-cf13-1235-25a9-363165383663.
> CSeq: 108575421 BYE.
> Contact: <sip:185.35.228.48:5060;transport=UDP <>>.
> Content-Length: 0.
> .
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> http://www.cluecon.com <http://www.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>
> http://www.freeswitch.org <http://www.freeswitch.org/>
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> http://www.cluecon.com <http://www.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>
> http://www.freeswitch.org <http://www.freeswitch.org/>
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170619/3a791c22/attachment-0001.html>

From Chris.Young at enghouse.com  Mon Jun 19 09:05:12 2017
From: Chris.Young at enghouse.com (Chris Young)
Date: Mon, 19 Jun 2017 09:05:12 +0000
Subject: [Freeswitch-users] "Cannot Blind Transfer 1 Legged calls" message
Message-ID: <95a7f732146d4cd8a3ca480e8a8e1679@UK-MAIL-001.edge.local>

Hi all,

I'm hoping that some of you clever folk may be able to help me out with a REFER problem I'm facing.

If I originate a call from FreeSWITCH to an IVR, and the IVR attempts to transfer the call to another destination using REFER, FreeSWITCH fails with a 403 Forbidden error and reports 'Cannot Blind Transfer 1 Legged calls' in the log. FreeSWITCH is quite correct that there is only one leg of course but should it not be possible for an INVITE to be sent to the address in the Refer-To header anyway? This scenario seems more or less the same as the first 'basic transfer' example in RFC5589 so I think it should be possible unless I am missing something obvious.

Are there any special configuration options needed to make this work? I tried setting proxy-refer but it didn't make any difference.

Kind regards,
Chris

Chris Young
Senior Software Engineer
[cid:image7482a0.PNG at dc00f514.4fb88205]
t:  +44 118 943 9249
e:  chris.young at enghouse.com<mailto:chris.young at enghouse.com>
w: www.enghouseinteractive.co.uk<http://www.enghouseinteractive.co.uk>
[cid:image6c1ba6.PNG at b8963a35.449174f5]
Enghouse Interactive (UK) Ltd is a company registered in England and Wales. Registered number: 04230977. Registered office: Imperium, Imperial Way, Reading, Berkshire, RG2 0TD

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170619/80ca2801/attachment.html>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: image001.png
Type: image/png
Size: 1045 bytes
Desc: image001.png
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170619/80ca2801/attachment.png>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: image002.png
Type: image/png
Size: 5097 bytes
Desc: image002.png
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170619/80ca2801/attachment-0001.png>

From adrian.worutowicz at esifrance.net  Mon Jun 19 12:52:31 2017
From: adrian.worutowicz at esifrance.net (Adrian Worutowicz)
Date: Mon, 19 Jun 2017 14:52:31 +0200
Subject: [Freeswitch-users] Build Problem in VS2015
In-Reply-To: <CAC7ttFE9Bq_UJx4J-iwf6gP7g1qrOmBxjmdJtEn_FKucs06njA@mail.gmail.com>
References: <80c901d2e5f4$7e26f930$7a74eb90$@tollfreegateway.com>
 <594417cf.462ded0a.12ca4.f4aeSMTPIN_ADDED_BROKEN@mx.google.com>
 <CAC7ttFE9Bq_UJx4J-iwf6gP7g1qrOmBxjmdJtEn_FKucs06njA@mail.gmail.com>
Message-ID: <008401d2e8fa$ea804ab0$bf80e010$@worutowicz@esifrance.net>

I asked a friend of mine to compile and it worked fine.

So it is either a pb of my VS2015 install or a question of Win7 32/64 bit (I have the 32bit version).

 

Thanks,

A.

De : FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de gregor at infomedia.si -- Gregor Nanger
Envoyé : vendredi 16 juin 2017 21:19
À : FreeSWITCH Users Help
Objet : Re: [Freeswitch-users] Build Problem in VS2015

 

I do not have problems compiling with visual Studio. Except for cloning, I use same command as stated in wiki. Then open in visual Studio and Build solution. 

 

On Fri, Jun 16, 2017, 19:39 Adrian Worutowicz <adrian.worutowicz at esifrance.net> wrote:

I followed your steps, but unfortunately I got the same result.

Probably I’m missing something in my VS install.

 

Thanks a lot anyway…

 

 

 

De : FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de krice at tollfreegateway.com
Envoyé : jeudi 15 juin 2017 18:29
À : 'FreeSWITCH Users Help'
Objet : Re: [Freeswitch-users] Build Problem in VS2015

 

Not sure whats you are doing incorrect here, but I have just built master, I use the built in git bits with VS2015, and then drop to a command prompt (via the team explorer tab, select branches, right click the repo and select open command prompt)

 

Then git pull, git clean -fdx, git reset –hard origin/master , git pull

 

>From here back to the solution explorer open the FreeSWITCH.2015 solution file and build as normal…

 

I think you have something skewed there old ssl vs new ssl bits 

 

From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Adrian Worutowicz
Sent: Thursday, June 15, 2017 4:44 AM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Build Problem in VS2015

 

Hello,

 

I try to recompile FS in VS2015 without success.

I took FS sources from git master.

 

git config --global core.autocrlf false

git clone https://stash.freeswitch.org/scm/fs/freeswitch.git /c/ESI/Components/FreeSwitch/ <https://stash.freeswitch.org/scm/fs/freeswitch.git%20/c/ESI/Components/FreeSwitch/> 

 

I have wix311 for VS2015 installed.

 

For example it searches in folder 'openssl-1.0.2k' while only a folder 'openssl' exists.

 

I tried to recompile mod_PortAudio, and I got c:\ESI\components\freeswitch\src\mod\endpoints\mod_portaudio\pablio.h(55): fatal error C1083: Impossible d'ouvrir le fichier include : 'portaudio.h' : No such file or directory.

 

Indeed 'portaudio.h' does not exist.

Plenty of other errors in the attached file.

 

What do I miss?

 

Thanks in advance,

Adrian.

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.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
http://www.freeswitch.org

-- 


Gregor Nanger

 

CTO
t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485 
• Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia 
• www.infomedia.si <http://www.infomedia.si/> 

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170619/ae7b7486/attachment.html>

From mike at jerris.com  Mon Jun 19 16:34:56 2017
From: mike at jerris.com (Michael Jerris)
Date: Mon, 19 Jun 2017 12:34:56 -0400
Subject: [Freeswitch-users] Callcenter module,
 can I originate call for an agent in uuid-standby mode ?
In-Reply-To: <CAEK98WarT_sPVc4HHZoVJqo+BEpGDduwH87w-Ar3C4NphMEwwg@mail.gmail.com>
References: <CAEK98WarT_sPVc4HHZoVJqo+BEpGDduwH87w-Ar3C4NphMEwwg@mail.gmail.com>
Message-ID: <4770F7F5-BCBB-47B6-A909-263EEFEADFE4@jerris.com>

mod_fifo is a much more feature rich version of a call queue than mod_callcenter.  You might want to check that out instead.

> On Jun 14, 2017, at 8:09 AM, Khalil Khamlichi <khamlichi.khalil at gmail.com> wrote:
> 
> Hi,
> 
> I need to give my agents ability to make manual calls, hopefully without leaving their actually established call (they are in uuid-standby mode and in Idle state so there is no live member on the line ). 
> 
> my questions:
> 
> Is it possible to originate a new call and bridge with agent uuid-standby session ? 
> would it not break the callcenter establised uuid-standby session ? 
> would the agent return to its uuid-standby session after the originated call is hangup ?
> 
> ofcourse if this is too complicated, I would just connect the agent thru a second line, while leaving his uuid-standby call on the first line, though it would be so cool to somehow stay on that same uuid-standby session and enjoy both calllcenter module and manual dialing.
> 
> Thanks in advance, and I appreciate your help.
> 
> Khalil



From mike at jerris.com  Mon Jun 19 16:43:14 2017
From: mike at jerris.com (Michael Jerris)
Date: Mon, 19 Jun 2017 12:43:14 -0400
Subject: [Freeswitch-users] Freeswitch sslv3 support
In-Reply-To: <CAGCz3RCjv0GHRkYmaPd31QLjd6pWBTdgxbbBmdeVRYwz0vU=RA@mail.gmail.com>
References: <CAGCz3RBiBhE7ism+Wfr81HShk42TDht4c-Yp_DY0cp35OCNqLQ@mail.gmail.com>
 <CAGCz3RCjv0GHRkYmaPd31QLjd6pWBTdgxbbBmdeVRYwz0vU=RA@mail.gmail.com>
Message-ID: <83A13BEB-579E-4B56-9C4B-9B0AA355A4EE@jerris.com>

our websocket code should already be limited to either tls 1.1 and tls 1.2 or just tls 1.2 (i can’t remember which)… if this isn’t the case, please open a Jira for this.  There should be no browsers that support web sockets that don’t support at least tls 1.1.



> On Jun 15, 2017, at 4:07 AM, Agustí Ubalde Bellot <agubbe at gmail.com> wrote:
> 
> Hi Brian,
> 
> Is possible to disable for web socket secure connections too?
> 
> 
> Thanks,
> Agustí
> 
> 2017-06-13 13:24 GMT+02:00 Agustí Ubalde Bellot <agubbe at gmail.com <mailto:agubbe at gmail.com>>:
> Hi all,
> 
> Is there a FreeSWITCH update where sslv3 support is disabled?
> 
> 
> Thanks,
> Agustí

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170619/4380d4b0/attachment-0001.html>

From mike at jerris.com  Mon Jun 19 16:45:56 2017
From: mike at jerris.com (Michael Jerris)
Date: Mon, 19 Jun 2017 12:45:56 -0400
Subject: [Freeswitch-users] question about HA solution
In-Reply-To: <CAJTkRNuenLDn-mu34cZ4dgxNH_P0Wq8nPx4VF21dmgBFB3Zq0w@mail.gmail.com>
References: <CADcrYG+RHGQ2fC7dSKK9x9WfVvYcTczOtbppCLgttRCfMGqArA@mail.gmail.com>
 <CAFLo2QxxbZ5Ged0J0p9_VfQ2nkGtuV0CDp9G763EjjuvqUv8iw@mail.gmail.com>
 <CADcrYG+sO0+WYUtYVN6pkpU-5SwyfurfshdUdhxb-c4Az1s49Q@mail.gmail.com>
 <8A13E0EC-FA13-4267-80F6-CE1A8E8360CF@jerris.com>
 <CADcrYGLbN1Ut6riF1Lf7UvwG9ypvR2ekWxt1m=e5BmqOzP_c=w@mail.gmail.com>
 <E55363B0-487F-4F65-8F3F-78620AB1AB73@jerris.com>
 <CADcrYGKK=cTmVzG5oY0MV3b+4CLqg0=YzGvCiDZNZdBqQe3mMA@mail.gmail.com>
 <CAEwTsAhpLWcAoK8jfaEZ+HS2NeNLLEgbyAZkNRiTJ4RMNRet4Q@mail.gmail.com>
 <CAJTkRNuenLDn-mu34cZ4dgxNH_P0Wq8nPx4VF21dmgBFB3Zq0w@mail.gmail.com>
Message-ID: <A337B848-B155-4ECC-BF35-99FE48EFA917@jerris.com>

what is “channels_pkey” … thats not something thats anywhere in our codebase.


> On Jun 15, 2017, at 5:40 AM, Igor Olhovskiy <igorolhovskiy at gmail.com> wrote:
> 
> Hi!
> Same situation here.
> Idea is:
> I’m having Freeswitch HA (keepalived, working, same database, calls recovering…)
> If I look on «show calls» at slave node, I see calls on master node. 
> I crash master node (with «fsctl crash»), calls are transferred to slave node, restored, but when I run «show calls» on this (slave) node again, I see 0 calls. But calls are actually going on.
> 
> So, it’s seems impossible to have 2nd recover on already recovered call.
> 
> 
> In DB logs seen an errors like
> insert into channels (uuid,direction,created,created_epoch, name,state,callstate,dialplan,context,hostname,initial_cid_name,initial_cid_num,initial_ip_addr,initial_dest,initial_dialplan,initial_context) values('57904410-a8ad-4c28-a88a-83bd2280e146','outbound','2017-06-15 19:30:32','1497519032','sofia/internal/113-akbepcb59gt2a at 172.17.240.50:5060 <http://113-akbepcb59gt2a at 172.17.240.50:5060/>','CS_INIT','DOWN','XML','sip303.empowervoice.com <http://sip303.empowervoice.com/>','blueAPACHE_test','103','103','172.17.240.50','113-akbepcb59gt2a','XML','sip303.empowervoice.com <http://sip303.empowervoice.com/>')
> Jun 15 19:30:31 E2-EVL-T-DB-01 postgres[28042]: [109-1] 2017-06-15 19:30:31 AEST [28042-103] freeswitch at freeswitch ERROR:  duplicate key value violates unique constraint «channels_pkey"
> Or like
> statement: insert into calls (call_uuid,call_created,call_created_epoch,caller_uuid,callee_uuid,hostname) values ('ffaf3eb5-3fc5-47fe-adef-cc4dddf53bab','2017-06-15 19:30:32','1497519032','ffaf3eb5-3fc5-47fe-adef-cc4dddf53bab','57904410-a8ad-4c28-a88a-83bd2280e146','blueAPACHE_test')
> Jun 15 19:30:31 E2-EVL-T-DB-01 postgres[28042]: [147-1] 2017-06-15 19:30:31 AEST [28042-142] freeswitch at freeswitch ERROR:  duplicate key value violates unique constraint «calls_pkey"
> 
> Also I see much queries like this
> delete from calls where (caller_uuid=‘ffaf3eb5-3fc5-47fe-adef-cc4dddf53bab’ or callee_uuid='ffaf3eb5-3fc5-47fe-adef-cc4dddf53bab')
> delete from recovery where runtime_uuid!=‘91f571c5-e0d2-462e-aa84-e4ca07052119’ and technology=‘sofia’….
> when calls are switched.
> 
> So, can this help to point an issue?
> 
> 2017-06-08 18:48 GMT+03:00 Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>>:
> check your db logs as nothing we are doing should be clearing those.
> 
> On Thu, Jun 8, 2017 at 4:08 AM Denys Pozniak <denys.pozniak at crazycall.com <mailto:denys.pozniak at crazycall.com>> wrote:
> Hello!
> 
> My configs:
> 
> switch.conf.xml
> 
>     <param name="switchname" value="freeswitch01"/>
>     <param name="core-recovery-db-dsn" value="odbc://freeswitch:root:ubuntu"/>
>     <param name="core-db-dsn" value="odbc://freeswitch:root:ubuntu"/>
>     <param name="auto-create-schemas" value="false"/>
>     <param name="auto-clear-sql" value="false"/>
> 
> 
> external.conf.xml
> 
>    <param name="track-calls" value="true"/>
>    <param name="odbc-dsn" value="odbc://freeswitch:root:ubuntu"/>
> 
> 
> On 7 June 2017 at 17:35, Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>> wrote:
> That param should keep it from doing so, if its not you are not setting it somehow or something else is wiping the db.
> 
>> On Jun 5, 2017, at 1:50 PM, Denys Pozniak <denys.pozniak at crazycall.com <mailto:denys.pozniak at crazycall.com>> wrote:
>> 
>> Yes, correct. But when you restart FS on slave, it will erase database. And option auto-clear-sql=false not working for me.
>> 
>> On Jun 5, 2017 6:32 PM, "Michael Jerris" <mike at jerris.com <mailto:mike at jerris.com>> wrote:
>> recovered calls will get new entries in the table.
>> 
>>> On Jun 5, 2017, at 7:41 AM, Denys Pozniak <denys.pozniak at crazycall.com <mailto:denys.pozniak at crazycall.com>> wrote:
>>> 
>>> Hello!
>>> 
>>> Thank you Raymond about your explanation, but I dont agree with some point: 
>>> If it really need an answer about your question -- "if it is possible to move calls back".   I think it's unnecessary.  - in my case I have two not equal servers, so I need to have only one as a master.
>>> If switchover happens I need to have ability to restore master back.
>>> 
>>> Thank you Luis for your link, you can do simple test to understand what I am talking about: do call -> check on master and slave #show channels -> restart FS on slave -> check on master #show channels. In my case I dont see any active calls after this, so restoring back is not possible.
>>> 
>>> 
>>> 
>>> On 3 June 2017 at 22:16, Luis Daniel Lucio Quiroz <luis.daniel.lucio at gmail.com <mailto:luis.daniel.lucio at gmail.com>> wrote:
>>> You may want to read this article.
>>> 
>>> http://inside-out.xyz/technology/how-to-configure-freeswitch-for-ha.html <http://inside-out.xyz/technology/how-to-configure-freeswitch-for-ha.html>
>>> 
>>> Le 31 mai 2017 6:29 PM, "Denys Pozniak" <denys.pozniak at crazycall.com <mailto:denys.pozniak at crazycall.com>> a écrit :
>>> Hello!
>>> 
>>> I built FS HA solution based on keepalived and mysql master-master.
>>> It works ok generally, but as I understand FS after restarting cleaning own database.
>>> 
>>> So when node1 fails calls jump to node2, after script restarts node1 it is not possible to move calls back.
>>> 
>>> Tried options in switch.conf.xml, but no luck:
>>> 
>>>     <param name="auto-create-schemas" value="false"/>
>>>     <param name="auto-clear-sql" value="false"/>
>>> 
>>> Is there is a way to solve this?
>>> 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> http://www.cluecon.com <http://www.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>
> http://www.freeswitch.org <http://www.freeswitch.org/>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> http://www.cluecon.com <http://www.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>
> http://www.freeswitch.org <http://www.freeswitch.org/>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> http://www.cluecon.com <http://www.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>
> http://www.freeswitch.org <http://www.freeswitch.org/>
> 
> 
> 
> -- 
> Best regards,
> Igor
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170619/06e4e245/attachment-0001.html>

From mike at jerris.com  Mon Jun 19 16:46:52 2017
From: mike at jerris.com (Michael Jerris)
Date: Mon, 19 Jun 2017 12:46:52 -0400
Subject: [Freeswitch-users] group_confirm_file multiple files
In-Reply-To: <CAEeV61jMv9ytvzX_8URn+M7wiC=vpW0+3Nr7VjbE-Ld8jvR5eA@mail.gmail.com>
References: <CAEeV61jMv9ytvzX_8URn+M7wiC=vpW0+3Nr7VjbE-Ld8jvR5eA@mail.gmail.com>
Message-ID: <F2F379FA-CE02-4750-8EBF-16D1E3FF16A3@jerris.com>

https://freeswitch.org/confluence/display/FREESWITCH/mod_dptools%3A+file_string <https://freeswitch.org/confluence/display/FREESWITCH/mod_dptools:+file_string>

> On Jun 15, 2017, at 5:54 AM, Matt Broad <matt at supportedbusiness.com> wrote:
> 
> Hi,
> 
> I'm wondering if it is possible to play multiple files using the group_confirm_file function.
> 
> I have 2 audio files that I would like to play 1 after the other and then wait for the confirm key.
> 
> 
> I have tried using mod_file_string, but get an error "Error from mpg123: File access error. (code 22)", I assume this is due to the fact it is reading the file string as one file rather than 2 separated by the ! delimiter.
> 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170619/699f4901/attachment.html>

From michael at mailworks.org  Mon Jun 19 16:48:11 2017
From: michael at mailworks.org (Michael Avers)
Date: Mon, 19 Jun 2017 09:48:11 -0700
Subject: [Freeswitch-users] Callcenter module,
 can I originate call for an agent in uuid-standby mode ?
In-Reply-To: <4770F7F5-BCBB-47B6-A909-263EEFEADFE4@jerris.com>
References: <CAEK98WarT_sPVc4HHZoVJqo+BEpGDduwH87w-Ar3C4NphMEwwg@mail.gmail.com>
 <4770F7F5-BCBB-47B6-A909-263EEFEADFE4@jerris.com>
Message-ID: <1497890891.572062.1014326656.673B64EF@webmail.messagingengine.com>

What makes mod_fifo better these days? Some reason I was under the impression mod_callcenter is a better choice. Can you please give some real world examples where mod_fifo excels compared to mod_callcenter, or features that are not possible to implement with the latter?

Thanks
Mike

On Mon, Jun 19, 2017, at 09:34 AM, Michael Jerris wrote:
> mod_fifo is a much more feature rich version of a call queue than mod_callcenter.  You might want to check that out instead.
> 


From mike at jerris.com  Mon Jun 19 18:08:48 2017
From: mike at jerris.com (Michael Jerris)
Date: Mon, 19 Jun 2017 14:08:48 -0400
Subject: [Freeswitch-users] When calling out ,
 freeswitch doesn't get DTMF ?
In-Reply-To: <CALXCt0q_s1CByZPrQLN8fa=1-_Kat5QZB4FVohZp589Q9kf=0Q@mail.gmail.com>
References: <f823cdd8-a354-3698-d8c5-484b1b4cc8b4@xbipin.com>
 <CAEJMVkDDQX=Ni3AOQxQOo=joGoAeYroxioVP=cu9w6Y2m7kLPA@mail.gmail.com>
 <15c898b5d58.2779.b07ebdf329620b8089087c7205b03f01@xbipin.com>
 <310cf71c.8fc3.15c8ba9c4f6.Coremail.eastour@163.com>
 <CAFGRPVqFA+tpaOQkAGGCR7s5gY0z0TbdArWFdkfnsaaG4Z9Dgw@mail.gmail.com>
 <2a18422f.ef8.15c99b7b9bb.Coremail.eastour@163.com>
 <CALXCt0pKKEJoSeXXZ=tn04nifEZvQCRSFCDKL8jSu+pfOgvRNQ@mail.gmail.com>
 <6c11dd0f.c2c1.15c9c727dc7.Coremail.eastour@163.com>
 <CAFLo2QwwwE6xeXBN9DkM1VEecEJpwnDMmQ7iT9EvSyx9RqE9mg@mail.gmail.com>
 <2f48505d.d9c5.15ca1d28424.Coremail.eastour@163.com>
 <59401B92020000310000A34D@mail.tedssupply.com>
 <CALXCt0qw4DCJ2Q0cF+kVEOpjCjp2T8eFgWbYx03sN=z5APBGGQ@mail.gmail.com>
 <59412DB4020000310000A381@mail.tedssupply.com>
 <CALXCt0rTBTBTfT+e1t0UCfpwmWyOhz1-yeQ2s6T8mSTYAnS-+w@mail.gmail.com>
 <7d1e3f12.4750.15ca994d89d.Coremail.eastour@163.com>
 <CALXCt0oRdP=8M_vbQpevzUs=NAO-rDVx0BqXJzv6mhy9Q5vZkg@mail.gmail.com>
 <7611bb49.72be.15caf10d2a0.Coremail.eastour@163.com>
 <CALXCt0q_s1CByZPrQLN8fa=1-_Kat5QZB4FVohZp589Q9kf=0Q@mail.gmail.com>
Message-ID: <8CCB1A80-0047-4D23-91BC-AFF7FB70036C@jerris.com>

its worth testing both most recent master and 1.6.18.

> On Jun 17, 2017, at 4:32 AM, Giovanni Maruzzelli <gmaruzz at gmail.com> wrote:
> 
> 
> 
> On 16 June 2017 at 06:02, chenyzhi <eastour at 163.com <mailto:eastour at 163.com>> wrote:
> Yes ,I can hear all the IVR prompt voices correctly.
> 
> I don't think it's  a NAT problem ,because both the x-lite and the freeswitch are in the same LAN.
> 
> The sip trace log is in the attatchment. Thank you.
> 
> PS  I tested this on another freeswitch box ,version:
> FreeSWITCH Version 1.6.16+git~20170403T142423Z~e6d643b29c~32bit (git e6d643b 2017-04-03 14:24:23Z 32bit)
> It can detect dtmf on outgoing calls.  Maybe this only happens on FreeSWITCH Version 1.9.0+git~20170518T231917Z~a1fc18aee5~32bit (git a1fc18a 2017-05-18 23:19:17Z 32bit)
> 
> 
> then use the stable version, and open a jira for this issue citing the master version you are using
> 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170619/04c63b9c/attachment.html>

From mike at jerris.com  Mon Jun 19 18:10:07 2017
From: mike at jerris.com (Michael Jerris)
Date: Mon, 19 Jun 2017 14:10:07 -0400
Subject: [Freeswitch-users] enable Portal, error 404
In-Reply-To: <8779966.9K6fzknjRZ@stefan-ubu>
References: <8779966.9K6fzknjRZ@stefan-ubu>
Message-ID: <F123F930-D3FB-4635-8AC2-0505589557E4@jerris.com>

bad credentials?

> On Jun 17, 2017, at 10:55 AM, Stefan Fuhrmann <stefan at fuhrmann.homedns.org> wrote:
> 
> Hello all,
> 
> Im new to freeswitch and have to ask, how can I enable the portal?
> I installed the debian installation and followed the instruction from wiki to 
> enable:
> https://wiki.freeswitch.org/wiki/Freeswitch_Portal
> It is based on mod_xml_rpc, the module is built by default but not loaded, so 
> you just need to load it (un-comment it in conf/autoload_configs/
> modules.conf.xml) 
> 
> load mod_xml_rpc
> 
> When I trying to access
> ip:8080/portal/index.html
> after login Im getting:
> error 404
> 
> What Im missing?
> 
> Can somone help?
> 
> Tia
> Stefan



From mike at jerris.com  Mon Jun 19 18:10:56 2017
From: mike at jerris.com (Michael Jerris)
Date: Mon, 19 Jun 2017 14:10:56 -0400
Subject: [Freeswitch-users] timer not properly configured
In-Reply-To: <CAKbxfG-MJFhV+wkmd9suzMbvSnHLJ_V8YeX1cn7=KHQJvzx7EA@mail.gmail.com>
References: <70e56fd8-4c40-53e1-2de2-bd8b879daad2@madovsky.org>
 <CAKbxfG-MJFhV+wkmd9suzMbvSnHLJ_V8YeX1cn7=KHQJvzx7EA@mail.gmail.com>
Message-ID: <EAF798C4-E893-4B90-9AA0-963F98E2A8AF@jerris.com>

FS-10405, fixed in master.  Its just cosmetic.

> On Jun 17, 2017, at 1:29 PM, Anthony Minessale <anthony.minessale at gmail.com> wrote:
> 
> Jira jira jira 
> On Sat, Jun 17, 2017 at 7:55 AM Madovsky <infos at madovsky.org <mailto:infos at madovsky.org>> wrote:
> Hi all,
> 
> last today git gives
> 
> switch_core_timer.c:117 Timer is not properly configured
> 
> everytime a call is hangup.
> 
> show timer gives
> 
> type,name,ikey
> timer,soft,CORE_SOFTTIMER_MODULE
> 
> 1 total.
> 
> 
> Thanks
> 
> F
> 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170619/b233e8ef/attachment.html>

From mike at jerris.com  Mon Jun 19 18:19:26 2017
From: mike at jerris.com (Michael Jerris)
Date: Mon, 19 Jun 2017 14:19:26 -0400
Subject: [Freeswitch-users] Debian Stretch
In-Reply-To: <CAO_GGArQoE0gBVHzOFqrvRg6nC6ZDRdf-_Dgp8pUitoNLL7EVA@mail.gmail.com>
References: <CAKDN4FeqnOGn+45=5GXoNM8xqmks66ew8uM0GX4JiFEVrqBd+w@mail.gmail.com>
 <CF850B37-D178-46C8-8AE6-6BB4CF6C4DF6@jerris.com>
 <CAKDN4Ffs7g4cGAqr-YJrO9Cj=p9D4jV0fxrUnmgZ00hcV1bD0w@mail.gmail.com>
 <733A7BE7-6E66-4BD8-8736-B6DDBFE695A4@jerris.com>
 <CAO_GGArQoE0gBVHzOFqrvRg6nC6ZDRdf-_Dgp8pUitoNLL7EVA@mail.gmail.com>
Message-ID: <9BCA607F-C887-4E6C-8A0E-DD4A7305D214@jerris.com>

Of course, I’m excited for that too… and as soon as they are ready we will let you know

> On Jun 18, 2017, at 2:59 PM, Volodymyr Fedorov <lexxua at gmail.com> wrote:
> 
> Hi Michael,
> so from today Stretch is current stable it will be really cool to have packages from freeswitch repository .
> 
> Thanks!
> 
> On Tue, Jun 13, 2017 at 4:14 AM, Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>> wrote:
> announcements will come out when we have real dates.
> 
>> On Jun 12, 2017, at 9:32 PM, Peter Rex <prex5609 at gmail.com <mailto:prex5609 at gmail.com>> wrote:
>> 
>> Thanks Michael. Hate to do this to you, but is there an estimate on 1.8 timeframe? Mailing list shows people were talking about configs and feature requests in January, but can't see much else. Maybe I'm not looking in the right place.
>> 
>> On Mon, Jun 12, 2017 at 6:44 PM, Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>> wrote:
>> Stretch won’t build yet.  I’ll have some patches over the next few weeks to fix that.  1.8 when released will likely target Stretch as its primary but still a bunch of testing to do.  The patches to fix build for stretch will go back into 1.6 branch, once they are complete and tested.
>> 
>> 
>>> On Jun 12, 2017, at 8:28 PM, Peter Rex <prex5609 at gmail.com <mailto:prex5609 at gmail.com>> wrote:
>>> 
>>> Stretch is the new stable on Saturday. I've looked through Confluence and the mailing lists but I can't find anything relevant. I see interesting possibilities at http://files.freeswitch.org/repo/deb <http://files.freeswitch.org/repo/deb>, but I thought I would ask the mailing list if there's a plan yet to add or move the _production_ build to Stretch.
> 
> 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170619/d4dffaef/attachment-0001.html>

From mike at jerris.com  Mon Jun 19 18:27:08 2017
From: mike at jerris.com (Michael Jerris)
Date: Mon, 19 Jun 2017 14:27:08 -0400
Subject: [Freeswitch-users] Callcenter module,
 can I originate call for an agent in uuid-standby mode ?
In-Reply-To: <1497890891.572062.1014326656.673B64EF@webmail.messagingengine.com>
References: <CAEK98WarT_sPVc4HHZoVJqo+BEpGDduwH87w-Ar3C4NphMEwwg@mail.gmail.com>
 <4770F7F5-BCBB-47B6-A909-263EEFEADFE4@jerris.com>
 <1497890891.572062.1014326656.673B64EF@webmail.messagingengine.com>
Message-ID: <A1F308C2-EBD6-4593-A181-F7D948279355@jerris.com>

mod_fifo has ALWAYS been superior, people assume otherwise because of the name.  Check it out, its pretty powerful.  mod_callcenter was written because people had a hard time understanding mod_fifo.  It supports agent tracking, some skills routing, inbound and outbound agents, etc.  If there is stuff missing we should really sort getting it into mod_fifo and abandon mod_callcenter.


> On Jun 19, 2017, at 12:48 PM, Michael Avers <michael at mailworks.org> wrote:
> 
> What makes mod_fifo better these days? Some reason I was under the impression mod_callcenter is a better choice. Can you please give some real world examples where mod_fifo excels compared to mod_callcenter, or features that are not possible to implement with the latter?
> 
> Thanks
> Mike
> 
> On Mon, Jun 19, 2017, at 09:34 AM, Michael Jerris wrote:
>> mod_fifo is a much more feature rich version of a call queue than mod_callcenter.  You might want to check that out instead.
>> 


From agubbe at gmail.com  Tue Jun 20 08:45:42 2017
From: agubbe at gmail.com (=?UTF-8?Q?Agust=C3=AD_Ubalde_Bellot?=)
Date: Tue, 20 Jun 2017 10:45:42 +0200
Subject: [Freeswitch-users] Freeswitch sslv3 support
In-Reply-To: <CAGCz3RCjv0GHRkYmaPd31QLjd6pWBTdgxbbBmdeVRYwz0vU=RA@mail.gmail.com>
References: <CAGCz3RBiBhE7ism+Wfr81HShk42TDht4c-Yp_DY0cp35OCNqLQ@mail.gmail.com>
 <CAGCz3RCjv0GHRkYmaPd31QLjd6pWBTdgxbbBmdeVRYwz0vU=RA@mail.gmail.com>
Message-ID: <CAGCz3RBuuhP=BCE3U69sVea2tHvLNUqWinzU2USTPhY2Fq8cqw@mail.gmail.com>

Hi Michael,

I have performed several connection tests forcing the sslv3 protocol over
secure web sockets and the connection is established. Instead, the same
test connecting to the TLS listening port, the connection is not set. The
protocol is successfully disabled in the configuration.
The version of FreeSWITCH I'm testing is 1.5.14. Is there any way to prove
that the sslv3 protocol is actually disabled in this release?


Thanks,
Agustí

2017-06-15 10:07 GMT+02:00 Agustí Ubalde Bellot <agubbe at gmail.com>:

> Hi Brian,
>
> Is possible to disable for web socket secure connections too?
>
>
> Thanks,
> Agustí
>
> 2017-06-13 13:24 GMT+02:00 Agustí Ubalde Bellot <agubbe at gmail.com>:
>
>> Hi all,
>>
>> Is there a FreeSWITCH update where sslv3 support is disabled?
>>
>>
>> Thanks,
>> Agustí
>>
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170620/23af47ea/attachment.html>

From italo at freeswitch.org  Tue Jun 20 12:45:22 2017
From: italo at freeswitch.org (=?UTF-8?B?w410YWxvIFJvc3Np?=)
Date: Tue, 20 Jun 2017 09:45:22 -0300
Subject: [Freeswitch-users] Calling on the community for Bug Marshals
In-Reply-To: <CAEJMVkChkspixRypT0rMCzawf4F14Tn3vUzYFzN-qtiqVmRnrg@mail.gmail.com>
References: <CAEJMVkChkspixRypT0rMCzawf4F14Tn3vUzYFzN-qtiqVmRnrg@mail.gmail.com>
Message-ID: <CABMX_w-Ts8x3SPP5=a+dm+w4vq8g4jFsQVhrXWbuHL5=8XR2zw@mail.gmail.com>

This is a great opportunity to learn and to be an expert in FreeSWITCH.
This was how I learn a lot!

:-)

On Mon, Jun 5, 2017 at 4:39 PM, Brian West <brian at freeswitch.org> wrote:

> FreeSWITCHers,
>
> We are in need of a few good bug marshals, We are trying to get 1.8 ready
> and out the door and the more help we have testing and working thru patches
> on JIRA the quicker it will arrive.  If you're interested in helping us out
> email me directly.  We are also considering bringing back a few days a week
> we are sitting in 888 and helping the community out with issues pending in
> JIRA.
>
> Also we are only about 2600 short on the gofund me for the Allison
> prompts, which will be delivered sometime this week.  ;)  So help us get
> over that last little bit this week.
>
> Thanks,
>
> --
>
> *Brian West*
> brian at freeswitch.org
>
> *Twitter: @FreeSWITCH , @briankwest*
>
> http://www.freeswitchbook.com
> http://www.freeswitchcookbook.com
>
> Book a phone call (CST) <https://freeswitch.com/appointment>
>
> Allison prompts for FreeSWITCH:
>
> *https://www.gofundme.com/allison-prompts-for-freeswitch*
> <https://www.gofundme.com/allison-prompts-for-freeswitch>
>
> Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
> /r/freeswitch <https://www.reddit.com/r/freeswitch>
>
> *T:*+19184209001 <(918)%20420-9001> | *F:*+19184209002 <(918)%20420-9002>
> | *M:*+1918424WEST (9378)
> *Skype:*briankwest
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org
>



-- 
Ítalo Rossi
italo at freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170620/fa663d35/attachment.html>

From raman.chv at gmail.com  Tue Jun 20 13:07:08 2017
From: raman.chv at gmail.com (Ram)
Date: Tue, 20 Jun 2017 18:37:08 +0530
Subject: [Freeswitch-users] Record-routes in NOTIFY
In-Reply-To: <CAEJMVkDCFmvyNX9Xb_zV=fOybZ8dr2apPf9QFE24TPSKiDOQPQ@mail.gmail.com>
References: <CAArNKuejWc2jm-YHesBwG35TpzmhJ45V9hB=-d=hK61NdhV1pQ@mail.gmail.com>
 <CAEJMVkDCFmvyNX9Xb_zV=fOybZ8dr2apPf9QFE24TPSKiDOQPQ@mail.gmail.com>
Message-ID: <CAArNKudxa9dDumew8Z7AG_qd-kaNdo=MZMWE_66bj3SH7wqVJQ@mail.gmail.com>

Raised the issue in JIRA: https://freeswitch.org/jira/browse/FS-10393


On Tue, Jun 13, 2017 at 7:27 PM, Brian West <brian at freeswitch.org> wrote:

> Any bug reports belong on JIRA,  https://freeswitch.org/
> confluence/display/FREESWITCH/Reporting+Bugs+to+JIRA
>
> Thanks,
>
>
> On Tue, Jun 13, 2017 at 3:35 AM, Ram <raman.chv at gmail.com> wrote:
>
>> Hi,
>>
>> Record routes in SUBSCRIBE is not honored in NOTIFY, In my case i am
>> having 3 record routes in SUBSCRIBE, but only one i.e top record route is
>> used for NOTIFY is causing routing issue. I am using freeswitch version
>> 1.6.17 for testing.
>>
>> Following is the trace for SUBSCRIBE and NOTIFY at freeswitch.
>>
>> T 2017/06/13 08:20:14.868294 10.1.30.27:55503 -> 10.2.30.63:5060 [AP]
>> SUBSCRIBE sip:500 at 52.64.221.219:5060;transport=tcp SIP/2.0
>> Record-Route: <sip:10.1.30.27:5060;transport=tcp;ftag=46baee66;lr>
>> Record-Route: <sip:10.1.30.174;transport=tcp;r2=on;lr=on>
>> Record-Route: <sip:10.1.30.178:5061;transport=tls;r2=on;lr=on>
>> Via: SIP/2.0/TCP 10.1.30.199;branch=z9hG4bKc038
>> .62a9c22ab84694b453503a45210a1392.0;i=1
>> Via: SIP/2.0/TCP 10.1.30.174;branch=z9hG4bKc038
>> .8d466a0ae4f1821a3f0d08e6602cdadc.0;i=82
>> Via: SIP/2.0/TLS 10.1.30.146:51890;received=10.
>> 1.30.146;rport=51890;branch=z9hG4bKd1633fda00007
>> From: "RamanTest"<sip:ramantest at freeconf.com>;tag=46baee66
>> To: "sip:500 at freeconf.com"<sip:500 at freeconf.com;transport=tls>;
>> tag=pv4B8Q9XUDtgD
>> Call-ID: 2d118609-1 at 10.1.30.180
>> CSeq: 1805684444 SUBSCRIBE
>> Max-Forwards: 69
>> Contact: "RamanTest"<sip:ramantest at 10.1.30.146:51890;transport=tls>
>> User-Agent: TestConference
>> Event: conference
>> Expires: 3600
>> Allow: INVITE,ACK,BYE,CANCEL,REFER,NOTIFY,OPTIONS,PRACK,UPDATE,INFO
>> ,MESSAGE,SUBSCRIBE,PUBLISH
>> Allow-Events: refer, presence
>> Supported: replaces, timer, gruu, join
>> Date: Tue, 13 Jun 2017 08:24:00 GMT
>> Content-Length: 0
>>
>>
>> ##
>> T 2017/06/13 08:20:14.873026 10.2.30.63:5060 -> 10.1.30.27:55503 [AP]
>> SIP/2.0 202 Accepted
>> Via: SIP/2.0/TCP 10.1.30.199;branch=z9hG4bKc038
>> .62a9c22ab84694b453503a45210a1392.0;i=1;received=10.1.30.27;rport=55503
>> Via: SIP/2.0/TCP 10.1.30.174;branch=z9hG4bKc038
>> .8d466a0ae4f1821a3f0d08e6602cdadc.0;i=82
>> Via: SIP/2.0/TLS 10.1.30.146:51890;received=10.
>> 1.30.146;rport=51890;branch=z9hG4bKd1633fda00007
>> Record-Route: <sip:10.1.30.27:5060;transport=tcp;ftag=46baee66;lr>
>> Record-Route: <sip:10.1.30.174;transport=tcp;r2=on;lr=on>
>> Record-Route: <sip:10.1.30.178:5061;transport=tls;r2=on;lr=on>
>> From: "RamanTest" <sip:ramantest at freeconf.com>;tag=46baee66
>> To: "sip:500 at freeconf.com" <sip:500 at freeconf.com;transpor
>> t=tls>;tag=pv4B8Q9XUDtgD
>> Call-ID: 2d118609-1 at 10.1.30.180
>> CSeq: 1805684444 SUBSCRIBE
>> Contact: <sip:500 at 52.64.221.219:5060;transport=tcp>
>> Expires: 3600
>> User-Agent: FreeSWITCH-mod_sofia/1.6.17~64bit
>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>> Supported: timer, path, replaces
>> Allow-Events: talk, hold, conference, presence, as-feature-event, dialog,
>> line-seize, call-info, sla, include-session-description, presence.winfo,
>> message-summary, refer
>> Subscription-State: active;expires=3600
>> Content-Length: 0
>>
>>
>> ####
>> T 2017/06/13 08:20:15.173591 10.2.30.63:58879 -> 10.1.30.27:5060 [AP]
>> NOTIFY sip:ramantest at 10.1.30.146:51890;transport=tls SIP/2.0
>> Via: SIP/2.0/TCP 52.64.221.219;rport;branch=z9hG4bKvr9Kyp8Fe829g
>> Route: <sip:10.1.30.27:5060>;transport=tcp;ftag=46baee66;lr
>> Record-Route: <sip:10.1.30.27:5060>;transport=tcp;ftag=46baee66;lr
>> Max-Forwards: 70
>> From: "sip:500 at freeconf.com" <sip:500 at freeconf.com;transpor
>> t=tls>;tag=pv4B8Q9XUDtgD;tag=pv4B8Q9XUDtgD
>> To: "RamanTest" <sip:ramantest at freeconf.com>;tag=46baee66
>> Call-ID: 2d118609-1 at 10.1.30.180
>> CSeq: 705660701 NOTIFY
>> Contact: <sip:500 at 52.64.221.219:5060;transport=tcp>;isfocus
>> User-Agent: FreeSWITCH-mod_sofia/1.6.17~64bit
>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>> Supported: timer, path, replaces
>> Event: conference
>> Allow-Events: talk, hold, conference, presence, as-feature-event, dialog,
>> line-seize, call-info, sla, include-session-description, presence.winfo,
>> message-summary, refer
>> Subscription-State: active;expires=3600
>> Content-Type: application/conference-info+xml
>> Content-Length: 1028
>>
>> <?xml version="1.0"?>
>> <conference-info version="1" state="full" xmlns="urn:ietf:params:xml:ns:conference-info"
>> entity="sip:500 at freeconf.com">
>>   <conference-description>
>>     <display-text>FreeSWITCH Conference</display-text>
>>     <conf-uris>
>>       <entry>
>>         <uri>sip:500 at freeconf.com</uri>
>>       </entry>
>>     </conf-uris>
>>   </conference-description>
>>   <conference-state>
>>     <user-count>1</user-count>
>>     <active>true</active>
>>   </conference-state>
>>   <users>
>>     <user state="full" entity="sip:ramantest at freeconf.com">
>>       <display-text>RamanTest</display-text>
>>       <endpoint entity="sip:ramantest at freeconf.com">
>>         <display-text>RamanTest</display-text>
>>         <status>connected</status>
>>         <joining-info>
>>           <when>2017-06-13T08:20:13+00:00</when>
>>         </joining-info>
>>         <media id="2a">
>>           <type>audio</type>
>>           <src-id>4048072604 <(404)%20807-2604></src-id>
>>           <status>sendrecv</status>
>>         </media>
>>       </endpoint>
>>     </user>
>>   </users>
>> </conference-info>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.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
>> http://www.freeswitch.org
>>
>
>
>
> --
>
> *Brian West*
> brian at freeswitch.org
>
> *Twitter: @FreeSWITCH , @briankwest*
>
> http://www.freeswitchbook.com
> http://www.freeswitchcookbook.com
>
> Book a phone call (CST) <https://freeswitch.com/appointment>
>
> Allison prompts for FreeSWITCH:
>
> *https://www.gofundme.com/allison-prompts-for-freeswitch*
> <https://www.gofundme.com/allison-prompts-for-freeswitch>
>
> Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
> /r/freeswitch <https://www.reddit.com/r/freeswitch>
>
> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
> *Skype:*briankwest
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170620/8ccb6e3f/attachment-0001.html>

From stefan at fuhrmann.homedns.org  Mon Jun 19 19:40:44 2017
From: stefan at fuhrmann.homedns.org (Stefan Fuhrmann)
Date: Mon, 19 Jun 2017 21:40:44 +0200
Subject: [Freeswitch-users] enable Portal, error 404
In-Reply-To: <F123F930-D3FB-4635-8AC2-0505589557E4@jerris.com>
References: <8779966.9K6fzknjRZ@stefan-ubu>
 <F123F930-D3FB-4635-8AC2-0505589557E4@jerris.com>
Message-ID: <1794751.l0UOKoQrpU@stefan-ubu>

Am Montag, 19. Juni 2017, 14:10:07 CEST schrieb Michael Jerris:
> bad credentials?

I dont think so:
user: freeswitch
pass: works

that is what I have found in config.
Is that wrong?

Tia
Stefan


From juraj.fabo at gmail.com  Mon Jun 19 22:14:41 2017
From: juraj.fabo at gmail.com (Juraj Fabo)
Date: Tue, 20 Jun 2017 00:14:41 +0200
Subject: [Freeswitch-users]  signaling_status Down with libpri
Message-ID: <CAAqVCyRyxtvDovbCmw-PtihL_h=hpLsZZv4G1d8q9NXgsVsoSQ@mail.gmail.com>

Dear list


I found this  report here and also in JIRA under
https://freeswitch.org/jira/browse/OPENZAP-243 where it was closed as
fixed after using updated wanpipe.

I am facing the very same issue with sangoma card and I am searching
for wanpipe version which would provide referenced fix.

However, most recent wanpipe I can find on the web is from 2016.

Please, would it be possible to provide precise info which
module/driver/library needs to be updated and from where the update
could be downloaded?


Actually I was really trying to followup existing subject from march,
but I suspect that this email will create new thread, for that case I
would like to apologize in advance.

Thank you very much



Juraj


From nabeel.10.ahmed at gmail.com  Tue Jun 20 09:16:13 2017
From: nabeel.10.ahmed at gmail.com (Nabeel Ahmad)
Date: Tue, 20 Jun 2017 14:16:13 +0500
Subject: [Freeswitch-users] Call limit on RTP-IP Address
Message-ID: <CAD_0LmbDQM=ZUo+G_OJnDLhrTCZzs4aYxscFzH-qa9S6FkSGjQ@mail.gmail.com>

Hello all,
If we assign more then one RTP-IP parameter to a sip profile , they are
used in round robin.
Its perfect . I want to know is there a way to use some limit on that ip ?
Say i've 5 ip address listening on box , and i want one concurrent call
limit on each media ip.
How can i set limit on profile level or set from dialplan.
I tried to do but it didn't work .

Any help or advice will be highly appreciated
Thanking all
Nabeel.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170620/34e4ca57/attachment.html>

From matt at supportedbusiness.com  Tue Jun 20 13:47:42 2017
From: matt at supportedbusiness.com (Matt Broad)
Date: Tue, 20 Jun 2017 14:47:42 +0100
Subject: [Freeswitch-users] group_confirm_file multiple files
In-Reply-To: <F2F379FA-CE02-4750-8EBF-16D1E3FF16A3@jerris.com>
References: <CAEeV61jMv9ytvzX_8URn+M7wiC=vpW0+3Nr7VjbE-Ld8jvR5eA@mail.gmail.com>
 <F2F379FA-CE02-4750-8EBF-16D1E3FF16A3@jerris.com>
Message-ID: <CAEeV61hF0=uDDFe2MvgLONsaun=dYROzLRwW-o4862VQMn0CNA@mail.gmail.com>

Thanks Michael!

I had looked at mod_file_string before but had overlooked file_string://

I was setting playback_delimiter=! which was not working.

For anyone trying to set this using js I have included an example below

session.execute("set",
"group_confirm_file=file_string://file1.mp3!file2.mp3");
session.execute("set", "group_confirm_key=5");

thanks
Matt


Matt Broad
Tel: +44 (0)203 011 1313 <+44%2020%203011%201313>
Web: www.supportedbusiness.com

On 19 June 2017 at 17:46, Michael Jerris <mike at jerris.com> wrote:

> https://freeswitch.org/confluence/display/FREESWITCH/
> mod_dptools%3A+file_string
> <https://freeswitch.org/confluence/display/FREESWITCH/mod_dptools:+file_string>
>
> On Jun 15, 2017, at 5:54 AM, Matt Broad <matt at supportedbusiness.com>
> wrote:
>
> Hi,
>
> I'm wondering if it is possible to play multiple files using the
> group_confirm_file function.
>
> I have 2 audio files that I would like to play 1 after the other and then
> wait for the confirm key.
>
>
> I have tried using mod_file_string, but get an error "Error from mpg123:
> File access error. (code 22)", I assume this is due to the fact it is
> reading the file string as one file rather than 2 separated by the !
> delimiter.
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170620/cfa28a1a/attachment.html>

From lists at telefaks.de  Tue Jun 20 13:53:51 2017
From: lists at telefaks.de (Peter Steinbach)
Date: Tue, 20 Jun 2017 15:53:51 +0200
Subject: [Freeswitch-users] Call limit on RTP-IP Address
In-Reply-To: <CAD_0LmbDQM=ZUo+G_OJnDLhrTCZzs4aYxscFzH-qa9S6FkSGjQ@mail.gmail.com>
References: <CAD_0LmbDQM=ZUo+G_OJnDLhrTCZzs4aYxscFzH-qa9S6FkSGjQ@mail.gmail.com>
Message-ID: <594928EF.6080308@telefaks.de>

Hello Ahmad,

we do it the following way

  * we have a background job, which periodically (every 1-2 sec)
    connects via esl to fo Freeswitch and gets it's channel
    informations. Then we do some calculations based on e.g. call state,
    domain, profile, ip, other customer information (in our case it's
    multi-tenant) 
  * for each calculation, we set a memcache key to a specific value (a
    new call is allowed / not allowed)
  * then, for a new call, we query the memcache key (mod_memcache) in
    the dialplan and decide, what to do with this call

But - of course - there will also be some other ways to do this,
dependent on your specific goal.

Best regards
Peter

On 06/20/17 11:16, Nabeel Ahmad wrote:
> Hello all,
> If we assign more then one RTP-IP parameter to a sip profile , they
> are used in round robin.
> Its perfect . I want to know is there a way to use some limit on that ip ?
> Say i've 5 ip address listening on box , and i want one concurrent
> call limit on each media ip.
> How can i set limit on profile level or set from dialplan.
> I tried to do but it didn't work .
>
> Any help or advice will be highly appreciated
> Thanking all 
> Nabeel.
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org


-- 
With kind regards
Peter Steinbach 

Telefaks Services GmbH
mailto:lists (att) telefaks.de
Internet: www.telefaks.de

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170620/a34a60f2/attachment.html>

From krice at freeswitch.org  Tue Jun 20 14:05:09 2017
From: krice at freeswitch.org (Ken Rice)
Date: Tue, 20 Jun 2017 09:05:09 -0500
Subject: [Freeswitch-users] Call limit on RTP-IP Address
In-Reply-To: <CAD_0LmbDQM=ZUo+G_OJnDLhrTCZzs4aYxscFzH-qa9S6FkSGjQ@mail.gmail.com>
References: <CAD_0LmbDQM=ZUo+G_OJnDLhrTCZzs4aYxscFzH-qa9S6FkSGjQ@mail.gmail.com>
Message-ID: <9cad01d2e9ce$3ab34e50$b019eaf0$@freeswitch.org>

Theres no way to limit or select which IP/Port combination is used from the available RTP IP/Port Range in the config you have.  The only way to do this would be to create a profile for each IP and then limit the number of calls per profile

 

From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Nabeel Ahmad
Sent: Tuesday, June 20, 2017 4:16 AM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Call limit on RTP-IP Address

 

Hello all,

If we assign more then one RTP-IP parameter to a sip profile , they are used in round robin.

Its perfect . I want to know is there a way to use some limit on that ip ?

Say i've 5 ip address listening on box , and i want one concurrent call limit on each media ip.

How can i set limit on profile level or set from dialplan.

I tried to do but it didn't work .

 

Any help or advice will be highly appreciated

Thanking all 

Nabeel.

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170620/f5f6c1df/attachment-0001.html>

From khamlichi.khalil at gmail.com  Tue Jun 20 14:00:46 2017
From: khamlichi.khalil at gmail.com (Khalil Khamlichi)
Date: Tue, 20 Jun 2017 14:00:46 +0000
Subject: [Freeswitch-users] Callcenter module,
 can I originate call for an agent in uuid-standby mode ?
In-Reply-To: <A1F308C2-EBD6-4593-A181-F7D948279355@jerris.com>
References: <CAEK98WarT_sPVc4HHZoVJqo+BEpGDduwH87w-Ar3C4NphMEwwg@mail.gmail.com>
 <4770F7F5-BCBB-47B6-A909-263EEFEADFE4@jerris.com>
 <1497890891.572062.1014326656.673B64EF@webmail.messagingengine.com>
 <A1F308C2-EBD6-4593-A181-F7D948279355@jerris.com>
Message-ID: <CAEK98WbEuigpyXoMb7-d0HQvUh7aZm_KZQhNrSr-d8pn+ev1iQ@mail.gmail.com>

alright, does mod_fifo support some sort of uuid_standby mode ?

On Mon, Jun 19, 2017 at 6:27 PM, Michael Jerris <mike at jerris.com> wrote:

> mod_fifo has ALWAYS been superior, people assume otherwise because of the
> name.  Check it out, its pretty powerful.  mod_callcenter was written
> because people had a hard time understanding mod_fifo.  It supports agent
> tracking, some skills routing, inbound and outbound agents, etc.  If there
> is stuff missing we should really sort getting it into mod_fifo and abandon
> mod_callcenter.
>
>
> > On Jun 19, 2017, at 12:48 PM, Michael Avers <michael at mailworks.org>
> wrote:
> >
> > What makes mod_fifo better these days? Some reason I was under the
> impression mod_callcenter is a better choice. Can you please give some real
> world examples where mod_fifo excels compared to mod_callcenter, or
> features that are not possible to implement with the latter?
> >
> > Thanks
> > Mike
> >
> > On Mon, Jun 19, 2017, at 09:34 AM, Michael Jerris wrote:
> >> mod_fifo is a much more feature rich version of a call queue than
> mod_callcenter.  You might want to check that out instead.
> >>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170620/4a607804/attachment.html>

From nabeel.10.ahmed at gmail.com  Tue Jun 20 14:11:53 2017
From: nabeel.10.ahmed at gmail.com (Nabeel Ahmad)
Date: Tue, 20 Jun 2017 19:11:53 +0500
Subject: [Freeswitch-users] Call limit on RTP-IP Address
In-Reply-To: <594928EF.6080308@telefaks.de>
References: <CAD_0LmbDQM=ZUo+G_OJnDLhrTCZzs4aYxscFzH-qa9S6FkSGjQ@mail.gmail.com>
 <594928EF.6080308@telefaks.de>
Message-ID: <CAD_0LmaueFi5az=ObP7DwsB2QnL=sPskmDVHbgME+kyM0MKizg@mail.gmail.com>

Hi Peter,

So far i also came to something similar.
i've all ips in db with status enable/disable.
on channel answer event , same ip toggles its status.
then updating profile rtp-ip variable and rescan profile.
and on destroy event again update status in Db and rescan the profile.
If i ignore frequent calls to DB , still i am not getting what will happen
if all ips are used . How will i restrict calls more then IPs i've . (can't
touch global session limit as there are other profiles where i dont want to
limit anything ).
So i thought to ask there must be some better way to do it.

@Ken :
each profile with their own RTP-iP can also work , how to limit number of
calls per profile ?



On Tue, Jun 20, 2017 at 6:53 PM, Peter Steinbach <lists at telefaks.de> wrote:

> Hello Ahmad,
>
> we do it the following way
>
>    - we have a background job, which periodically (every 1-2 sec)
>    connects via esl to fo Freeswitch and gets it's channel informations. Then
>    we do some calculations based on e.g. call state, domain, profile, ip,
>    other customer information (in our case it's multi-tenant)
>    - for each calculation, we set a memcache key to a specific value (a
>    new call is allowed / not allowed)
>    - then, for a new call, we query the memcache key (mod_memcache) in
>    the dialplan and decide, what to do with this call
>
> But - of course - there will also be some other ways to do this, dependent
> on your specific goal.
>
> Best regards
> Peter
>
>
> On 06/20/17 11:16, Nabeel Ahmad wrote:
>
> Hello all,
> If we assign more then one RTP-IP parameter to a sip profile , they are
> used in round robin.
> Its perfect . I want to know is there a way to use some limit on that ip ?
> Say i've 5 ip address listening on box , and i want one concurrent call
> limit on each media ip.
> How can i set limit on profile level or set from dialplan.
> I tried to do but it didn't work .
>
> Any help or advice will be highly appreciated
> Thanking all
> Nabeel.
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>
> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.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-usershttp://www.freeswitch.org
>
>
>
> --
> With kind regards
> Peter Steinbach
>
> Telefaks Services GmbHmailto:lists <lists> (att) telefaks.de
> Internet: www.telefaks.de
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170620/3fa98783/attachment.html>

From me at nevian.org  Tue Jun 20 14:17:24 2017
From: me at nevian.org (Serge S. Yuriev)
Date: Tue, 20 Jun 2017 17:17:24 +0300
Subject: [Freeswitch-users] Call limit on RTP-IP Address
In-Reply-To: <9cad01d2e9ce$3ab34e50$b019eaf0$@freeswitch.org>
References: <CAD_0LmbDQM=ZUo+G_OJnDLhrTCZzs4aYxscFzH-qa9S6FkSGjQ@mail.gmail.com>
 <9cad01d2e9ce$3ab34e50$b019eaf0$@freeswitch.org>
Message-ID: <5a52f788-d3b1-a3fb-4e0d-daeb247eb782@nevian.org>

Is this multi-IP config expected to work this way or it's eventuality/bug?

On 20/06/17 17:05, Ken Rice wrote:
> Theres no way to limit or select which IP/Port combination is used from 
> the available RTP IP/Port Range in the config you have.  The only way to 
> do this would be to create a profile for each IP and then limit the 
> number of calls per profile
> 
> *From:* FreeSWITCH-users 
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of 
> *Nabeel Ahmad
> *Sent:* Tuesday, June 20, 2017 4:16 AM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* [Freeswitch-users] Call limit on RTP-IP Address
> 
> Hello all,
> 
> If we assign more then one RTP-IP parameter to a sip profile , they are 
> used in round robin.
> 
> Its perfect . I want to know is there a way to use some limit on that ip ?
> 
> Say i've 5 ip address listening on box , and i want one concurrent call 
> limit on each media ip.
> 
> How can i set limit on profile level or set from dialplan.
> 
> I tried to do but it didn't work .
> 
> Any help or advice will be highly appreciated
> 
> Thanking all
> 
> Nabeel.


-- 
Serge S. Yuriev
Lead VoIP engineer



From david.villasmil.work at gmail.com  Tue Jun 20 14:30:35 2017
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Tue, 20 Jun 2017 14:30:35 +0000
Subject: [Freeswitch-users] Calling on the community for Bug Marshals
In-Reply-To: <CABMX_w-Ts8x3SPP5=a+dm+w4vq8g4jFsQVhrXWbuHL5=8XR2zw@mail.gmail.com>
References: <CAEJMVkChkspixRypT0rMCzawf4F14Tn3vUzYFzN-qtiqVmRnrg@mail.gmail.com>
 <CABMX_w-Ts8x3SPP5=a+dm+w4vq8g4jFsQVhrXWbuHL5=8XR2zw@mail.gmail.com>
Message-ID: <CAFGRPVpcLXFbHt7YMrSTajyz-xSQSz9hidgRncLYnsM0P0Qevw@mail.gmail.com>

Anything i can help with?
On Tue, Jun 20, 2017 at 2:46 PM Ítalo Rossi <italo at freeswitch.org> wrote:

> This is a great opportunity to learn and to be an expert in FreeSWITCH.
> This was how I learn a lot!
>
> :-)
>
> On Mon, Jun 5, 2017 at 4:39 PM, Brian West <brian at freeswitch.org> wrote:
>
>> FreeSWITCHers,
>>
>> We are in need of a few good bug marshals, We are trying to get 1.8 ready
>> and out the door and the more help we have testing and working thru patches
>> on JIRA the quicker it will arrive.  If you're interested in helping us out
>> email me directly.  We are also considering bringing back a few days a week
>> we are sitting in 888 and helping the community out with issues pending in
>> JIRA.
>>
>> Also we are only about 2600 short on the gofund me for the Allison
>> prompts, which will be delivered sometime this week.  ;)  So help us get
>> over that last little bit this week.
>>
>> Thanks,
>>
>> --
>>
>> *Brian West*
>> brian at freeswitch.org
>>
>> *Twitter: @FreeSWITCH , @briankwest*
>>
>> http://www.freeswitchbook.com
>> http://www.freeswitchcookbook.com
>>
>> Book a phone call (CST) <https://freeswitch.com/appointment>
>>
>> Allison prompts for FreeSWITCH:
>>
>> *https://www.gofundme.com/allison-prompts-for-freeswitch*
>> <https://www.gofundme.com/allison-prompts-for-freeswitch>
>>
>> Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
>> /r/freeswitch <https://www.reddit.com/r/freeswitch>
>>
>> *T:*+19184209001 <(918)%20420-9001> | *F:*+19184209002 <(918)%20420-9002>
>> | *M:*+1918424WEST (9378)
>> *Skype:*briankwest
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.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
>> http://www.freeswitch.org
>>
>
>
>
> --
> Ítalo Rossi
> italo at freeswitch.org
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170620/d9080e93/attachment-0001.html>

From david.villasmil.work at gmail.com  Tue Jun 20 14:31:40 2017
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Tue, 20 Jun 2017 14:31:40 +0000
Subject: [Freeswitch-users] enable Portal, error 404
In-Reply-To: <1794751.l0UOKoQrpU@stefan-ubu>
References: <8779966.9K6fzknjRZ@stefan-ubu>
 <F123F930-D3FB-4635-8AC2-0505589557E4@jerris.com>
 <1794751.l0UOKoQrpU@stefan-ubu>
Message-ID: <CAFGRPVpC4Sp-gviha=JLgn2X5PPxcYbCbvCtq1sv=jziK0UfEw@mail.gmail.com>

Does netstat shows the port in use?
On Tue, Jun 20, 2017 at 3:29 PM Stefan Fuhrmann <stefan at fuhrmann.homedns.org>
wrote:

> Am Montag, 19. Juni 2017, 14:10:07 CEST schrieb Michael Jerris:
> > bad credentials?
>
> I dont think so:
> user: freeswitch
> pass: works
>
> that is what I have found in config.
> Is that wrong?
>
> Tia
> Stefan
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170620/831508b9/attachment.html>

From krice at freeswitch.org  Tue Jun 20 14:40:49 2017
From: krice at freeswitch.org (Ken Rice)
Date: Tue, 20 Jun 2017 09:40:49 -0500
Subject: [Freeswitch-users] Call limit on RTP-IP Address
In-Reply-To: <5a52f788-d3b1-a3fb-4e0d-daeb247eb782@nevian.org>
References: <CAD_0LmbDQM=ZUo+G_OJnDLhrTCZzs4aYxscFzH-qa9S6FkSGjQ@mail.gmail.com>
 <9cad01d2e9ce$3ab34e50$b019eaf0$@freeswitch.org>
 <5a52f788-d3b1-a3fb-4e0d-daeb247eb782@nevian.org>
Message-ID: <9d7b01d2e9d3$366f97e0$a34ec7a0$@freeswitch.org>

This is not a bug... there is just no way to select which IP or port is used nor is there an effective way to limit it short of just killing the call due to the way the IP/Port allocation works in the RTP stack currently.

-----Original Message-----
From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Serge S. Yuriev
Sent: Tuesday, June 20, 2017 9:17 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Call limit on RTP-IP Address

Is this multi-IP config expected to work this way or it's eventuality/bug?

On 20/06/17 17:05, Ken Rice wrote:
> Theres no way to limit or select which IP/Port combination is used 
> from the available RTP IP/Port Range in the config you have.  The only 
> way to do this would be to create a profile for each IP and then limit 
> the number of calls per profile
> 
> *From:* FreeSWITCH-users
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of 
> *Nabeel Ahmad
> *Sent:* Tuesday, June 20, 2017 4:16 AM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* [Freeswitch-users] Call limit on RTP-IP Address
> 
> Hello all,
> 
> If we assign more then one RTP-IP parameter to a sip profile , they 
> are used in round robin.
> 
> Its perfect . I want to know is there a way to use some limit on that ip ?
> 
> Say i've 5 ip address listening on box , and i want one concurrent 
> call limit on each media ip.
> 
> How can i set limit on profile level or set from dialplan.
> 
> I tried to do but it didn't work .
> 
> Any help or advice will be highly appreciated
> 
> Thanking all
> 
> Nabeel.


--
Serge S. Yuriev
Lead VoIP engineer


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.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
http://www.freeswitch.org



From nabeel.10.ahmed at gmail.com  Tue Jun 20 14:20:46 2017
From: nabeel.10.ahmed at gmail.com (Nabeel Ahmad)
Date: Tue, 20 Jun 2017 19:20:46 +0500
Subject: [Freeswitch-users] Call limit on RTP-IP Address
In-Reply-To: <5a52f788-d3b1-a3fb-4e0d-daeb247eb782@nevian.org>
References: <CAD_0LmbDQM=ZUo+G_OJnDLhrTCZzs4aYxscFzH-qa9S6FkSGjQ@mail.gmail.com>
 <9cad01d2e9ce$3ab34e50$b019eaf0$@freeswitch.org>
 <5a52f788-d3b1-a3fb-4e0d-daeb247eb782@nevian.org>
Message-ID: <CAD_0Lmb+LGsiOMTBsJSiSdqA3iRR8HaFnDBhCscMfwDUPXEp7w@mail.gmail.com>

Its never expected to work this WAY Nor its a Bug.
Its a kind of feature request.
As FS is a flexible product which allows to do things in many ways , trying
to find better approach to do it.


On Tue, Jun 20, 2017 at 7:17 PM, Serge S. Yuriev <me at nevian.org> wrote:

> Is this multi-IP config expected to work this way or it's eventuality/bug?
>
> On 20/06/17 17:05, Ken Rice wrote:
>
>> Theres no way to limit or select which IP/Port combination is used from
>> the available RTP IP/Port Range in the config you have.  The only way to do
>> this would be to create a profile for each IP and then limit the number of
>> calls per profile
>>
>> *From:* FreeSWITCH-users [mailto:freeswitch-users-bounc
>> es at lists.freeswitch.org] *On Behalf Of *Nabeel Ahmad
>> *Sent:* Tuesday, June 20, 2017 4:16 AM
>> *To:* freeswitch-users at lists.freeswitch.org
>> *Subject:* [Freeswitch-users] Call limit on RTP-IP Address
>>
>> Hello all,
>>
>> If we assign more then one RTP-IP parameter to a sip profile , they are
>> used in round robin.
>>
>> Its perfect . I want to know is there a way to use some limit on that ip ?
>>
>> Say i've 5 ip address listening on box , and i want one concurrent call
>> limit on each media ip.
>>
>> How can i set limit on profile level or set from dialplan.
>>
>> I tried to do but it didn't work .
>>
>> Any help or advice will be highly appreciated
>>
>> Thanking all
>>
>> Nabeel.
>>
>
>
> --
> Serge S. Yuriev
> Lead VoIP engineer
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170620/c81f48a5/attachment.html>

From krice at freeswitch.org  Tue Jun 20 14:42:07 2017
From: krice at freeswitch.org (Ken Rice)
Date: Tue, 20 Jun 2017 09:42:07 -0500
Subject: [Freeswitch-users] Call limit on RTP-IP Address
In-Reply-To: <CAD_0LmaueFi5az=ObP7DwsB2QnL=sPskmDVHbgME+kyM0MKizg@mail.gmail.com>
References: <CAD_0LmbDQM=ZUo+G_OJnDLhrTCZzs4aYxscFzH-qa9S6FkSGjQ@mail.gmail.com>
 <594928EF.6080308@telefaks.de>
 <CAD_0LmaueFi5az=ObP7DwsB2QnL=sPskmDVHbgME+kyM0MKizg@mail.gmail.com>
Message-ID: <9d7c01d2e9d3$64cc0830$2e641890$@freeswitch.org>

https://freeswitch.org/confluence/display/FREESWITCH/mod_dptools%3A+Limit

 

once you have it on a profile you can limit on the profile name using the built in limit framework… 

 

this doesn’t work with the multi-rtpip setup as there is no way to indicate which one is to be used before calling bridge

 

From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Nabeel Ahmad
Sent: Tuesday, June 20, 2017 9:12 AM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Call limit on RTP-IP Address

 

Hi Peter,


So far i also came to something similar.

i've all ips in db with status enable/disable.

on channel answer event , same ip toggles its status.

then updating profile rtp-ip variable and rescan profile.

and on destroy event again update status in Db and rescan the profile.

If i ignore frequent calls to DB , still i am not getting what will happen if all ips are used . How will i restrict calls more then IPs i've . (can't touch global session limit as there are other profiles where i dont want to limit anything ).

So i thought to ask there must be some better way to do it.

@Ken : 

each profile with their own RTP-iP can also work , how to limit number of calls per profile ?  

 

 

 

On Tue, Jun 20, 2017 at 6:53 PM, Peter Steinbach <lists at telefaks.de <mailto:lists at telefaks.de> > wrote:

Hello Ahmad,

we do it the following way

*	we have a background job, which periodically (every 1-2 sec) connects via esl to fo Freeswitch and gets it's channel informations. Then we do some calculations based on e.g. call state, domain, profile, ip, other customer information (in our case it's multi-tenant) 
*	for each calculation, we set a memcache key to a specific value (a new call is allowed / not allowed)
*	then, for a new call, we query the memcache key (mod_memcache) in the dialplan and decide, what to do with this call 

But - of course - there will also be some other ways to do this, dependent on your specific goal.

Best regards
Peter



On 06/20/17 11:16, Nabeel Ahmad wrote:

Hello all, 

If we assign more then one RTP-IP parameter to a sip profile , they are used in round robin.

Its perfect . I want to know is there a way to use some limit on that ip ?

Say i've 5 ip address listening on box , and i want one concurrent call limit on each media ip.

How can i set limit on profile level or set from dialplan.

I tried to do but it didn't work .

 

Any help or advice will be highly appreciated

Thanking all 

Nabeel.

 

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org <mailto:consulting at freeswitch.org> 
http://www.freeswitchsolutions.com
 
Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.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
http://www.freeswitch.org






-- 
With kind regards
Peter Steinbach 
 
Telefaks Services GmbH
mailto:lists (att) telefaks.de <http://telefaks.de> 
Internet: www.telefaks.de <http://www.telefaks.de> 
 


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org <mailto:consulting at freeswitch.org> 
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.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
http://www.freeswitch.org

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170620/5e075fdf/attachment-0001.html>

From achinthau at gmail.com  Tue Jun 20 15:05:06 2017
From: achinthau at gmail.com (Achintha)
Date: Tue, 20 Jun 2017 20:35:06 +0530
Subject: [Freeswitch-users] switch_core_sqldb.c:2987 invalied cdr data,
	call not recoverd
Message-ID: <CA+Jwzxv0dqLGraawMjE-u7C943uk0Hoxo6WEgh0kh7o6J7FdPQ@mail.gmail.com>

hi all,

I configured two freeswitch servers (freeswitch 1.6.18 on debian 8 ) with
call recovery feature. It is working properly on extension to extension ,
out bound and IVR Calls.
Then i tried to land an incoming call, the call gets connected to the queue
and then freeswitch generate a call to agent and bridge it with queued call
and both sides can hear properly.
Then i crashed the primary server, call got landed to the second freeswitch
server but rtp is not functioning, and in the second freeswitch console, it
printed  "switch_core_sqldb.c:2987 invalied cdr data, call not recovered".
But the call was not disconnected.

I used the following configurations

*switch.conf.xml : *

                         switch name is same on both servers
                         core-db-dsn and core-recovery-db-dsn  configured
with pgsql
*both sip profiles:*
                        odbc-dsn configured with pgsql
                        track-calls elabled

please provide me a solution to sort out this.

-- 
Best Regards..
Achintha Udukumbura
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170620/864aed38/attachment.html>

From me at nevian.org  Tue Jun 20 15:35:14 2017
From: me at nevian.org (Serge S. Yuriev)
Date: Tue, 20 Jun 2017 18:35:14 +0300
Subject: [Freeswitch-users] Call limit on RTP-IP Address
In-Reply-To: <9d7b01d2e9d3$366f97e0$a34ec7a0$@freeswitch.org>
References: <CAD_0LmbDQM=ZUo+G_OJnDLhrTCZzs4aYxscFzH-qa9S6FkSGjQ@mail.gmail.com>
 <9cad01d2e9ce$3ab34e50$b019eaf0$@freeswitch.org>
 <5a52f788-d3b1-a3fb-4e0d-daeb247eb782@nevian.org>
 <9d7b01d2e9d3$366f97e0$a34ec7a0$@freeswitch.org>
Message-ID: <ac831996-c793-c1ba-48c1-b8d2017eaf63@nevian.org>

I mean if we can define multiple IPs at all. I was under impression 
parser uses only one last defined.

On 20/06/17 17:40, Ken Rice wrote:
> This is not a bug... there is just no way to select which IP or port is used nor is there an effective way to limit it short of just killing the call due to the way the IP/Port allocation works in the RTP stack currently.
> 
> -----Original Message-----
> From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Serge S. Yuriev
> Sent: Tuesday, June 20, 2017 9:17 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Call limit on RTP-IP Address
> 
> Is this multi-IP config expected to work this way or it's eventuality/bug?
> 
> On 20/06/17 17:05, Ken Rice wrote:
>> Theres no way to limit or select which IP/Port combination is used
>> from the available RTP IP/Port Range in the config you have.  The only
>> way to do this would be to create a profile for each IP and then limit
>> the number of calls per profile
>>
>> *From:* FreeSWITCH-users
>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of
>> *Nabeel Ahmad
>> *Sent:* Tuesday, June 20, 2017 4:16 AM
>> *To:* freeswitch-users at lists.freeswitch.org
>> *Subject:* [Freeswitch-users] Call limit on RTP-IP Address
>>
>> Hello all,
>>
>> If we assign more then one RTP-IP parameter to a sip profile , they
>> are used in round robin.
>>
>> Its perfect . I want to know is there a way to use some limit on that ip ?
>>
>> Say i've 5 ip address listening on box , and i want one concurrent
>> call limit on each media ip.
>>
>> How can i set limit on profile level or set from dialplan.
>>
>> I tried to do but it didn't work .
>>
>> Any help or advice will be highly appreciated
>>
>> Thanking all
>>
>> Nabeel.
> 
> 
> --
> Serge S. Yuriev
> Lead VoIP engineer
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org
> 

-- 
Serge S. Yuriev
Lead VoIP engineer



From mike at jerris.com  Tue Jun 20 15:41:23 2017
From: mike at jerris.com (Michael Jerris)
Date: Tue, 20 Jun 2017 11:41:23 -0400
Subject: [Freeswitch-users] Freeswitch sslv3 support
In-Reply-To: <CAGCz3RBuuhP=BCE3U69sVea2tHvLNUqWinzU2USTPhY2Fq8cqw@mail.gmail.com>
References: <CAGCz3RBiBhE7ism+Wfr81HShk42TDht4c-Yp_DY0cp35OCNqLQ@mail.gmail.com>
 <CAGCz3RCjv0GHRkYmaPd31QLjd6pWBTdgxbbBmdeVRYwz0vU=RA@mail.gmail.com>
 <CAGCz3RBuuhP=BCE3U69sVea2tHvLNUqWinzU2USTPhY2Fq8cqw@mail.gmail.com>
Message-ID: <E5E088B6-5FC4-47E3-BC84-D07B40552EF7@jerris.com>

Wait… you are testing against some ancient dev version and not current release?  Is that a typo?  If not, this makes no sense at all, please explain.

> On Jun 20, 2017, at 4:45 AM, Agustí Ubalde Bellot <agubbe at gmail.com> wrote:
> 
> Hi Michael,
> 
> I have performed several connection tests forcing the sslv3 protocol over secure web sockets and the connection is established. Instead, the same test connecting to the TLS listening port, the connection is not set. The protocol is successfully disabled in the configuration.
> The version of FreeSWITCH I'm testing is 1.5.14. Is there any way to prove that the sslv3 protocol is actually disabled in this release?
> 
> 
> Thanks,
> Agustí
> 
> 2017-06-15 10:07 GMT+02:00 Agustí Ubalde Bellot <agubbe at gmail.com <mailto:agubbe at gmail.com>>:
> Hi Brian,
> 
> Is possible to disable for web socket secure connections too?
> 
> 
> Thanks,
> Agustí
> 
> 2017-06-13 13:24 GMT+02:00 Agustí Ubalde Bellot <agubbe at gmail.com <mailto:agubbe at gmail.com>>:
> Hi all,
> 
> Is there a FreeSWITCH update where sslv3 support is disabled?
> 
> 
> Thanks,
> Agustí
> 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170620/9aba97df/attachment.html>

From mike at jerris.com  Tue Jun 20 15:42:38 2017
From: mike at jerris.com (Michael Jerris)
Date: Tue, 20 Jun 2017 11:42:38 -0400
Subject: [Freeswitch-users] signaling_status Down with libpri
In-Reply-To: <CAAqVCyRyxtvDovbCmw-PtihL_h=hpLsZZv4G1d8q9NXgsVsoSQ@mail.gmail.com>
References: <CAAqVCyRyxtvDovbCmw-PtihL_h=hpLsZZv4G1d8q9NXgsVsoSQ@mail.gmail.com>
Message-ID: <59FD8240-CD8B-48FC-A165-819FBE5C9CA8@jerris.com>

If you are looking for help setting up sangoma software, please contact sangoma support.  They provide their own support for their hardware and drivers.

> On Jun 19, 2017, at 6:14 PM, Juraj Fabo <juraj.fabo at gmail.com> wrote:
> 
> Dear list
> 
> 
> I found this  report here and also in JIRA under
> https://freeswitch.org/jira/browse/OPENZAP-243 where it was closed as
> fixed after using updated wanpipe.
> 
> I am facing the very same issue with sangoma card and I am searching
> for wanpipe version which would provide referenced fix.
> 
> However, most recent wanpipe I can find on the web is from 2016.
> 
> Please, would it be possible to provide precise info which
> module/driver/library needs to be updated and from where the update
> could be downloaded?
> 
> 
> Actually I was really trying to followup existing subject from march,
> but I suspect that this email will create new thread, for that case I
> would like to apologize in advance.
> 
> Thank you very much



From mike at jerris.com  Tue Jun 20 15:43:29 2017
From: mike at jerris.com (Michael Jerris)
Date: Tue, 20 Jun 2017 11:43:29 -0400
Subject: [Freeswitch-users] Callcenter module,
 can I originate call for an agent in uuid-standby mode ?
In-Reply-To: <CAEK98WbEuigpyXoMb7-d0HQvUh7aZm_KZQhNrSr-d8pn+ev1iQ@mail.gmail.com>
References: <CAEK98WarT_sPVc4HHZoVJqo+BEpGDduwH87w-Ar3C4NphMEwwg@mail.gmail.com>
 <4770F7F5-BCBB-47B6-A909-263EEFEADFE4@jerris.com>
 <1497890891.572062.1014326656.673B64EF@webmail.messagingengine.com>
 <A1F308C2-EBD6-4593-A181-F7D948279355@jerris.com>
 <CAEK98WbEuigpyXoMb7-d0HQvUh7aZm_KZQhNrSr-d8pn+ev1iQ@mail.gmail.com>
Message-ID: <8C4372DB-F2EB-4614-82EC-185B5AA33C65@jerris.com>

Not sure what that means exactly.

> On Jun 20, 2017, at 10:00 AM, Khalil Khamlichi <khamlichi.khalil at gmail.com> wrote:
> 
> alright, does mod_fifo support some sort of uuid_standby mode ? 
> 
> On Mon, Jun 19, 2017 at 6:27 PM, Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>> wrote:
> mod_fifo has ALWAYS been superior, people assume otherwise because of the name.  Check it out, its pretty powerful.  mod_callcenter was written because people had a hard time understanding mod_fifo.  It supports agent tracking, some skills routing, inbound and outbound agents, etc.  If there is stuff missing we should really sort getting it into mod_fifo and abandon mod_callcenter.
> 
> 
> > On Jun 19, 2017, at 12:48 PM, Michael Avers <michael at mailworks.org <mailto:michael at mailworks.org>> wrote:
> >
> > What makes mod_fifo better these days? Some reason I was under the impression mod_callcenter is a better choice. Can you please give some real world examples where mod_fifo excels compared to mod_callcenter, or features that are not possible to implement with the latter?
> >
> > Thanks
> > Mike
> >
> > On Mon, Jun 19, 2017, at 09:34 AM, Michael Jerris wrote:
> >> mod_fifo is a much more feature rich version of a call queue than mod_callcenter.  You might want to check that out instead.
> >>

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170620/5fe01813/attachment-0001.html>

From mike at jerris.com  Tue Jun 20 15:44:59 2017
From: mike at jerris.com (Michael Jerris)
Date: Tue, 20 Jun 2017 11:44:59 -0400
Subject: [Freeswitch-users] switch_core_sqldb.c:2987 invalied cdr data,
 call not recoverd
In-Reply-To: <CA+Jwzxv0dqLGraawMjE-u7C943uk0Hoxo6WEgh0kh7o6J7FdPQ@mail.gmail.com>
References: <CA+Jwzxv0dqLGraawMjE-u7C943uk0Hoxo6WEgh0kh7o6J7FdPQ@mail.gmail.com>
Message-ID: <5A5DBC29-DCE6-417F-8B48-2BE9A3390523@jerris.com>

would need a bug report on this one with full logs and config and how to reproduce to look into it.

> On Jun 20, 2017, at 11:05 AM, Achintha <achinthau at gmail.com> wrote:
> 
> hi all,
> 
> I configured two freeswitch servers (freeswitch 1.6.18 on debian 8 ) with call recovery feature. It is working properly on extension to extension , out bound and IVR Calls.
> Then i tried to land an incoming call, the call gets connected to the queue and then freeswitch generate a call to agent and bridge it with queued call and both sides can hear properly.
> Then i crashed the primary server, call got landed to the second freeswitch server but rtp is not functioning, and in the second freeswitch console, it printed  "switch_core_sqldb.c:2987 invalied cdr data, call not recovered". But the call was not disconnected.
> 
> I used the following configurations
> 
> switch.conf.xml : 
> 
>                          switch name is same on both servers
>                          core-db-dsn and core-recovery-db-dsn  configured with pgsql
> both sip profiles:
>                         odbc-dsn configured with pgsql
>                         track-calls elabled
> 
> please provide me a solution to sort out this.
> 
> -- 
> Best Regards..
> Achintha Udukumbura
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170620/c1007fa0/attachment.html>

From krice at freeswitch.org  Tue Jun 20 15:52:29 2017
From: krice at freeswitch.org (Ken Rice)
Date: Tue, 20 Jun 2017 10:52:29 -0500
Subject: [Freeswitch-users] Call limit on RTP-IP Address
In-Reply-To: <ac831996-c793-c1ba-48c1-b8d2017eaf63@nevian.org>
References: <CAD_0LmbDQM=ZUo+G_OJnDLhrTCZzs4aYxscFzH-qa9S6FkSGjQ@mail.gmail.com>
 <9cad01d2e9ce$3ab34e50$b019eaf0$@freeswitch.org>
 <5a52f788-d3b1-a3fb-4e0d-daeb247eb782@nevian.org>
 <9d7b01d2e9d3$366f97e0$a34ec7a0$@freeswitch.org>
 <ac831996-c793-c1ba-48c1-b8d2017eaf63@nevian.org>
Message-ID: <9e2901d2e9dd$391c0eb0$ab542c10$@freeswitch.org>

No Defining multiple RTP IPs has been there for a while.... you don’t define them in different lines, the parser will filter out previous ones, you define them all together

    <param name="rtp-ip" value1.2.3.4,1.2.3.5,1.2.3.6,1.2.3.7"/>

The stack will then round robin them.  This feature was added several years ago so that FreeSWITCH can handle the required RTP load in traffic flows that can exceed that of 1Gig-E network connections while only using 1 SIP Profile for traffic. It's still useful although less of a requirement with 10GigE coming down in price. (you can now find managed 48port 10GE switches, NICs and Cables on the secondary market for a combined cost under $200/port now.


-----Original Message-----
From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Serge S. Yuriev
Sent: Tuesday, June 20, 2017 10:35 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Call limit on RTP-IP Address

I mean if we can define multiple IPs at all. I was under impression parser uses only one last defined.

On 20/06/17 17:40, Ken Rice wrote:
> This is not a bug... there is just no way to select which IP or port is used nor is there an effective way to limit it short of just killing the call due to the way the IP/Port allocation works in the RTP stack currently.
> 
> -----Original Message-----
> From: FreeSWITCH-users 
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of 
> Serge S. Yuriev
> Sent: Tuesday, June 20, 2017 9:17 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Call limit on RTP-IP Address
> 
> Is this multi-IP config expected to work this way or it's eventuality/bug?
> 
> On 20/06/17 17:05, Ken Rice wrote:
>> Theres no way to limit or select which IP/Port combination is used 
>> from the available RTP IP/Port Range in the config you have.  The 
>> only way to do this would be to create a profile for each IP and then 
>> limit the number of calls per profile
>>
>> *From:* FreeSWITCH-users
>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of 
>> *Nabeel Ahmad
>> *Sent:* Tuesday, June 20, 2017 4:16 AM
>> *To:* freeswitch-users at lists.freeswitch.org
>> *Subject:* [Freeswitch-users] Call limit on RTP-IP Address
>>
>> Hello all,
>>
>> If we assign more then one RTP-IP parameter to a sip profile , they 
>> are used in round robin.
>>
>> Its perfect . I want to know is there a way to use some limit on that ip ?
>>
>> Say i've 5 ip address listening on box , and i want one concurrent 
>> call limit on each media ip.
>>
>> How can i set limit on profile level or set from dialplan.
>>
>> I tried to do but it didn't work .
>>
>> Any help or advice will be highly appreciated
>>
>> Thanking all
>>
>> Nabeel.
> 
> 
> --
> Serge S. Yuriev
> Lead VoIP engineer
> 
> 
> ______________________________________________________________________
> ___ Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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-use
> rs
> http://www.freeswitch.org
> 
> 
> ______________________________________________________________________
> ___ Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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-use
> rs
> http://www.freeswitch.org
> 

--
Serge S. Yuriev
Lead VoIP engineer


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.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
http://www.freeswitch.org



From agubbe at gmail.com  Tue Jun 20 15:55:58 2017
From: agubbe at gmail.com (=?UTF-8?Q?Agust=C3=AD_Ubalde_Bellot?=)
Date: Tue, 20 Jun 2017 17:55:58 +0200
Subject: [Freeswitch-users] Freeswitch sslv3 support
In-Reply-To: <CAGCz3RBuuhP=BCE3U69sVea2tHvLNUqWinzU2USTPhY2Fq8cqw@mail.gmail.com>
References: <CAGCz3RBiBhE7ism+Wfr81HShk42TDht4c-Yp_DY0cp35OCNqLQ@mail.gmail.com>
 <CAGCz3RCjv0GHRkYmaPd31QLjd6pWBTdgxbbBmdeVRYwz0vU=RA@mail.gmail.com>
 <CAGCz3RBuuhP=BCE3U69sVea2tHvLNUqWinzU2USTPhY2Fq8cqw@mail.gmail.com>
Message-ID: <CAGCz3RDgNu_fP8za98iQ0kV93Sz3exHwxNnbzof63rBtHynk4w@mail.gmail.com>

Hi Michael,

Yes, the version I am using is a development version (1.5.14). In any case,
I have performed the same tests in version 1.6 and have the same behavior.
Instead, the verto module does block the sslv3 protocol.


Thanks,
Agustí

2017-06-20 10:45 GMT+02:00 Agustí Ubalde Bellot <agubbe at gmail.com>:

> Hi Michael,
>
> I have performed several connection tests forcing the sslv3 protocol over
> secure web sockets and the connection is established. Instead, the same
> test connecting to the TLS listening port, the connection is not set. The
> protocol is successfully disabled in the configuration.
> The version of FreeSWITCH I'm testing is 1.5.14. Is there any way to prove
> that the sslv3 protocol is actually disabled in this release?
>
>
> Thanks,
> Agustí
>
> 2017-06-15 10:07 GMT+02:00 Agustí Ubalde Bellot <agubbe at gmail.com>:
>
>> Hi Brian,
>>
>> Is possible to disable for web socket secure connections too?
>>
>>
>> Thanks,
>> Agustí
>>
>> 2017-06-13 13:24 GMT+02:00 Agustí Ubalde Bellot <agubbe at gmail.com>:
>>
>>> Hi all,
>>>
>>> Is there a FreeSWITCH update where sslv3 support is disabled?
>>>
>>>
>>> Thanks,
>>> Agustí
>>>
>>
>>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170620/9c411a8c/attachment.html>

From joelists at tm.net.uk  Tue Jun 20 16:21:52 2017
From: joelists at tm.net.uk (Joseph Waite)
Date: Tue, 20 Jun 2017 17:21:52 +0100
Subject: [Freeswitch-users] Radius auth based on IP without sending 407
	Proxy authentication required
Message-ID: <ABA4EF5D-1776-49FA-B29D-A582E1562B71@tm.net.uk>

Hi Guys

I am trying to configure a Sofia Profile that will not send a 407 Proxy Authentication Required, but will still authenticate the incoming invite via Radius based on the IP address of the INVITE.

If I change the Auth_calls to false on the Sofia profile, it doesn’t send the 407, but then it doesn’t authenticate the call.

Any help would be most appreciated.

Regards

From mike at jerris.com  Tue Jun 20 16:29:38 2017
From: mike at jerris.com (Michael Jerris)
Date: Tue, 20 Jun 2017 12:29:38 -0400
Subject: [Freeswitch-users] Freeswitch sslv3 support
In-Reply-To: <CAGCz3RDgNu_fP8za98iQ0kV93Sz3exHwxNnbzof63rBtHynk4w@mail.gmail.com>
References: <CAGCz3RBiBhE7ism+Wfr81HShk42TDht4c-Yp_DY0cp35OCNqLQ@mail.gmail.com>
 <CAGCz3RCjv0GHRkYmaPd31QLjd6pWBTdgxbbBmdeVRYwz0vU=RA@mail.gmail.com>
 <CAGCz3RBuuhP=BCE3U69sVea2tHvLNUqWinzU2USTPhY2Fq8cqw@mail.gmail.com>
 <CAGCz3RDgNu_fP8za98iQ0kV93Sz3exHwxNnbzof63rBtHynk4w@mail.gmail.com>
Message-ID: <E4ED19AB-9524-4942-96F2-3FDAE7C410CC@jerris.com>

I just reviewed the code. Looks like we disable it all on verto, but not on sofia.  should get me a Jira on this (I was just told one just got made)… we should fix that.  That being said, the sofia web socket support was basically a demo to prove we could do it before we finished verto, there is little reason to use sip over websockets and I never recommend it.  Also, using years old development code should be considered a massive security vulnerability and I would STRONGLY recommend against it.  


> On Jun 20, 2017, at 11:55 AM, Agustí Ubalde Bellot <agubbe at gmail.com> wrote:
> 
> Hi Michael,
> 
> Yes, the version I am using is a development version (1.5.14). In any case, I have performed the same tests in version 1.6 and have the same behavior.
> Instead, the verto module does block the sslv3 protocol.
> 
> 
> Thanks,
> Agustí
> 
> 2017-06-20 10:45 GMT+02:00 Agustí Ubalde Bellot <agubbe at gmail.com <mailto:agubbe at gmail.com>>:
> Hi Michael,
> 
> I have performed several connection tests forcing the sslv3 protocol over secure web sockets and the connection is established. Instead, the same test connecting to the TLS listening port, the connection is not set. The protocol is successfully disabled in the configuration.
> The version of FreeSWITCH I'm testing is 1.5.14. Is there any way to prove that the sslv3 protocol is actually disabled in this release?
> 
> 
> Thanks,
> Agustí
> 
> 2017-06-15 10:07 GMT+02:00 Agustí Ubalde Bellot <agubbe at gmail.com <mailto:agubbe at gmail.com>>:
> Hi Brian,
> 
> Is possible to disable for web socket secure connections too?
> 
> 
> Thanks,
> Agustí
> 
> 2017-06-13 13:24 GMT+02:00 Agustí Ubalde Bellot <agubbe at gmail.com <mailto:agubbe at gmail.com>>:
> Hi all,
> 
> Is there a FreeSWITCH update where sslv3 support is disabled?

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170620/e68483fe/attachment-0001.html>

From s.safarov at gmail.com  Tue Jun 20 16:32:15 2017
From: s.safarov at gmail.com (Sergey Safarov)
Date: Tue, 20 Jun 2017 16:32:15 +0000
Subject: [Freeswitch-users] Radius auth based on IP without sending 407
 Proxy authentication required
In-Reply-To: <ABA4EF5D-1776-49FA-B29D-A582E1562B71@tm.net.uk>
References: <ABA4EF5D-1776-49FA-B29D-A582E1562B71@tm.net.uk>
Message-ID: <CAHtxdDePe2OQ38_FiXdxhVYShnH-AihQ6A0Ln=OjRgT=s9rTZw@mail.gmail.com>

You can try mod_xml_radius and generate directory record with cidr adribute.

вт, 20 июня 2017 г., 19:25 Joseph Waite <joelists at tm.net.uk>:

> Hi Guys
>
> I am trying to configure a Sofia Profile that will not send a 407 Proxy
> Authentication Required, but will still authenticate the incoming invite
> via Radius based on the IP address of the INVITE.
>
> If I change the Auth_calls to false on the Sofia profile, it doesn’t send
> the 407, but then it doesn’t authenticate the call.
>
> Any help would be most appreciated.
>
> Regards
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170620/1db4dab3/attachment.html>

From acheraime at gmail.com  Tue Jun 20 16:35:22 2017
From: acheraime at gmail.com (acheraime at gmail.com)
Date: Tue, 20 Jun 2017 12:35:22 -0400
Subject: [Freeswitch-users] Calling on the community for Bug Marshals
In-Reply-To: <CAFGRPVpcLXFbHt7YMrSTajyz-xSQSz9hidgRncLYnsM0P0Qevw@mail.gmail.com>
References: <CAEJMVkChkspixRypT0rMCzawf4F14Tn3vUzYFzN-qtiqVmRnrg@mail.gmail.com>
 <CABMX_w-Ts8x3SPP5=a+dm+w4vq8g4jFsQVhrXWbuHL5=8XR2zw@mail.gmail.com>
 <CAFGRPVpcLXFbHt7YMrSTajyz-xSQSz9hidgRncLYnsM0P0Qevw@mail.gmail.com>
Message-ID: <CB05F33E-A385-4C80-B132-990993D9FEB6@gmail.com>

I would be happy to help. Let me know what the "requirements" are.

Sent from my iPhone

> On Jun 20, 2017, at 10:30 AM, David Villasmil <david.villasmil.work at gmail.com> wrote:
> 
> Anything i can help with?
>> On Tue, Jun 20, 2017 at 2:46 PM Ítalo Rossi <italo at freeswitch.org> wrote:
>> This is a great opportunity to learn and to be an expert in FreeSWITCH. This was how I learn a lot!
>> 
>> :-)
>> 
>>> On Mon, Jun 5, 2017 at 4:39 PM, Brian West <brian at freeswitch.org> wrote:
>>> FreeSWITCHers,
>>> 
>>> We are in need of a few good bug marshals, We are trying to get 1.8 ready and out the door and the more help we have testing and working thru patches on JIRA the quicker it will arrive.  If you're interested in helping us out email me directly.  We are also considering bringing back a few days a week we are sitting in 888 and helping the community out with issues pending in JIRA.
>>> 
>>> Also we are only about 2600 short on the gofund me for the Allison prompts, which will be delivered sometime this week.  ;)  So help us get over that last little bit this week.
>>> 
>>> Thanks,
>>> 
>>> -- 
>>> Brian West
>>> brian at freeswitch.org
>>> 
>>> Twitter: @FreeSWITCH , @briankwest
>>> 
>>> http://www.freeswitchbook.com 
>>> http://www.freeswitchcookbook.com
>>> 
>>> Book a phone call (CST)
>>> 
>>> Allison prompts for FreeSWITCH:
>>> 
>>> https://www.gofundme.com/allison-prompts-for-freeswitch
>>> 
>>> Got Bugs? Report them here! | Reddit: /r/freeswitch
>>> 
>>> T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378)
>>> Skype:briankwest
>>> 
>>> 
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.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
>>> http://www.freeswitch.org
>> 
>> 
>> 
>> -- 
>> Ítalo Rossi
>> italo at freeswitch.org
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.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
>> http://www.freeswitch.org
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170620/c145005f/attachment.html>

From s.safarov at gmail.com  Tue Jun 20 16:36:22 2017
From: s.safarov at gmail.com (Sergey Safarov)
Date: Tue, 20 Jun 2017 16:36:22 +0000
Subject: [Freeswitch-users] Radius auth based on IP without sending 407
 Proxy authentication required
References: <ABA4EF5D-1776-49FA-B29D-A582E1562B71@tm.net.uk>
 <CAHtxdDePe2OQ38_FiXdxhVYShnH-AihQ6A0Ln=OjRgT=s9rTZw@mail.gmail.com>
Message-ID: <CAHtxdDcz+x0v_=ohLRkHdN2x39ULJorO6QNqbfhBCubmLqdi4A@mail.gmail.com>

Also required to generate "domain" acl when freeswith starts, executed
command "reloadacl" and updated client ip on radius server side

вт, 20 июня 2017 г., 19:32 Sergey Safarov <s.safarov at gmail.com>:

> You can try mod_xml_radius and generate directory record with cidr
> adribute.
>
> вт, 20 июня 2017 г., 19:25 Joseph Waite <joelists at tm.net.uk>:
>
>> Hi Guys
>>
>> I am trying to configure a Sofia Profile that will not send a 407 Proxy
>> Authentication Required, but will still authenticate the incoming invite
>> via Radius based on the IP address of the INVITE.
>>
>> If I change the Auth_calls to false on the Sofia profile, it doesn’t send
>> the 407, but then it doesn’t authenticate the call.
>>
>> Any help would be most appreciated.
>>
>> Regards
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.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
>> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170620/fb99a036/attachment.html>

From joelists at tm.net.uk  Tue Jun 20 16:41:01 2017
From: joelists at tm.net.uk (Joseph Waite)
Date: Tue, 20 Jun 2017 17:41:01 +0100
Subject: [Freeswitch-users] Radius auth based on IP without sending 407
	Proxy authentication required
In-Reply-To: <CAHtxdDePe2OQ38_FiXdxhVYShnH-AihQ6A0Ln=OjRgT=s9rTZw@mail.gmail.com>
References: <ABA4EF5D-1776-49FA-B29D-A582E1562B71@tm.net.uk>
 <CAHtxdDePe2OQ38_FiXdxhVYShnH-AihQ6A0Ln=OjRgT=s9rTZw@mail.gmail.com>
Message-ID: <E931B170-4B18-448D-B075-6247DC18D99F@tm.net.uk>

I am using mod_xml_radius, however my issue is that if I enable auth_calls in profile it sends a 407 Proxy Authentication Required sip message, and if I set auth_calls to false it doesn’t authenticate with Radius, it simply passes call straight into the dial plan.


> On 20 Jun 2017, at 17:32, Sergey Safarov <s.safarov at gmail.com> wrote:
> 
> You can try mod_xml_radius and generate directory record with cidr adribute.
> 
> 
> вт, 20 июня 2017 г., 19:25 Joseph Waite <joelists at tm.net.uk <mailto:joelists at tm.net.uk>>:
> Hi Guys
> 
> I am trying to configure a Sofia Profile that will not send a 407 Proxy Authentication Required, but will still authenticate the incoming invite via Radius based on the IP address of the INVITE.
> 
> If I change the Auth_calls to false on the Sofia profile, it doesn’t send the 407, but then it doesn’t authenticate the call.
> 
> Any help would be most appreciated.
> 
> Regards
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> http://www.cluecon.com <http://www.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>
> http://www.freeswitch.org <http://www.freeswitch.org/>_________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20170620/f3aa2d86/attachment-0001.html>

From s.safarov at gmail.com  Tue Jun 20 17:15:16 2017
From: s.safarov at gmail.com (Sergey Safarov)
Date: Tue, 20 Jun 2017 17:15:16 +0000
Subject: [Freeswitch-users] Radius auth based on IP without sending 407
 Proxy authentication required
In-Reply-To: <E931B170-4B18-448D-B075-6247DC18D99F@tm.net.uk>
References: <ABA4EF5D-1776-49FA-B29D-A582E1562B71@tm.net.uk>
 <CAHtxdDePe2OQ38_FiXdxhVYShnH-AihQ6A0Ln=OjRgT=s9rTZw@mail.gmail.com>
 <E931B170-4B18-448D-B075-6247DC18D99F@tm.net.uk>
Message-ID: <CAHtxdDdops_NNHC7fa05ONfPpu-xVaE3zmMwgFHTmW_YTG8Q9g@mail.gmail.com>

When you generate manually ACL with trusted IP or via mod_xml_radius then
you can accept call without 407 message.
Also if you add cird attribute to user directory then FreeSwitch can map
call to user via cidr attribute.

вт, 20 июн. 2017 г. в 19:43, Joseph Waite <joelists at tm.net.uk>:

> I am using mod_xml_radius, however my issue is that if I enable auth_calls
> in profile it sends a 407 Proxy Authentication Required sip message, and if
> I set auth_calls to false it doesn’t authenticate with Radius, it simply
> passes call straight into the dial plan.
>
>
> On 20 Jun 2017, at 17:32, Sergey Safarov <s.safarov at gmail.com> wrote:
>
> You can try mod_xml_radius and generate directory record with cidr
> adribute.
>
> вт, 20 июня 2017 г., 19:25 Joseph Waite <joelists at tm.net.uk>:
>
>> Hi Guys
>>
>> I am trying to configure a Sofia Profile that will not send a 407 Proxy
>> Authentication Required, but will still authenticate the incoming invite
>> via Radius based on the IP address of the INVITE.
>>
>> If I change the Auth_calls to false on the Sofia profile, it doesn’t send
>> the 407, but then it doesn’t authenticate the call.
>>
>> Any help would be most appreciated.
>>
>> Regards
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.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
>> http://www.freeswitch.org
>
> _______________________________________________________________