From alex at freeswitch.com  Fri Feb  1 04:12:32 2019
From: alex at freeswitch.com (Alexey Sibyakin)
Date: Fri, 1 Feb 2019 13:12:32 +0900
Subject: [Freeswitch-users] Debian 9 FreeSWITCH 1.8 Sangoma Card A102
In-Reply-To: <CAJVaC-tLVsdoOfqKWx4S47_Lt+37US2fbvWMuJX2+VLD-Mt1sw@mail.gmail.com>
References: <CAJVaC-tLVsdoOfqKWx4S47_Lt+37US2fbvWMuJX2+VLD-Mt1sw@mail.gmail.com>
Message-ID: <CAGJLwFg+c4cs+7THz9sv6sRk-1Oe=toZ9Ft_10n8ORwMZ31QHQ@mail.gmail.com>

There are too few lines in your log so it's hard to say for sure what's
going on, but try to do

export CFLAGS="-Wno-error"
export CXXFLAGS="-Wno-error"

before compilation.

Regards,

Alex

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



On Fri, Feb 1, 2019 at 2:10 AM Alaadin Abdurrahman <alaadin.abd at gmail.com>
wrote:

> Dears,
> I have an issue when i try to compile and make freeswitch 1.8 with
> mod_freetdm to use sangoma card.
> please note that i followed every step in the sangoma wiki
>
>  https://wiki.freepbx.org/display/PC/Telephony+Cards+for+FreeSWITCH
>
> and i still get this error message
>
> cc1: all warnings being treated as errors
> Makefile:1209: recipe for target
> 'ftmod_sangoma_isdn_la-ftmod_sangoma_isdn.lo' failed
> make[2]: *** [ftmod_sangoma_isdn_la-ftmod_sangoma_isdn.lo] Error 1
> make[2]: Leaving directory '/usr/src/freeswitch/libs/freetdm'
> Makefile:15: recipe for target '../libfreetdm.la' failed
> make[1]: *** [../libfreetdm.la] Error 2
> make[1]: Leaving directory '/usr/src/freeswitch/libs/freetdm/mod_freetdm'
> ../../../build/modmake.rules:89: recipe for target 'all' failed
> make: *** [all] Error 1
> root at debian:/usr/src/freeswitch/libs/freetdm/mod_freetdm#
>
> best regards
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From bipin at xbipin.com  Fri Feb  1 06:12:42 2019
From: bipin at xbipin.com (Bipin Patel)
Date: Fri, 1 Feb 2019 10:12:42 +0400
Subject: [Freeswitch-users] Sofia stops responding after a few days
In-Reply-To: <aee6a482-f936-b248-18b4-fb7d46cc473e@xbipin.com>
References: <5c53627a-9596-840e-efa0-f978f784fc7f@xbipin.com>
 <008c01d37b01$f96b9f40$ec42ddc0$@smartic.es>
 <1607ec61170.279b.b07ebdf329620b8089087c7205b03f01@xbipin.com>
 <8a89ffbd-db08-1931-db1e-78b9e6b392e5@airmail.cc>
 <adb3ad5e-189d-0f4c-a8c6-d310230cd984@xbipin.com>
 <CAEJMVkAP2Fsq9hyu+p5RyTdfuRAOt_s0RnH7n0Sg0hq3fV0Ddg@mail.gmail.com>
 <801bc750-2c8b-2dc3-698b-43265e0eb484@xbipin.com>
 <0287837f-8ff5-185f-c0ec-8878a84e3f7a@xbipin.com>
 <aee6a482-f936-b248-18b4-fb7d46cc473e@xbipin.com>
Message-ID: <ac868900-a26e-b135-199e-35a93e8f8ae0@xbipin.com>

hi,

still plagued by this issue without a solution and on windows when this 
happens cant even stop the service, have to kill it and then start it to 
get it to work. Happens only when using a SIP profile for TLS and SRTP


Regards,
Bipin


------------------------------------------------------------------------
On 13/3/2018 6:57 PM, Bipin Patel wrote:
> hi,
>
> its been quiet some time and cant seem to get over this issue
>
>
> Regards,
> Bipin
>
>
> ------------------------------------------------------------------------
> -------- Original Message --------
> Subject: Re: [Freeswitch-users] Sofia stops responding after a few days
> From: Bipin Patel <bipin at xbipin.com>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Date: 1/28/2018, 9:51:23 AM
>> hi,
>>
>> any pointers on how to solve this?
>>
>>
>> Regards,
>> Bipin
>>
>>
>> ------------------------------------------------------------------------
>> -------- Original Message --------
>> Subject: Re: [Freeswitch-users] Sofia stops responding after a few days
>> From: Bipin Patel <bipin at xbipin.com>
>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> Date: 1/26/2018, 7:02:49 PM
>>> hi,
>>>
>>> im on the latest master but this has been happening since few months 
>>> now and i almost checked all the modules and disabled everything 
>>> other than xml_curl, sofia on tls which i use and its happening only 
>>> on the sofia TLS profile
>>> Happens every few days at random times
>>>
>>> FreeSWITCH Version 1.9.0+git~20180119T195505Z~3f8585f636~64bit (git 
>>> 3f8585f 2018-01-19 19:55:05Z 64bit)
>>>
>>>
>>> Regards,
>>> Bipin
>>>
>>>
>>> ------------------------------------------------------------------------
>>> -------- Original Message --------
>>> Subject: Re: [Freeswitch-users] Sofia stops responding after a few days
>>> From: Brian West <brian at freeswitch.com>
>>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>> Date: 1/26/2018, 6:53:56 PM
>>>> Critical information needed, What FreeSWITCH Revision?
>>>>
>>>> /b
>>>>
>>>>
>>>> On Fri, Jan 26, 2018 at 8:49 AM, Bipin Patel <bipin at xbipin.com 
>>>> <mailto:bipin at xbipin.com>> wrote:
>>>>
>>>>     hi,
>>>>
>>>>     after trying a lot i still suffer the issue and when this
>>>>     happens i see the below constantly flooding the cli
>>>>
>>>>     ..\..\sofia-sip\libsofia-sip-ua\tport\tport_type_tls.c:434
>>>>     tport_tls_recv() tport_type_tls(00000006A25BB0C0): tls_read()
>>>>     returned -1
>>>>     ..\..\sofia-sip\libsofia-sip-ua\tport\tport.c:2864
>>>>     tport_recv_event() tport_recv_event(00000006A25BB0C0)
>>>>     ..\..\sofia-sip\libsofia-sip-ua\tport\tport_type_tls.c:434
>>>>     tport_tls_recv() tport_type_tls(00000006A25BB0C0): tls_read()
>>>>     returned -1
>>>>     ..\..\sofia-sip\libsofia-sip-ua\tport\tport.c:2864
>>>>     tport_recv_event() tport_recv_event(00000006A25BB0C0)
>>>>     ..\..\sofia-sip\libsofia-sip-ua\tport\tport_type_tls.c:434
>>>>     tport_tls_recv() tport_type_tls(00000006A25BB0C0): tls_read()
>>>>     returned -1
>>>>     ..\..\sofia-sip\libsofia-sip-ua\tport\tport.c:2864
>>>>     tport_recv_event() tport_recv_event(00000006A25BB0C0)
>>>>     ..\..\sofia-sip\libsofia-sip-ua\tport\tport_type_tls.c:434
>>>>     tport_tls_recv() tport_type_tls(00000006A25BB0C0): tls_read()
>>>>     returned -1
>>>>     ..\..\sofia-sip\libsofia-sip-ua\tport\tport.c:2864
>>>>     tport_recv_event() tport_recv_event(00000006A25BB0C0)
>>>>     ..\..\sofia-sip\libsofia-sip-ua\tport\tport_type_tls.c:434
>>>>     tport_tls_recv() tport_type_tls(00000006A25BB0C0): tls_read()
>>>>     returned -1
>>>>     ..\..\sofia-sip\libsofia-sip-ua\tport\tport.c:2864
>>>>     tport_recv_event() tport_recv_event(00000006A25BB0C0)
>>>>     ..\..\sofia-sip\libsofia-sip-ua\tport\tport_type_tls.c:434
>>>>     tport_tls_recv() tport_type_tls(00000006A25BB0C0): tls_read()
>>>>     returned -1
>>>>
>>>>
>>>>     sorry for double posting, sent it to the wrong thread earlier
>>>>
>>>>     Regards,
>>>>     Bipin
>>>>
>>>>
>>>>     ------------------------------------------------------------------------
>>>>     -------- Original Message --------
>>>>     Subject: Re: [Freeswitch-users] Sofia stops responding after a
>>>>     few days
>>>>     From: Ryan Harris <ryharris at airmail.cc>
>>>>     <mailto:ryharris at airmail.cc>
>>>>     To: freeswitch-users at lists.freeswitch.org
>>>>     <mailto:freeswitch-users at lists.freeswitch.org>
>>>>     Date: 12/22/2017, 8:55:42 PM
>>>>>
>>>>>     Hello,
>>>>>
>>>>>     This comment in the default sip_profiles/internal.xml has
>>>>>     caught my eye in the past:
>>>>>     https://freeswitch.org/stash/projects/FS/repos/freeswitch/browse/conf/vanilla/sip_profiles/internal.xml#63-82
>>>>>     <https://freeswitch.org/stash/projects/FS/repos/freeswitch/browse/conf/vanilla/sip_profiles/internal.xml#63-82>
>>>>>
>>>>>     Maybe you can enable the watchdog and see if you can get a
>>>>>     useful core dump.
>>>>>
>>>>>
>>>>>     On 12/22/2017 10:09 AM, Bipin Patel wrote:
>>>>>>
>>>>>>     Hi,
>>>>>>
>>>>>>     Thanks for the reply but in my case the server already has
>>>>>>     8gb of ram and it's not at all busy and plus this instance of
>>>>>>     fs hardly has less than 10 registrations at any given time.
>>>>>>
>>>>>>     On December 22, 2017 12:53:04 PM Miguel Jesús López Valverde
>>>>>>     <mjlopez at smartic.es> <mailto:mjlopez at smartic.es> wrote:
>>>>>>
>>>>>>>     I had similar problems with FS installed under an Amazon EFS
>>>>>>>     instance. When FS did not attend registration requests and
>>>>>>>     executed the "sofia profile internal restart" command, it
>>>>>>>     did not load the profile and it no longer appeared before
>>>>>>>     the "sofia status" query.
>>>>>>>
>>>>>>>     I checked that this instance was short of ram memory and I
>>>>>>>     changed the instance to a higher one with more memory. Since
>>>>>>>     then I have not appreciated this problem again.
>>>>>>>
>>>>>>>     Receive a greeting.
>>>>>>>
>>>>>>>     *De:*FreeSWITCH-users
>>>>>>>     [mailto:freeswitch-users-bounces at lists.freeswitch.org
>>>>>>>     <mailto:freeswitch-users-bounces at lists.freeswitch.org>] *En
>>>>>>>     nombre de *Bipin Patel
>>>>>>>     *Enviado el:* viernes, 22 de diciembre de 2017 7:46
>>>>>>>     *Para:* FreeSWITCH Users Help
>>>>>>>     <freeswitch-users at lists.freeswitch.org>
>>>>>>>     <mailto:freeswitch-users at lists.freeswitch.org>
>>>>>>>     *Asunto:* [Freeswitch-users] Sofia stops responding after a
>>>>>>>     few days
>>>>>>>
>>>>>>>     hi,
>>>>>>>
>>>>>>>     I have 2 instances of FS running on a single windows box,
>>>>>>>     first instance uses normal sip UDP profiles mainly used for
>>>>>>>     routing calls to carriers and its running as a service since
>>>>>>>     a few months without any issues. The second instance runs a
>>>>>>>     sip TLS profile and accepts inbound registrations and
>>>>>>>     forwards calls in sip UDP ahead, it uses xml_curl for
>>>>>>>     directory users but the problem is every 2-3 days the
>>>>>>>     inbound TLS profile stops responding to registrations, and
>>>>>>>     when that happens i cant even stop and restart the service,
>>>>>>>     have to kill it and start again. FS_CLI works but doesnt
>>>>>>>     show any error, at first i though it could be the xml_curl
>>>>>>>     causing the issue but later realized it never sends any
>>>>>>>     requests when sofia stops responding.
>>>>>>>
>>>>>>>     i have been banging my head from the past week or so but not
>>>>>>>     able to find the cause, could any1 help in guiding me what
>>>>>>>     to check when this happens so can find the root cause, im
>>>>>>>     using commercial certs for TLS and when its running there r
>>>>>>>     no issues other than sofia stops responding every few days,
>>>>>>>     it happens at random times.
>>>>>>>
>>>>>>>     -- 
>>>>>>>     Regards,
>>>>>>>     Bipin
>>>>>>>
>>>>>>>     ------------------------------------------------------------------------
>>>>>>>
>>>>>>>     <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>
>>>>>>>
>>>>>>>
>>>>>>>     <#m_7793210244807471277_DAB4FAD8-2DD7-40BB-A1B8-4E2AA1F9FDF2>
>>>>>>>     _________________________________________________________________________
>>>>>>>     Professional FreeSWITCH Consulting Services:
>>>>>>>     consulting at freeswitch.org <mailto:consulting%40freeswitch.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%40lists.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
>>>>>>
>>>>>>
>>>>>>     _________________________________________________________________________
>>>>>>     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
>>>>>
>>>>>
>>>>>
>>>>>     _________________________________________________________________________
>>>>>     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
>>>>
>>>>
>>>>     _________________________________________________________________________
>>>>     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
>>>>
>>>>
>>>>
>>>>
>>>> -- 
>>>>
>>>> Brian West | Co-founder and Developer
>>>>
>>>> Need Commercial support? email sales at freeswitch.com 
>>>> <mailto:sales at freeswitch.com>
>>>>
>>>> FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 
>>>> <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g>
>>>>
>>>> Email: brian at freeswitch.com <mailto:brian at freeswitch.com>
>>>>
>>>> Mobile: 918-424-9378
>>>>
>>>> Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/>
>>>>
>>>> color-facebook-96.png 
>>>> <https://www.facebook.com/freeswitch/>color-twitter-96.png 
>>>> <https://twitter.com/freeswitch?ref_src=twsrc%5Egoogle%7Ctwcamp%5Eserp%7Ctwgr%5Eauthor>
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at 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

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From alaadin.abd at gmail.com  Fri Feb  1 12:26:08 2019
From: alaadin.abd at gmail.com (Alaadin Abdurrahman)
Date: Fri, 1 Feb 2019 14:26:08 +0200
Subject: [Freeswitch-users] Debian 9 FreeSWITCH 1.8 Sangoma Card A102
In-Reply-To: <CAGJLwFg+c4cs+7THz9sv6sRk-1Oe=toZ9Ft_10n8ORwMZ31QHQ@mail.gmail.com>
References: <CAJVaC-tLVsdoOfqKWx4S47_Lt+37US2fbvWMuJX2+VLD-Mt1sw@mail.gmail.com>
 <CAGJLwFg+c4cs+7THz9sv6sRk-1Oe=toZ9Ft_10n8ORwMZ31QHQ@mail.gmail.com>
Message-ID: <CAJVaC-stX=ooZ0TCgsi2-aOXuDvVmAj9pftNQOLKLOOBLk8HVA@mail.gmail.com>

Thanks Alex, I will give it a try.

On Fri, Feb 1, 2019 at 10:00 AM Alexey Sibyakin <alex at freeswitch.com> wrote:

> There are too few lines in your log so it's hard to say for sure what's
> going on, but try to do
>
> export CFLAGS="-Wno-error"
> export CXXFLAGS="-Wno-error"
>
> before compilation.
>
> Regards,
>
> Alex
>
> Alex Sibyakin | Support Engineer SignalWire | 228 Hamilton Ave 3rd Floor,
> Palo Alto, CA 94303 Email: alex at freeswitch.com Website:
> https://www.signalwire.com
>
>
>
> On Fri, Feb 1, 2019 at 2:10 AM Alaadin Abdurrahman <alaadin.abd at gmail.com>
> wrote:
>
>> Dears,
>> I have an issue when i try to compile and make freeswitch 1.8 with
>> mod_freetdm to use sangoma card.
>> please note that i followed every step in the sangoma wiki
>>
>>  https://wiki.freepbx.org/display/PC/Telephony+Cards+for+FreeSWITCH
>>
>> and i still get this error message
>>
>> cc1: all warnings being treated as errors
>> Makefile:1209: recipe for target
>> 'ftmod_sangoma_isdn_la-ftmod_sangoma_isdn.lo' failed
>> make[2]: *** [ftmod_sangoma_isdn_la-ftmod_sangoma_isdn.lo] Error 1
>> make[2]: Leaving directory '/usr/src/freeswitch/libs/freetdm'
>> Makefile:15: recipe for target '../libfreetdm.la' failed
>> make[1]: *** [../libfreetdm.la] Error 2
>> make[1]: Leaving directory '/usr/src/freeswitch/libs/freetdm/mod_freetdm'
>> ../../../build/modmake.rules:89: recipe for target 'all' failed
>> make: *** [all] Error 1
>> root at debian:/usr/src/freeswitch/libs/freetdm/mod_freetdm#
>>
>> best regards
>>
> _________________________________________________________________________
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From rick at magicmail.mooo.com  Fri Feb  1 12:52:07 2019
From: rick at magicmail.mooo.com (Rick Jarvis)
Date: Fri, 1 Feb 2019 12:52:07 +0000
Subject: [Freeswitch-users] Verto not segregating domains
Message-ID: <etPan.5c5440f7.1dc9d55c.2c3@magicmail.mooo.com>

Ok I know this must be me confusing domains / contexts etc, but can’t work out where I’ve gone wrong.

Situation: Verto, using xmlcurl from database server for directory. Verto users are user@ with multiple domains to segregate conferences. 

The xml being returned is below - I *thought* that simply having the <domain name=“whatever”> would keep things separate if the domains were different, but it seems this isn’t the case...

<document type="freeswitch/xml">
    <section name="directory">
      <domain name="domain2">
        <groups>
          <group name="default">
           <users>
              <user id="user">
                <params>
                  <param name="password" value="m6h20sj1y1pmysmayst3"/>
                  <param name="vm-password" value="982375982735"/>
                  <param name="jsonrpc-allowed-methods" value="verto"/>
                  <param name="jsonrpc-allowed-event-channels" value="demo,conference,presence"/>
                  <param name="jsonrpc-allowed-jsapi" value="all"/>
                  <param name="jsonrpc-allowed-fsapi" value="all"/>
                  <param name="verto-context" value="default"/>
                  <param name="verto-dialplan" value="XML"/>
                </params>
                <variables>
                  <variable name="toll_allow" value="domestic,international,local"/>
                  <variable name="accountcode" value="user"/>
                  <variable name="user_context" value="default"/>
                  <variable name="effective_caller_id_name" value="user"/>
                  <variable name="effective_caller_id_number" value="user"/>
                  <variable name="outbound_caller_id_name" value="user"/>
                  <variable name="outbound_caller_id_number" value="user"/>
                </variables>
              </user>
              </users>
          </group>
      </groups>
  </domain>
</section>
</document>

Output from verto status here shows users correctly on different domains, but maybe the prefix is what’s affecting it here? Or some other setting?

default-v4::user at domain1	client	                      x.x.x.x:43249	CONN_REG (WSS)
default-v4::user at domain2	client	                      y.y.y.y:55987	CONN_REG (WSS)


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From ch.chhatra at gmail.com  Fri Feb  1 13:05:09 2019
From: ch.chhatra at gmail.com (Chhorm Chhatra)
Date: Fri, 1 Feb 2019 20:05:09 +0700
Subject: [Freeswitch-users] MOD_CALLCENTER become unload and unknown
 while freeswitch is still running
In-Reply-To: <CAGJLwFi2oTV5XeF7HzhLPwq7nBdbfhDHbEYJJ4Mf+wxh8NTfbQ@mail.gmail.com>
References: <CAJssr-iSR5v3_3ctqSTScnBOKe4GXyTxME8s2EDH80jtrvvSHg@mail.gmail.com>
 <CAGJLwFi2oTV5XeF7HzhLPwq7nBdbfhDHbEYJJ4Mf+wxh8NTfbQ@mail.gmail.com>
Message-ID: <CAJssr-g2zOHhhf8gRAvEkg-=2nFuqnQ5ybOvNND4darnJ8WGXg@mail.gmail.com>

Thank you for your help.

On Tue, 15 Jan 2019 at 07:59, Alexey Sibyakin <alex at freeswitch.com> wrote:

> Hi,
>
> Try to update to 1.8 first. FreeSWITCH 1.6 is EOL.
>
> Regards,
>
> Alex
>
> Alex Sibyakin | Support Engineer SignalWire | 228 Hamilton Ave 3rd Floor,
> Palo Alto, CA 94303 Email: alex at freeswitch.com Website:
> https://www.signalwire.com
>
>
>
> On Tue, Jan 15, 2019 at 2:07 AM Chhorm Chhatra <ch.chhatra at gmail.com>
> wrote:
>
>> Hi,
>> Currently, I faced a very strange issue that
>> 1) After several days of FreeSWITCH running, I could not connect to its
>> CLI at all. But Freeswitch is still working as it is. I have to tail the
>> log file to debug this issue.
>> 2) When the issue (1) happens, the application *callcenter *is unknown
>> and therefore the callcenter failed.
>> FS version: 1.6
>> Any help would be appreciated.
>> Thank you in advance.
>> Best regards,
>> Chhorm Chhatra
>> _________________________________________________________________________
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From ch.chhatra at gmail.com  Fri Feb  1 13:13:14 2019
From: ch.chhatra at gmail.com (Chhorm Chhatra)
Date: Fri, 1 Feb 2019 20:13:14 +0700
Subject: [Freeswitch-users] No Media in SIP Calling from 3G Mobile Internet
	to Freeswitch
Message-ID: <CAJssr-iW9F5z4R+apRmu7SvLtR7yCL56wkWgwovzGSygfAtzYQ@mail.gmail.com>

Hi,
Currently, I face an issue that I could not hear the audio sound at all
when I try to call to FreeSWITCH (9386) via its public domain name from the
SIP client of my mobile phone with mobile internet.
In this case, SIP registration with TLS and media encryption with SRTP is
working perfectly because I could see the secure symbol in Linphone *but* I
could not hear any audio at all. After 30 seconds of the silent call, it
hanged up automatically.

*SIP Client on the 3G Mobile Internet > Freeswitch (tls:voip.test.com
<http://voip.test.com>, SRTP) *
*SIP Registration Sucess, Call No Media*

My question is that is this the problem of NAT? If so, it would be great if
you could help me solve this problem.
Any help would be really appreciated.

Best regards,
Chhorm Chhatra
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From me at anatoli.ws  Fri Feb  1 13:29:09 2019
From: me at anatoli.ws (Anatoli)
Date: Fri, 1 Feb 2019 10:29:09 -0300
Subject: [Freeswitch-users] Anybody want to go in on some Spanish GM
 voices recordings?
In-Reply-To: <CAJ0sghRn7RswtrMJ0hcS8P7Z14khRbvHscu-hwPh9NU7GbBG8w@mail.gmail.com>
References: <CAJ0sghRn7RswtrMJ0hcS8P7Z14khRbvHscu-hwPh9NU7GbBG8w@mail.gmail.com>
Message-ID: <094eac6e-cc19-329c-0bad-fab59947be7b@anatoli.ws>

Hi Chad,

Please see this conversation thread: 
http://lists.freeswitch.org/pipermail/freeswitch-users/2017-December/128504.html. 
There's more to FS with Spanish prompts than just recording the voice 
(this mail in the thread mentions the details: 
http://lists.freeswitch.org/pipermail/freeswitch-users/2017-December/128486.html).

Regards,
Anatoli

*From:* Chad Phillips <chad at apartmentlines.com>
*Sent:* Wednesday, January 30, 2019 21:12
*To:* 'Freeswitch Users Help' <freeswitch-users at lists.freeswitch.org>
*Subject:* [Freeswitch-users] Anybody want to go in on some Spanish GM 
voices recordings?

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

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

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

Chad

_________________________________________________________________________
Professional FreeSWITCH Services
sales at freeswitch.com
https://freeswitch.com

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

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


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From m.shirazi at gmail.com  Fri Feb  1 16:01:12 2019
From: m.shirazi at gmail.com (Mehdi Shirazi)
Date: Fri, 1 Feb 2019 19:31:12 +0330
Subject: [Freeswitch-users] Freeswitch As SBC for sip_phones
Message-ID: <CAG2rjpsb68CJo_B5RPsoBBmorYh8v-YvCXbmDNp2BsiHwXtJgQ@mail.gmail.com>

Hi
I know that Freeswitch is a B2BUA and not proxy but I have seen SBC's based
on Freeswitch also used for sipphones and siptrunks.
Is it possible to passthru sip registration with Freeswitch now ?

Regards
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From s.safarov at gmail.com  Fri Feb  1 21:34:51 2019
From: s.safarov at gmail.com (Sergey Safarov)
Date: Sat, 2 Feb 2019 00:34:51 +0300
Subject: [Freeswitch-users] Freeswitch As SBC for sip_phones
In-Reply-To: <CAG2rjpsb68CJo_B5RPsoBBmorYh8v-YvCXbmDNp2BsiHwXtJgQ@mail.gmail.com>
References: <CAG2rjpsb68CJo_B5RPsoBBmorYh8v-YvCXbmDNp2BsiHwXtJgQ@mail.gmail.com>
Message-ID: <CAHtxdDcGvYPD9_OgxgakzBQKV9GD=sjHR2BtW57SWccN47EWgA@mail.gmail.com>

In this case you can try kamailio + rtpengine.

пт, 1 февр. 2019 г., 19:35 Mehdi Shirazi <m.shirazi at gmail.com>:

> Hi
> I know that Freeswitch is a B2BUA and not proxy but I have seen SBC's
> based on Freeswitch also used for sipphones and siptrunks.
> Is it possible to passthru sip registration with Freeswitch now ?
>
> Regards
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From alex at freeswitch.com  Sat Feb  2 03:50:25 2019
From: alex at freeswitch.com (Alexey Sibyakin)
Date: Sat, 2 Feb 2019 12:50:25 +0900
Subject: [Freeswitch-users] Verto not segregating domains
In-Reply-To: <etPan.5c5440f7.1dc9d55c.2c3@magicmail.mooo.com>
References: <etPan.5c5440f7.1dc9d55c.2c3@magicmail.mooo.com>
Message-ID: <CAGJLwFjhPD=VgRukWqtm_4Rc+1VoMZ+1u1MGh71-i_zQqYjjHQ@mail.gmail.com>

Hey,

Are you sure that

<param name="force-register-domain" value="$${domain}"/>

in verto.conf.xml is commented?

Regards,

Alex

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



On Sat, Feb 2, 2019 at 3:01 AM Rick Jarvis <rick at magicmail.mooo.com> wrote:

> Ok I know this must be me confusing domains / contexts etc, but can’t work
> out where I’ve gone wrong.
>
> Situation: Verto, using xmlcurl from database server for directory. Verto
> users are user@ with multiple domains to segregate conferences.
>
> The xml being returned is below - I *thought* that simply having the
> <domain name=“whatever”> would keep things separate if the domains were
> different, but it seems this isn’t the case...
>
> <document type="freeswitch/xml">
>     <section name="directory">
>       <domain name="domain2">
>         <groups>
>           <group name="default">
>            <users>
>               <user id="user">
>                 <params>
>                   <param name="password" value="m6h20sj1y1pmysmayst3"/>
>                   <param name="vm-password" value="982375982735"/>
>                   <param name="jsonrpc-allowed-methods" value="verto"/>
>                   <param name="jsonrpc-allowed-event-channels"
> value="demo,conference,presence"/>
>                   <param name="jsonrpc-allowed-jsapi" value="all"/>
>                   <param name="jsonrpc-allowed-fsapi" value="all"/>
>                   <param name="verto-context" value="default"/>
>                   <param name="verto-dialplan" value="XML"/>
>                 </params>
>                 <variables>
>                   <variable name="toll_allow"
> value="domestic,international,local"/>
>                   <variable name="accountcode" value="user"/>
>                   <variable name="user_context" value="default"/>
>                   <variable name="effective_caller_id_name" value="user"/>
>                   <variable name="effective_caller_id_number"
> value="user"/>
>                   <variable name="outbound_caller_id_name" value="user"/>
>                   <variable name="outbound_caller_id_number" value="user"/>
>                 </variables>
>               </user>
>               </users>
>           </group>
>       </groups>
>   </domain>
> </section>
> </document>
>
> Output from verto status here shows users correctly on different domains,
> but maybe the prefix is what’s affecting it here? Or some other setting?
>
> default-v4::user at domain1 client                       x.x.x.x:43249 CONN_REG
> (WSS)
>
> default-v4::user at domain2 client                      y.y.y.y:55987 CONN_REG
> (WSS)
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From alex at freeswitch.com  Sat Feb  2 03:56:18 2019
From: alex at freeswitch.com (Alexey Sibyakin)
Date: Sat, 2 Feb 2019 12:56:18 +0900
Subject: [Freeswitch-users] No Media in SIP Calling from 3G Mobile
 Internet to Freeswitch
In-Reply-To: <CAJssr-iW9F5z4R+apRmu7SvLtR7yCL56wkWgwovzGSygfAtzYQ@mail.gmail.com>
References: <CAJssr-iW9F5z4R+apRmu7SvLtR7yCL56wkWgwovzGSygfAtzYQ@mail.gmail.com>
Message-ID: <CAGJLwFirQfQgx0Gk-O8GChoMhSbNJ_+67e-jSErbvpyNOXagEg@mail.gmail.com>

Hi,

Yes, that sounds like a network (NAT/Firewall) issue. Your mobile provider
may use a chain of several NATs and not every one of them is working
properly. Try VPN or another SIP client.

Regards,

Alex

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



On Sat, Feb 2, 2019 at 2:44 AM Chhorm Chhatra <ch.chhatra at gmail.com> wrote:

> Hi,
> Currently, I face an issue that I could not hear the audio sound at all
> when I try to call to FreeSWITCH (9386) via its public domain name from the
> SIP client of my mobile phone with mobile internet.
> In this case, SIP registration with TLS and media encryption with SRTP is
> working perfectly because I could see the secure symbol in Linphone *but* I
> could not hear any audio at all. After 30 seconds of the silent call, it
> hanged up automatically.
>
> *SIP Client on the 3G Mobile Internet > Freeswitch (tls:voip.test.com
> <http://voip.test.com>, SRTP) *
> *SIP Registration Sucess, Call No Media*
>
> My question is that is this the problem of NAT? If so, it would be great
> if you could help me solve this problem.
> Any help would be really appreciated.
>
> Best regards,
> Chhorm Chhatra
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From bipin at xbipin.com  Sat Feb  2 09:38:39 2019
From: bipin at xbipin.com (Bipin Patel)
Date: Sat, 02 Feb 2019 13:38:39 +0400
Subject: [Freeswitch-users] Freeswitch As SBC for sip_phones
In-Reply-To: <CAHtxdDcGvYPD9_OgxgakzBQKV9GD=sjHR2BtW57SWccN47EWgA@mail.gmail.com>
References: <CAG2rjpsb68CJo_B5RPsoBBmorYh8v-YvCXbmDNp2BsiHwXtJgQ@mail.gmail.com>
 <CAHtxdDcGvYPD9_OgxgakzBQKV9GD=sjHR2BtW57SWccN47EWgA@mail.gmail.com>
Message-ID: <168ad930118.279b.b07ebdf329620b8089087c7205b03f01@xbipin.com>

Actually we are using it as SBC but it won't directly send the registration 
as pass through, we use xml_curl and query the DB and if I'd/pass match 
then send same ahead to any other switch so this way the inbound 
registration credentials do go ahead



On February 2, 2019 2:20:24 AM Sergey Safarov <s.safarov at gmail.com> wrote:
> In this case you can try kamailio + rtpengine.
>
>
> пт, 1 февр. 2019 г., 19:35 Mehdi Shirazi <m.shirazi at gmail.com>:
> Hi
> I know that Freeswitch is a B2BUA and not proxy but I have seen SBC's based 
> on Freeswitch also used for sipphones and siptrunks.
> Is it possible to passthru sip registration with Freeswitch now ?
>
> Regards
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com

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From igor.potjevlesch at gmail.com  Sat Feb  2 10:33:13 2019
From: igor.potjevlesch at gmail.com (igor.potjevlesch at gmail.com)
Date: Sat, 2 Feb 2019 11:33:13 +0100
Subject: [Freeswitch-users] Issues with INVITE without SDP
Message-ID: <003701d4bae2$b3b5fa50$1b21eef0$@gmail.com>

Hello !

 

I'm facing two issues with scenarios with INVITE without SDP.

To support this behaviour towards the ingress peer, I activate the following
settings:

<param name="enable-3pcc" value="true"/> : per peer ; I also tried with
"proxy" instead of "true" but I'm not sure to clearly understand the
difference.

 

I also put this parameter on my internal interface. Everything is working
fine except that this setting bypass the B2BUA behaviour for media. I mean
that the IP and port for media are no longer my Freeswitch interfaces but
the gateways into the core network that finally handle the calls (but the
other parts of SDP are the one expected and configured on the Freeswitch
interface).

 

How can I have both support of INVITE without SDP and media proxy active
together?

 

The second issue still start with an in-dialog INVITE without SDP. But
instead of re-negotiating the SDP, FS sends the SDP as previous in 200OK.
Whereas we expect that FS send SDP with device capability. Moreover, I
expect to have the o-line version incremented.

 

Regards,

 

Igor.

 

 

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From mj310792 at gmail.com  Fri Feb  1 19:37:40 2019
From: mj310792 at gmail.com (mittali jangid)
Date: Sat, 2 Feb 2019 01:07:40 +0530
Subject: [Freeswitch-users] FIFO notify to all subscribed clients for single
	re-subscribe request
Message-ID: <CADnqS3PtsMh+ganWAXacrPnfm8gurOEb5Qsjv6th=MC6he+XYg@mail.gmail.com>

Hello,

We have two polycom phones configured with BLF of fifo-orbit extension
number as address.

When either of the phone re-register or re-subscribe, freeswitch generates
notify to all the subscribed phones.

For example we have FIFO orbit extension as 5900. Phone A and B have BLF
with address "queue+5900". When phone B sends subscribe to Freeswitch, it
responds with 202 followed by notify but also sends notify to phone A.

Is it functionality of Freeswitch or some sort of bug. ?

As if there are many FIFO addresses configured for a large system with such
multiple polycom phones, there will be flooding of notify, at cost of
resource utilization of server.
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From imfanee at gmail.com  Sat Feb  2 02:22:53 2019
From: imfanee at gmail.com (Faisal Hanif)
Date: Sat, 2 Feb 2019 07:22:53 +0500
Subject: [Freeswitch-users] Freeswitch As SBC for sip_phones
In-Reply-To: <CAG2rjpsb68CJo_B5RPsoBBmorYh8v-YvCXbmDNp2BsiHwXtJgQ@mail.gmail.com>
References: <CAG2rjpsb68CJo_B5RPsoBBmorYh8v-YvCXbmDNp2BsiHwXtJgQ@mail.gmail.com>
Message-ID: <CAO5BruAj1VCQo_rKt_GSiV7F=MuFkaYtbz6oTeW3xiLftEMhPQ@mail.gmail.com>

It is not actually passthrough but you can simulate the scenario by
creating a gateway for every user .

On Sat, Feb 2, 2019, 12:05 AM Mehdi Shirazi <m.shirazi at gmail.com wrote:

> Hi
> I know that Freeswitch is a B2BUA and not proxy but I have seen SBC's
> based on Freeswitch also used for sipphones and siptrunks.
> Is it possible to passthru sip registration with Freeswitch now ?
>
> Regards
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From imfanee at gmail.com  Sat Feb  2 02:26:30 2019
From: imfanee at gmail.com (Faisal Hanif)
Date: Sat, 2 Feb 2019 07:26:30 +0500
Subject: [Freeswitch-users] Verto not segregating domains
In-Reply-To: <etPan.5c5440f7.1dc9d55c.2c3@magicmail.mooo.com>
References: <etPan.5c5440f7.1dc9d55c.2c3@magicmail.mooo.com>
Message-ID: <CAO5BruCK=-a4aKttvyVP9ZOR_WkVkkLzmC+8iG+ZVx6-Ry-DTA@mail.gmail.com>

To separate conference s you can simply difference the context for
different groups

On Fri, Feb 1, 2019, 10:47 PM Rick Jarvis <rick at magicmail.mooo.com wrote:

> Ok I know this must be me confusing domains / contexts etc, but can’t work
> out where I’ve gone wrong.
>
> Situation: Verto, using xmlcurl from database server for directory. Verto
> users are user@ with multiple domains to segregate conferences.
>
> The xml being returned is below - I *thought* that simply having the
> <domain name=“whatever”> would keep things separate if the domains were
> different, but it seems this isn’t the case...
>
> <document type="freeswitch/xml">
>     <section name="directory">
>       <domain name="domain2">
>         <groups>
>           <group name="default">
>            <users>
>               <user id="user">
>                 <params>
>                   <param name="password" value="m6h20sj1y1pmysmayst3"/>
>                   <param name="vm-password" value="982375982735"/>
>                   <param name="jsonrpc-allowed-methods" value="verto"/>
>                   <param name="jsonrpc-allowed-event-channels"
> value="demo,conference,presence"/>
>                   <param name="jsonrpc-allowed-jsapi" value="all"/>
>                   <param name="jsonrpc-allowed-fsapi" value="all"/>
>                   <param name="verto-context" value="default"/>
>                   <param name="verto-dialplan" value="XML"/>
>                 </params>
>                 <variables>
>                   <variable name="toll_allow"
> value="domestic,international,local"/>
>                   <variable name="accountcode" value="user"/>
>                   <variable name="user_context" value="default"/>
>                   <variable name="effective_caller_id_name" value="user"/>
>                   <variable name="effective_caller_id_number"
> value="user"/>
>                   <variable name="outbound_caller_id_name" value="user"/>
>                   <variable name="outbound_caller_id_number" value="user"/>
>                 </variables>
>               </user>
>               </users>
>           </group>
>       </groups>
>   </domain>
> </section>
> </document>
>
> Output from verto status here shows users correctly on different domains,
> but maybe the prefix is what’s affecting it here? Or some other setting?
>
> default-v4::user at domain1 client                       x.x.x.x:43249 CONN_REG
> (WSS)
>
> default-v4::user at domain2 client                      y.y.y.y:55987 CONN_REG
> (WSS)
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From imfanee at gmail.com  Sat Feb  2 05:34:23 2019
From: imfanee at gmail.com (Faisal Hanif)
Date: Sat, 2 Feb 2019 10:34:23 +0500
Subject: [Freeswitch-users] No Media in SIP Calling from 3G Mobile
 Internet to Freeswitch
In-Reply-To: <CAGJLwFirQfQgx0Gk-O8GChoMhSbNJ_+67e-jSErbvpyNOXagEg@mail.gmail.com>
References: <CAJssr-iW9F5z4R+apRmu7SvLtR7yCL56wkWgwovzGSygfAtzYQ@mail.gmail.com>
 <CAGJLwFirQfQgx0Gk-O8GChoMhSbNJ_+67e-jSErbvpyNOXagEg@mail.gmail.com>
Message-ID: <CAO5BruACk1uf9bHdf3xreD+yWKDuoATyQ53dXU00ah3260icBw@mail.gmail.com>

Adding a stun can help too, if it is Linphone on Android then please add
stun.linphone.org in stun settings further disable AVFP & ICE

On Sat, Feb 2, 2019, 9:32 AM Alexey Sibyakin <alex at freeswitch.com wrote:

> Hi,
>
> Yes, that sounds like a network (NAT/Firewall) issue. Your mobile provider
> may use a chain of several NATs and not every one of them is working
> properly. Try VPN or another SIP client.
>
> Regards,
>
> Alex
>
> Alex Sibyakin | Support Engineer SignalWire | 228 Hamilton Ave 3rd Floor,
> Palo Alto, CA 94303 Email: alex at freeswitch.com Website:
> https://www.signalwire.com
>
>
>
> On Sat, Feb 2, 2019 at 2:44 AM Chhorm Chhatra <ch.chhatra at gmail.com>
> wrote:
>
>> Hi,
>> Currently, I face an issue that I could not hear the audio sound at all
>> when I try to call to FreeSWITCH (9386) via its public domain name from the
>> SIP client of my mobile phone with mobile internet.
>> In this case, SIP registration with TLS and media encryption with SRTP is
>> working perfectly because I could see the secure symbol in Linphone *but* I
>> could not hear any audio at all. After 30 seconds of the silent call, it
>> hanged up automatically.
>>
>> *SIP Client on the 3G Mobile Internet > Freeswitch (tls:voip.test.com
>> <http://voip.test.com>, SRTP) *
>> *SIP Registration Sucess, Call No Media*
>>
>> My question is that is this the problem of NAT? If so, it would be great
>> if you could help me solve this problem.
>> Any help would be really appreciated.
>>
>> Best regards,
>> Chhorm Chhatra
>> _________________________________________________________________________
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From rick at magicmail.mooo.com  Sun Feb  3 15:22:09 2019
From: rick at magicmail.mooo.com (Rick Jarvis)
Date: Sun, 3 Feb 2019 15:22:09 +0000
Subject: [Freeswitch-users] Verto not segregating domains
In-Reply-To: <CAGJLwFjhPD=VgRukWqtm_4Rc+1VoMZ+1u1MGh71-i_zQqYjjHQ@mail.gmail.com>
References: <etPan.5c5440f7.1dc9d55c.2c3@magicmail.mooo.com>
 <CAGJLwFjhPD=VgRukWqtm_4Rc+1VoMZ+1u1MGh71-i_zQqYjjHQ@mail.gmail.com>
Message-ID: <etPan.5c570721.b03330f.2c3@magicmail.mooo.com>

Yes, this is my veto config:

    <profile name="default-v4">
      <param name="bind-local" value="$${local_ip_v4}:8081"/>
      <param name="bind-local" value="$${local_ip_v4}:8082" secure="true"/>
      <!--<param name="force-register-domain" value="$${domain}"/>-->
      <param name="secure-combined" value="$${certs_dir}/wss.pem"/>
      <param name="secure-chain" value="$${certs_dir}/wss.pem"/>
      <param name="userauth" value="true"/>
      <!-- setting this to true will allow anyone to register even with no account so use with care -->
      <param name="blind-reg" value="false"/>
      <param name="mcast-ip" value="224.1.1.1"/>
      <param name="mcast-port" value="1337"/>
      <param name="rtp-ip" value="$${local_ip_v4}"/>
      <!--  <param name="ext-rtp-ip" value=""/> -->
      <param name="local-network" value="localnet.auto"/>
      <param name="outbound-codec-string" value="opus,vp8"/>
      <param name="inbound-codec-string" value="opus,vp8"/>

      <param name="apply-candidate-acl" value="localnet.auto"/>
      <param name="apply-candidate-acl" value="wan_v4.auto"/>
      <param name="apply-candidate-acl" value="rfc1918.auto"/>
      <param name="apply-candidate-acl" value="any_v4.auto"/>
      <param name="timer-name" value="soft"/>

    </profile>


From: Alexey Sibyakin <alex at freeswitch.com>
Reply: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Date: 2 February 2019 at 03:51:42
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject:  Re: [Freeswitch-users] Verto not segregating domains  

Hey,

Are you sure that

<param name="force-register-domain" value="$${domain}"/>

in verto.conf.xml is commented?

Regards,

Alex

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



On Sat, Feb 2, 2019 at 3:01 AM Rick Jarvis <rick at magicmail.mooo.com> wrote:
Ok I know this must be me confusing domains / contexts etc, but can’t work out where I’ve gone wrong.

Situation: Verto, using xmlcurl from database server for directory. Verto users are user@ with multiple domains to segregate conferences. 

The xml being returned is below - I *thought* that simply having the <domain name=“whatever”> would keep things separate if the domains were different, but it seems this isn’t the case...

<document type="freeswitch/xml">
    <section name="directory">
      <domain name="domain2">
        <groups>
          <group name="default">
           <users>
              <user id="user">
                <params>
                  <param name="password" value="m6h20sj1y1pmysmayst3"/>
                  <param name="vm-password" value="982375982735"/>
                  <param name="jsonrpc-allowed-methods" value="verto"/>
                  <param name="jsonrpc-allowed-event-channels" value="demo,conference,presence"/>
                  <param name="jsonrpc-allowed-jsapi" value="all"/>
                  <param name="jsonrpc-allowed-fsapi" value="all"/>
                  <param name="verto-context" value="default"/>
                  <param name="verto-dialplan" value="XML"/>
                </params>
                <variables>
                  <variable name="toll_allow" value="domestic,international,local"/>
                  <variable name="accountcode" value="user"/>
                  <variable name="user_context" value="default"/>
                  <variable name="effective_caller_id_name" value="user"/>
                  <variable name="effective_caller_id_number" value="user"/>
                  <variable name="outbound_caller_id_name" value="user"/>
                  <variable name="outbound_caller_id_number" value="user"/>
                </variables>
              </user>
              </users>
          </group>
      </groups>
  </domain>
</section>
</document>

Output from verto status here shows users correctly on different domains, but maybe the prefix is what’s affecting it here? Or some other setting?

default-v4::user at domain1 client                       x.x.x.x:43249 CONN_REG (WSS)
default-v4::user at domain2 client                      y.y.y.y:55987 CONN_REG (WSS)


_________________________________________________________________________
Professional FreeSWITCH Services
sales at freeswitch.com
https://freeswitch.com

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

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
https://freeswitch.com
_________________________________________________________________________
Professional FreeSWITCH Services
sales at freeswitch.com
https://freeswitch.com

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

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
https://freeswitch.com
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From rick at magicmail.mooo.com  Sun Feb  3 15:25:33 2019
From: rick at magicmail.mooo.com (Rick Jarvis)
Date: Sun, 3 Feb 2019 15:25:33 +0000
Subject: [Freeswitch-users] Verto not segregating domains
In-Reply-To: <CAO5BruCK=-a4aKttvyVP9ZOR_WkVkkLzmC+8iG+ZVx6-Ry-DTA@mail.gmail.com>
References: <etPan.5c5440f7.1dc9d55c.2c3@magicmail.mooo.com>
 <CAO5BruCK=-a4aKttvyVP9ZOR_WkVkkLzmC+8iG+ZVx6-Ry-DTA@mail.gmail.com>
Message-ID: <etPan.5c5707ed.3f5b411d.2c3@magicmail.mooo.com>

If I change the context to the same as the domain name, I get


 2019-02-03 15:17:27.471687 [WARNING] mod_dialplan_xml.c:667 Context domain2 not found


…because, I guess, the domains are being dynamically generated by the remote server and served over XML.

Is the solution then to somehow generate the contexts via XML? And why does changing the domain name not keep things separate, as I had thought?


From: Faisal Hanif <imfanee at gmail.com>
Reply: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Date: 2 February 2019 at 21:27:08
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject:  Re: [Freeswitch-users] Verto not segregating domains  

To separate conference s you can simply difference the context for different groups 

On Fri, Feb 1, 2019, 10:47 PM Rick Jarvis <rick at magicmail.mooo.com wrote:
Ok I know this must be me confusing domains / contexts etc, but can’t work out where I’ve gone wrong.

Situation: Verto, using xmlcurl from database server for directory. Verto users are user@ with multiple domains to segregate conferences. 

The xml being returned is below - I *thought* that simply having the <domain name=“whatever”> would keep things separate if the domains were different, but it seems this isn’t the case...

<document type="freeswitch/xml">
    <section name="directory">
      <domain name="domain2">
        <groups>
          <group name="default">
           <users>
              <user id="user">
                <params>
                  <param name="password" value="m6h20sj1y1pmysmayst3"/>
                  <param name="vm-password" value="982375982735"/>
                  <param name="jsonrpc-allowed-methods" value="verto"/>
                  <param name="jsonrpc-allowed-event-channels" value="demo,conference,presence"/>
                  <param name="jsonrpc-allowed-jsapi" value="all"/>
                  <param name="jsonrpc-allowed-fsapi" value="all"/>
                  <param name="verto-context" value="default"/>
                  <param name="verto-dialplan" value="XML"/>
                </params>
                <variables>
                  <variable name="toll_allow" value="domestic,international,local"/>
                  <variable name="accountcode" value="user"/>
                  <variable name="user_context" value="default"/>
                  <variable name="effective_caller_id_name" value="user"/>
                  <variable name="effective_caller_id_number" value="user"/>
                  <variable name="outbound_caller_id_name" value="user"/>
                  <variable name="outbound_caller_id_number" value="user"/>
                </variables>
              </user>
              </users>
          </group>
      </groups>
  </domain>
</section>
</document>

Output from verto status here shows users correctly on different domains, but maybe the prefix is what’s affecting it here? Or some other setting?

default-v4::user at domain1 client                       x.x.x.x:43249 CONN_REG (WSS)
default-v4::user at domain2 client                      y.y.y.y:55987 CONN_REG (WSS)


_________________________________________________________________________
Professional FreeSWITCH Services
sales at freeswitch.com
https://freeswitch.com

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

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
https://freeswitch.com
_________________________________________________________________________
Professional FreeSWITCH Services
sales at freeswitch.com
https://freeswitch.com

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

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
https://freeswitch.com
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From david.villasmil.work at gmail.com  Sun Feb  3 17:07:35 2019
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Sun, 3 Feb 2019 17:07:35 +0000
Subject: [Freeswitch-users] Freeswitch As SBC for sip_phones
In-Reply-To: <CAO5BruAj1VCQo_rKt_GSiV7F=MuFkaYtbz6oTeW3xiLftEMhPQ@mail.gmail.com>
References: <CAG2rjpsb68CJo_B5RPsoBBmorYh8v-YvCXbmDNp2BsiHwXtJgQ@mail.gmail.com>
 <CAO5BruAj1VCQo_rKt_GSiV7F=MuFkaYtbz6oTeW3xiLftEMhPQ@mail.gmail.com>
Message-ID: <CAFGRPVo8TU2Kmc9fX52qtHwArDN2tS3F-_aXyk+3cxbYmsaxXQ@mail.gmail.com>

You can’t relay Registrations. The second would be a completely new
registration. So you’d need to register the second ONLY when the first
leg(phone) registers. A mess. It’s simpler, easier and smarter to use a
proxy.
On Sat, 2 Feb 2019 at 21:36, Faisal Hanif <imfanee at gmail.com> wrote:

> It is not actually passthrough but you can simulate the scenario by
> creating a gateway for every user .
>
> On Sat, Feb 2, 2019, 12:05 AM Mehdi Shirazi <m.shirazi at gmail.com wrote:
>
>> Hi
>> I know that Freeswitch is a B2BUA and not proxy but I have seen SBC's
>> based on Freeswitch also used for sipphones and siptrunks.
>> Is it possible to passthru sip registration with Freeswitch now ?
>>
>> Regards
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From david.villasmil.work at gmail.com  Sun Feb  3 17:12:02 2019
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Sun, 3 Feb 2019 17:12:02 +0000
Subject: [Freeswitch-users] Issues with INVITE without SDP
In-Reply-To: <003701d4bae2$b3b5fa50$1b21eef0$@gmail.com>
References: <003701d4bae2$b3b5fa50$1b21eef0$@gmail.com>
Message-ID: <CAFGRPVprf0WXMeNoAXo7xoiQKQkWPHMsUJeJPmQ+AOJQUCFPpw@mail.gmail.com>

Have you seen
https://freeswitch.org/jira/si/jira.issueviews:issue-html/FS-6833/FS-6833.html

On Sat, 2 Feb 2019 at 11:22, <igor.potjevlesch at gmail.com> wrote:

> Hello !
>
>
>
> I'm facing two issues with scenarios with INVITE without SDP.
>
> To support this behaviour towards the ingress peer, I activate the
> following settings:
>
> <param name="enable-3pcc" value="true"/> : per peer ; I also tried with
> "proxy" instead of "true" but I'm not sure to clearly understand the
> difference.
>
>
>
> I also put this parameter on my internal interface. Everything is working
> fine except that this setting bypass the B2BUA behaviour for media. I mean
> that the IP and port for media are no longer my Freeswitch interfaces but
> the gateways into the core network that finally handle the calls (but the
> other parts of SDP are the one expected and configured on the Freeswitch
> interface).
>
>
>
> How can I have both support of INVITE without SDP and media proxy active
> together?
>
>
>
> The second issue still start with an in-dialog INVITE without SDP. But
> instead of re-negotiating the SDP, FS sends the SDP as previous in 200OK.
> Whereas we expect that FS send SDP with device capability. Moreover, I
> expect to have the o-line version incremented.
>
>
>
> Regards,
>
>
>
> Igor.
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From alex at freeswitch.com  Mon Feb  4 01:41:46 2019
From: alex at freeswitch.com (Alexey Sibyakin)
Date: Mon, 4 Feb 2019 10:41:46 +0900
Subject: [Freeswitch-users] Verto not segregating domains
In-Reply-To: <etPan.5c5707ed.3f5b411d.2c3@magicmail.mooo.com>
References: <etPan.5c5440f7.1dc9d55c.2c3@magicmail.mooo.com>
 <CAO5BruCK=-a4aKttvyVP9ZOR_WkVkkLzmC+8iG+ZVx6-Ry-DTA@mail.gmail.com>
 <etPan.5c5707ed.3f5b411d.2c3@magicmail.mooo.com>
Message-ID: <CAGJLwFiUruarGUSvd88u_6bc1vhoaF4VqBKEw4LHs9ndUGWQ_A@mail.gmail.com>

Sure,

mod_xml_curl you are using for your domains can do dialplan as well
https://freeswitch.org/confluence/display/FREESWITCH/mod_xml_curl

Regards,

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



On Mon, Feb 4, 2019 at 1:28 AM Rick Jarvis <rick at magicmail.mooo.com> wrote:

> If I change the context to the same as the domain name, I get
>
>
>  2019-02-03 15:17:27.471687 [WARNING] mod_dialplan_xml.c:667 Context
> domain2 not found
>
>
>
> …because, I guess, the domains are being dynamically generated by the
> remote server and served over XML.
>
> Is the solution then to somehow generate the contexts via XML? And why
> does changing the domain name not keep things separate, as I had thought?
>
>
> From: Faisal Hanif <imfanee at gmail.com> <imfanee at gmail.com>
> Reply: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> <freeswitch-users at lists.freeswitch.org>
> Date: 2 February 2019 at 21:27:08
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> <freeswitch-users at lists.freeswitch.org>
> Subject:  Re: [Freeswitch-users] Verto not segregating domains
>
> To separate conference s you can simply difference the context for
> different groups
>
> On Fri, Feb 1, 2019, 10:47 PM Rick Jarvis <rick at magicmail.mooo.com wrote:
>
>> Ok I know this must be me confusing domains / contexts etc, but can’t
>> work out where I’ve gone wrong.
>>
>> Situation: Verto, using xmlcurl from database server for directory. Verto
>> users are user@ with multiple domains to segregate conferences.
>>
>> The xml being returned is below - I *thought* that simply having the
>> <domain name=“whatever”> would keep things separate if the domains were
>> different, but it seems this isn’t the case...
>>
>> <document type="freeswitch/xml">
>>     <section name="directory">
>>       <domain name="domain2">
>>         <groups>
>>           <group name="default">
>>            <users>
>>               <user id="user">
>>                 <params>
>>                   <param name="password" value="m6h20sj1y1pmysmayst3"/>
>>                   <param name="vm-password" value="982375982735"/>
>>                   <param name="jsonrpc-allowed-methods" value="verto"/>
>>                   <param name="jsonrpc-allowed-event-channels"
>> value="demo,conference,presence"/>
>>                   <param name="jsonrpc-allowed-jsapi" value="all"/>
>>                   <param name="jsonrpc-allowed-fsapi" value="all"/>
>>                   <param name="verto-context" value="default"/>
>>                   <param name="verto-dialplan" value="XML"/>
>>                 </params>
>>                 <variables>
>>                   <variable name="toll_allow"
>> value="domestic,international,local"/>
>>                   <variable name="accountcode" value="user"/>
>>                   <variable name="user_context" value="default"/>
>>                   <variable name="effective_caller_id_name" value="user"/>
>>                   <variable name="effective_caller_id_number"
>> value="user"/>
>>                   <variable name="outbound_caller_id_name" value="user"/>
>>                   <variable name="outbound_caller_id_number"
>> value="user"/>
>>                 </variables>
>>               </user>
>>               </users>
>>           </group>
>>       </groups>
>>   </domain>
>> </section>
>> </document>
>>
>> Output from verto status here shows users correctly on different domains,
>> but maybe the prefix is what’s affecting it here? Or some other setting?
>>
>> default-v4::user at domain1 client                       x.x.x.x:43249
>> CONN_REG (WSS)
>>
>> default-v4::user at domain2 client                      y.y.y.y:55987
>> CONN_REG (WSS)
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From igor.potjevlesch at gmail.com  Mon Feb  4 14:33:19 2019
From: igor.potjevlesch at gmail.com (igor.potjevlesch at gmail.com)
Date: Mon, 4 Feb 2019 15:33:19 +0100
Subject: [Freeswitch-users] Issues with INVITE without SDP
In-Reply-To: <CAFGRPVprf0WXMeNoAXo7xoiQKQkWPHMsUJeJPmQ+AOJQUCFPpw@mail.gmail.com>
References: <003701d4bae2$b3b5fa50$1b21eef0$@gmail.com>
 <CAFGRPVprf0WXMeNoAXo7xoiQKQkWPHMsUJeJPmQ+AOJQUCFPpw@mail.gmail.com>
Message-ID: <001e01d4bc96$9304d230$b90e7690$@gmail.com>

Hi David,

 

Thank you!

I didn't but now I have read the post and I don't really understand the solution. I'm running 1.6.20.

 

Regards,

 

Igor.

 

De : FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> De la part de David Villasmil
Envoyé : dimanche 3 février 2019 18:12
À : FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Objet : Re: [Freeswitch-users] Issues with INVITE without SDP

 

Have you seen https://freeswitch.org/jira/si/jira.issueviews:issue-html/FS-6833/FS-6833.html

 

On Sat, 2 Feb 2019 at 11:22, <igor.potjevlesch at gmail.com <mailto:igor.potjevlesch at gmail.com> > wrote:

Hello !

 

I'm facing two issues with scenarios with INVITE without SDP.

To support this behaviour towards the ingress peer, I activate the following settings:

<param name="enable-3pcc" value="true"/> : per peer ; I also tried with "proxy" instead of "true" but I'm not sure to clearly understand the difference.

 

I also put this parameter on my internal interface. Everything is working fine except that this setting bypass the B2BUA behaviour for media. I mean that the IP and port for media are no longer my Freeswitch interfaces but the gateways into the core network that finally handle the calls (but the other parts of SDP are the one expected and configured on the Freeswitch interface).

 

How can I have both support of INVITE without SDP and media proxy active together?

 

The second issue still start with an in-dialog INVITE without SDP. But instead of re-negotiating the SDP, FS sends the SDP as previous in 200OK. Whereas we expect that FS send SDP with device capability. Moreover, I expect to have the o-line version incremented.

 

Regards,

 

Igor.

 

 

_________________________________________________________________________
Professional FreeSWITCH Services
sales at freeswitch.com <mailto:sales at freeswitch.com> 
https://freeswitch.com

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

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org> 
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From gmaruzz at gmail.com  Mon Feb  4 14:58:26 2019
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Mon, 4 Feb 2019 15:58:26 +0100
Subject: [Freeswitch-users] TLS traffic visualization HOWTO
Message-ID: <CALXCt0pZsTBMZxfCTafy+9kpGx5hOJhX3HyQ22e3BLt9LFXuPw@mail.gmail.com>

# How to TRACE and visualize TLS and non-TLS SIP traffic in real time
(thanks to Homer's Lorenzo Mangani for pointing me toward Frida)

apt-get install python-pip
pip install frida
pip install hexdump
wget https://raw.githubusercontent.com/google/ssl_logger/master/ssl_logger.py
<https://raw.githubusercontent.com/>

#first ssh terminal
# create fifo pipe, then will send the content from fifo pipe to an
sngrep without gui, which will be reading pcap from stdin, and sending
eep packets to the other sngrep (third terminal)

mkfifo /tmp/pipe
cat /tmp/pipe | sngrep -N -q -H udp:127.0.0.1:5077 -I -

#second ssh terminal
# writes as pcap to fifo pipe what freeswitch writes and reads from ssl lib

python ssl_logger_giova.py -pcap /tmp/pipe freeswitch

#third ssh terminal
# sngrep that receives packets from both the Ethernet device, and the
eep packets sent by the other sngrep (eg, the tls packets ssl_logger
grabs from freeswitch's ssl lib)

sngrep -L udp:127.0.0.1:5077

(you may want to edit ssl_logger.py and change 228 to be 101 -
LINKTYPE_IPV4 to be LINKTYPE_RAW )




-- 
Sincerely,

Giovanni Maruzzelli
OpenTelecom.IT
cell: +39 347 266 56 18
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From rick at magicmail.mooo.com  Mon Feb  4 19:18:09 2019
From: rick at magicmail.mooo.com (Rick Jarvis)
Date: Mon, 4 Feb 2019 19:18:09 +0000
Subject: [Freeswitch-users] Verto not segregating domains
In-Reply-To: <CAGJLwFiUruarGUSvd88u_6bc1vhoaF4VqBKEw4LHs9ndUGWQ_A@mail.gmail.com>
References: <etPan.5c5440f7.1dc9d55c.2c3@magicmail.mooo.com>
 <CAO5BruCK=-a4aKttvyVP9ZOR_WkVkkLzmC+8iG+ZVx6-Ry-DTA@mail.gmail.com>
 <etPan.5c5707ed.3f5b411d.2c3@magicmail.mooo.com>
 <CAGJLwFiUruarGUSvd88u_6bc1vhoaF4VqBKEw4LHs9ndUGWQ_A@mail.gmail.com>
Message-ID: <etPan.5c588ff1.68636df.2c3@magicmail.mooo.com>

Thanks. Ok, so I figured as a ‘quick fix’ until I can get that figured out, I would add the context to the conference name (coming from a ‘mycontext’ var previously set). But, the change just doesn’t take! This is the dial plan entry:

    <extension name="conferences_new">
      <condition field="destination_number" expression="^CONF_(.+)">
        <action application="log" data="CRIT joining $1_${mycontext}" /> <!-- correctly shows 'joining user_domain2' -->
        <action application="answer"/>
        <action application="conference" data="$1_${mycontext}”/> <!-- shows ‘conference user_conf successfully joined’, which was the old conference name format :( -->
      </condition>
    </extension>

I’ve reloaded xml (obvs) and even restarted fs completely, but it’s still reflecting the old conference names! Driving me mad… is verto bypassing the conference name value and picking it up from somewhere else?


From: Alexey Sibyakin <alex at freeswitch.com>
Reply: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Date: 4 February 2019 at 01:43:05
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject:  Re: [Freeswitch-users] Verto not segregating domains  

Sure,

mod_xml_curl you are using for your domains can do dialplan as well
https://freeswitch.org/confluence/display/FREESWITCH/mod_xml_curl

Regards,

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



On Mon, Feb 4, 2019 at 1:28 AM Rick Jarvis <rick at magicmail.mooo.com> wrote:
If I change the context to the same as the domain name, I get


 2019-02-03 15:17:27.471687 [WARNING] mod_dialplan_xml.c:667 Context domain2 not found


…because, I guess, the domains are being dynamically generated by the remote server and served over XML.

Is the solution then to somehow generate the contexts via XML? And why does changing the domain name not keep things separate, as I had thought?


From: Faisal Hanif <imfanee at gmail.com>
Reply: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Date: 2 February 2019 at 21:27:08
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject:  Re: [Freeswitch-users] Verto not segregating domains

To separate conference s you can simply difference the context for different groups 

On Fri, Feb 1, 2019, 10:47 PM Rick Jarvis <rick at magicmail.mooo.com wrote:
Ok I know this must be me confusing domains / contexts etc, but can’t work out where I’ve gone wrong.

Situation: Verto, using xmlcurl from database server for directory. Verto users are user@ with multiple domains to segregate conferences. 

The xml being returned is below - I *thought* that simply having the <domain name=“whatever”> would keep things separate if the domains were different, but it seems this isn’t the case...

<document type="freeswitch/xml">
    <section name="directory">
      <domain name="domain2">
        <groups>
          <group name="default">
           <users>
              <user id="user">
                <params>
                  <param name="password" value="m6h20sj1y1pmysmayst3"/>
                  <param name="vm-password" value="982375982735"/>
                  <param name="jsonrpc-allowed-methods" value="verto"/>
                  <param name="jsonrpc-allowed-event-channels" value="demo,conference,presence"/>
                  <param name="jsonrpc-allowed-jsapi" value="all"/>
                  <param name="jsonrpc-allowed-fsapi" value="all"/>
                  <param name="verto-context" value="default"/>
                  <param name="verto-dialplan" value="XML"/>
                </params>
                <variables>
                  <variable name="toll_allow" value="domestic,international,local"/>
                  <variable name="accountcode" value="user"/>
                  <variable name="user_context" value="default"/>
                  <variable name="effective_caller_id_name" value="user"/>
                  <variable name="effective_caller_id_number" value="user"/>
                  <variable name="outbound_caller_id_name" value="user"/>
                  <variable name="outbound_caller_id_number" value="user"/>
                </variables>
              </user>
              </users>
          </group>
      </groups>
  </domain>
</section>
</document>

Output from verto status here shows users correctly on different domains, but maybe the prefix is what’s affecting it here? Or some other setting?

default-v4::user at domain1 client                       x.x.x.x:43249 CONN_REG (WSS)
default-v4::user at domain2 client                      y.y.y.y:55987 CONN_REG (WSS)


_________________________________________________________________________
Professional FreeSWITCH Services
sales at freeswitch.com
https://freeswitch.com

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

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
https://freeswitch.com
_________________________________________________________________________
Professional FreeSWITCH Services
sales at freeswitch.com
https://freeswitch.com

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

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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_________________________________________________________________________
Professional FreeSWITCH Services
sales at freeswitch.com
https://freeswitch.com

Official FreeSWITCH Sites
https://freeswitch.com/oss
https://freeswitch.org/confluence
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FreeSWITCH-users at lists.freeswitch.org
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_________________________________________________________________________
Professional FreeSWITCH Services
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Official FreeSWITCH Sites
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From kaiduanx at yahoo.ca  Mon Feb  4 21:56:45 2019
From: kaiduanx at yahoo.ca (kaiduan xie)
Date: Mon, 4 Feb 2019 21:56:45 +0000 (UTC)
Subject: [Freeswitch-users] DTMF negotiation
In-Reply-To: <CAGJLwFgFMCsT2hdBxkx432YBe=AV45XHxpmf4_oKotNYPEapaQ@mail.gmail.com>
References: <510252213.6154862.1519830130424.ref@mail.yahoo.com>
 <510252213.6154862.1519830130424@mail.yahoo.com>
 <CAGJLwFiycm-ePQVpA_9EshvrgNAr+JKtU_MxzchA-CKq43-+TA@mail.gmail.com>
 <188910919.6866630.1519918789572@mail.yahoo.com>
 <CAGJLwFgFMCsT2hdBxkx432YBe=AV45XHxpmf4_oKotNYPEapaQ@mail.gmail.com>
Message-ID: <1875291968.3278986.1549317405765@mail.yahoo.com>

 
Alex,
Can you show the configuration example in vanilla configuration? I could not find out the example. To refresh your memory, I need to setup Freeswitch to send DTMF based on 200 SDP answer in B call leg.
1. If answer supports telephone event, then FreeSwitch sends DTMF in rfc2833.
2. If answer does not support telephone event, then FreeSwitch sends DTMF inband.
The call is a bridged call and DTMF always comes from A call leg in rfc2833.
Many thanks for help,
/Kaiduan

    On Tuesday, March 6, 2018, 6:28:22 p.m. EST, Alexey Sibyakin <alex at freeswitch.com> wrote:  
 
 Yes, it's possible. Vanilla config has example.
Alex
On Fri, Mar 2, 2018 at 12:39 AM, kaiduan xie <kaiduanx at yahoo.ca> wrote:

Alex,
Thanks for the help.
With <param name="dtmf-type" value="none"/> in sofia profile, the offer from FreeSwitch does not add telephone event support.

With <action application="set" data="execute_on_media=start_ dtmf_generate"/>, FreeSwitch always sends DTMF inband even the offer from FreeSwitch and answer both supports telephone event.
What I want is DTMF negotiation as below,
FreeSwitch always adds telephone event support in offer,
1. If answer supports telephone event, then FreeSwitch sends DTMF in rfc2833.2. If answer does not support telephone event, then FreeSwitch sends DTMF inband.
Is it possible?
Thanks again.
/Kaiduan
    On Thursday, March 1, 2018 3:22 AM, Alexey Sibyakin <alex at freeswitch.com> wrote:
 

 Hi,
You need <action application="set" data="execute_on_media=start_ dtmf_generate"/>in dialplan and probably<param name="dtmf-type" value="none"/>
in sofia profile
Alex
On Thu, Mar 1, 2018 at 12:02 AM, kaiduan xie <kaiduanx at yahoo.ca> wrote:

Hi,
We are encountering the following DTMF issue,
FreeSwitch adds telephone-event support in INVITE to SBC, SBC replies back without telephone-event support in 200. 
However FreeSwitch still sends out DTMF in RFC 2833 event.
I think FreeSwitch should send out DTMF inband instead of 2833 event, what is the configuration for this?
Many thanks for help,
/Kaiduan
______________________________ ______________________________ _____________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www. freeswitchsolutions.com

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

FreeSWITCH-users mailing list
FreeSWITCH-users at lists. freeswitch.org
http://lists.freeswitch.org/ mailman/listinfo/freeswitch- users
UNSUBSCRIBE:http://lists. freeswitch.org/mailman/ options/freeswitch-users
http://www.freeswitch.org




-- 

Alex Sibyakin | Support EngineerFreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045Email: alex at freeswitch.comWebsite: https://www.FreeSWITCH.comNeed commercial support? Contact sales at freeswitch.com f or details.
______________________________ ______________________________ _____________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www. freeswitchsolutions.com

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

FreeSWITCH-users mailing list
FreeSWITCH-users at 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




-- 

Alex Sibyakin | Support EngineerFreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045Email: alex at freeswitch.comWebsite: https://www.FreeSWITCH.comNeed commercial support? Contact sales at freeswitch.com for details.
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com

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

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org  
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From rick at magicmail.mooo.com  Tue Feb  5 19:56:12 2019
From: rick at magicmail.mooo.com (Rick Jarvis)
Date: Tue, 5 Feb 2019 19:56:12 +0000
Subject: [Freeswitch-users] IPv6
Message-ID: <etPan.5c59ea5c.59c3ee71.2c3@magicmail.mooo.com>

Looking for some help with FreeSWITCH in a semi-v6 situation… I’ll explain what I mean:

	VoIP handsets on private IPv4 network -> NAT to IPv6 Internet address -> IPv4 public FreeSWITCH server

Not working, and few too many variables to get my head around. I guess whether there is an ALG on the the private router is still an issue as it’s IPv4 at that point. But should the public IPv6 address speak ok to the public IPv4 FreeSWITCH server, or would it help if I give it an IPv6 address too? (Don’t really understand how the 6->4 thing works, it’s voodoo! :)

Thanks, R
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From brian at freeswitch.com  Tue Feb  5 20:08:49 2019
From: brian at freeswitch.com (Brian West)
Date: Tue, 5 Feb 2019 14:08:49 -0600
Subject: [Freeswitch-users] IPv6
In-Reply-To: <etPan.5c59ea5c.59c3ee71.2c3@magicmail.mooo.com>
References: <etPan.5c59ea5c.59c3ee71.2c3@magicmail.mooo.com>
Message-ID: <CAEJMVkDgoajr3bkB_jWz58C0wrLhkVQ4RBNzKNvZKdQNSrY9JA@mail.gmail.com>

Please say that topology isn't so... really?  Chances are it won't.

On Tue, Feb 5, 2019 at 1:56 PM Rick Jarvis <rick at magicmail.mooo.com> wrote:

> Looking for some help with FreeSWITCH in a semi-v6 situation… I’ll explain
> what I mean:
>
> VoIP handsets on private IPv4 network -> NAT to IPv6 Internet address ->
> IPv4 public FreeSWITCH server
>
> Not working, and few too many variables to get my head around. I guess
> whether there is an ALG on the the private router is still an issue as it’s
> IPv4 at that point. But should the public IPv6 address speak ok to the
> public IPv4 FreeSWITCH server, or would it help if I give it an IPv6
> address too? (Don’t really understand how the 6->4 thing works, it’s
> voodoo! :)
>
> Thanks, R
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com



-- 

Brian West | Co-founder and Developer

Need Commercial support? email sales at freeswitch.com

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

Email: brian at freeswitch.com

Mobile: 918-424-9378

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

[image: https://www.facebook.com/signalwireinc?src=email]
<https://www.facebook.com/freeswitch> [image:
https://twitter.com/freeswitch] <https://twitter.com/freeswitch>
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From rick at magicmail.mooo.com  Tue Feb  5 20:53:52 2019
From: rick at magicmail.mooo.com (Rick Jarvis)
Date: Tue, 5 Feb 2019 20:53:52 +0000
Subject: [Freeswitch-users] IPv6
In-Reply-To: <CAEJMVkDgoajr3bkB_jWz58C0wrLhkVQ4RBNzKNvZKdQNSrY9JA@mail.gmail.com>
References: <etPan.5c59ea5c.59c3ee71.2c3@magicmail.mooo.com>
 <CAEJMVkDgoajr3bkB_jWz58C0wrLhkVQ4RBNzKNvZKdQNSrY9JA@mail.gmail.com>
Message-ID: <etPan.5c59f7e0.77702f5c.2c3@magicmail.mooo.com>

That’s how it is at the moment, but what do I need to do to fix it? Will putting a v6 IP on FreeSWITCH (and an associated AAAA record) help? Or is the problem that the handsets are on a private v4 network? Or both?!


From: Brian West <brian at freeswitch.com>
Reply: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Date: 5 February 2019 at 20:09:42
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject:  Re: [Freeswitch-users] IPv6  

Please say that topology isn't so... really?  Chances are it won't.

On Tue, Feb 5, 2019 at 1:56 PM Rick Jarvis <rick at magicmail.mooo.com> wrote:
Looking for some help with FreeSWITCH in a semi-v6 situation… I’ll explain what I mean:

VoIP handsets on private IPv4 network -> NAT to IPv6 Internet address -> IPv4 public FreeSWITCH server

Not working, and few too many variables to get my head around. I guess whether there is an ALG on the the private router is still an issue as it’s IPv4 at that point. But should the public IPv6 address speak ok to the public IPv4 FreeSWITCH server, or would it help if I give it an IPv6 address too? (Don’t really understand how the 6->4 thing works, it’s voodoo! :)

Thanks, R
_________________________________________________________________________
Professional FreeSWITCH Services
sales at freeswitch.com
https://freeswitch.com

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

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


--


Brian West | Co-founder and
Developer

Need Commercial support? email sales at freeswitch.com

FreeSWITCH Solutions | 17345 Civic Drive
#2531 Brookfield, WI 53045

Email:  
brian at freeswitch.com

Mobile: 918-424-9378

Website:  
https://www.FreeSWITCH.com

  

_________________________________________________________________________
Professional FreeSWITCH Services
sales at freeswitch.com
https://freeswitch.com

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

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
https://freeswitch.com
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From brian at freeswitch.com  Wed Feb  6 00:17:58 2019
From: brian at freeswitch.com (Brian West)
Date: Tue, 5 Feb 2019 18:17:58 -0600
Subject: [Freeswitch-users] IPv6
In-Reply-To: <etPan.5c59f7e0.77702f5c.2c3@magicmail.mooo.com>
References: <etPan.5c59ea5c.59c3ee71.2c3@magicmail.mooo.com>
 <CAEJMVkDgoajr3bkB_jWz58C0wrLhkVQ4RBNzKNvZKdQNSrY9JA@mail.gmail.com>
 <etPan.5c59f7e0.77702f5c.2c3@magicmail.mooo.com>
Message-ID: <CAEJMVkD1wbJLXogQj5L9u1XWRZYBLBGq6wusoX8HZxAeXUXe8Q@mail.gmail.com>

Not currently designed to cross the streams in that manner, 464xlate is
probably what you need, I've never had the time to really set that up, I
have seen it work on T-Mobile on Bria, where the handset is pure v6 and it
arrives the server all v4'ed up.

On Tue, Feb 5, 2019 at 2:54 PM Rick Jarvis <rick at magicmail.mooo.com> wrote:

> That’s how it is at the moment, but what do I need to do to fix it? Will
> putting a v6 IP on FreeSWITCH (and an associated AAAA record) help? Or is
> the problem that the handsets are on a private v4 network? Or both?!
>
>
> From: Brian West <brian at freeswitch.com> <brian at freeswitch.com>
> Reply: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> <freeswitch-users at lists.freeswitch.org>
> Date: 5 February 2019 at 20:09:42
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> <freeswitch-users at lists.freeswitch.org>
> Subject:  Re: [Freeswitch-users] IPv6
>
> Please say that topology isn't so... really?  Chances are it won't.
>
> On Tue, Feb 5, 2019 at 1:56 PM Rick Jarvis <rick at magicmail.mooo.com>
> wrote:
>
>> Looking for some help with FreeSWITCH in a semi-v6 situation… I’ll
>> explain what I mean:
>>
>> VoIP handsets on private IPv4 network -> NAT to IPv6 Internet address ->
>> IPv4 public FreeSWITCH server
>>
>> Not working, and few too many variables to get my head around. I guess
>> whether there is an ALG on the the private router is still an issue as it’s
>> IPv4 at that point. But should the public IPv6 address speak ok to the
>> public IPv4 FreeSWITCH server, or would it help if I give it an IPv6
>> address too? (Don’t really understand how the 6->4 thing works, it’s
>> voodoo! :)
>>
>> Thanks, R
>> _________________________________________________________________________
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
>
>
> --
>
> Brian West | Co-founder and Developer
>
> Need Commercial support? email sales at freeswitch.com
>
> FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045
> <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g>
>
> Email: brian at freeswitch.com
>
> Mobile: 918-424-9378
>
> Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/>
>
> [image: https://www.facebook.com/signalwireinc?src=email]
> <https://www.facebook.com/freeswitch> [image:
> https://twitter.com/freeswitch] <https://twitter.com/freeswitch>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com



-- 

Brian West | Co-founder and Developer

Need Commercial support? email sales at freeswitch.com

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

Email: brian at freeswitch.com

Mobile: 918-424-9378

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

[image: https://www.facebook.com/signalwireinc?src=email]
<https://www.facebook.com/freeswitch> [image:
https://twitter.com/freeswitch] <https://twitter.com/freeswitch>
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From jurijs.ivolga at gmail.com  Wed Feb  6 10:20:25 2019
From: jurijs.ivolga at gmail.com (Jurijs Ivolga)
Date: Wed, 6 Feb 2019 12:20:25 +0200
Subject: [Freeswitch-users] IPv6
In-Reply-To: <CAEJMVkD1wbJLXogQj5L9u1XWRZYBLBGq6wusoX8HZxAeXUXe8Q@mail.gmail.com>
References: <etPan.5c59ea5c.59c3ee71.2c3@magicmail.mooo.com>
 <CAEJMVkDgoajr3bkB_jWz58C0wrLhkVQ4RBNzKNvZKdQNSrY9JA@mail.gmail.com>
 <etPan.5c59f7e0.77702f5c.2c3@magicmail.mooo.com>
 <CAEJMVkD1wbJLXogQj5L9u1XWRZYBLBGq6wusoX8HZxAeXUXe8Q@mail.gmail.com>
Message-ID: <CAOrA2UY+meDdHNhG8TK=oL-vZYcNDodWOMKe9QAPoNApKU-jng@mail.gmail.com>

Hi,

Do you see a call arriving to Freeswitch?

I was able to make Freeswitch work with NAT64 and in this case call is
arriving to Freeswitch, only there was a problem that in SDP is IPv6
address. But I'm not sure that this is your case.

Jurijs


On Wed, Feb 6, 2019 at 2:18 AM Brian West <brian at freeswitch.com> wrote:

> Not currently designed to cross the streams in that manner, 464xlate is
> probably what you need, I've never had the time to really set that up, I
> have seen it work on T-Mobile on Bria, where the handset is pure v6 and it
> arrives the server all v4'ed up.
>
> On Tue, Feb 5, 2019 at 2:54 PM Rick Jarvis <rick at magicmail.mooo.com>
> wrote:
>
>> That’s how it is at the moment, but what do I need to do to fix it? Will
>> putting a v6 IP on FreeSWITCH (and an associated AAAA record) help? Or is
>> the problem that the handsets are on a private v4 network? Or both?!
>>
>>
>> From: Brian West <brian at freeswitch.com> <brian at freeswitch.com>
>> Reply: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> <freeswitch-users at lists.freeswitch.org>
>> Date: 5 February 2019 at 20:09:42
>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> <freeswitch-users at lists.freeswitch.org>
>> Subject:  Re: [Freeswitch-users] IPv6
>>
>> Please say that topology isn't so... really?  Chances are it won't.
>>
>> On Tue, Feb 5, 2019 at 1:56 PM Rick Jarvis <rick at magicmail.mooo.com>
>> wrote:
>>
>>> Looking for some help with FreeSWITCH in a semi-v6 situation… I’ll
>>> explain what I mean:
>>>
>>> VoIP handsets on private IPv4 network -> NAT to IPv6 Internet address ->
>>> IPv4 public FreeSWITCH server
>>>
>>> Not working, and few too many variables to get my head around. I guess
>>> whether there is an ALG on the the private router is still an issue as it’s
>>> IPv4 at that point. But should the public IPv6 address speak ok to the
>>> public IPv4 FreeSWITCH server, or would it help if I give it an IPv6
>>> address too? (Don’t really understand how the 6->4 thing works, it’s
>>> voodoo! :)
>>>
>>> Thanks, R
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Services
>>> sales at freeswitch.com
>>> https://freeswitch.com
>>>
>>> Official FreeSWITCH Sites
>>> https://freeswitch.com/oss
>>> https://freeswitch.org/confluence
>>> https://cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> https://freeswitch.com
>>
>>
>>
>> --
>>
>> Brian West | Co-founder and Developer
>>
>> Need Commercial support? email sales at freeswitch.com
>>
>> FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045
>> <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g>
>>
>> Email: brian at freeswitch.com
>>
>> Mobile: 918-424-9378
>>
>> Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/>
>>
>> [image: https://www.facebook.com/signalwireinc?src=email]
>> <https://www.facebook.com/freeswitch> [image:
>> https://twitter.com/freeswitch] <https://twitter.com/freeswitch>
>> _________________________________________________________________________
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
>
>
> --
>
> Brian West | Co-founder and Developer
>
> Need Commercial support? email sales at freeswitch.com
>
> FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045
> <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g>
>
> Email: brian at freeswitch.com
>
> Mobile: 918-424-9378
>
> Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/>
>
> [image: https://www.facebook.com/signalwireinc?src=email]
> <https://www.facebook.com/freeswitch> [image:
> https://twitter.com/freeswitch] <https://twitter.com/freeswitch>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From rick at magicmail.mooo.com  Wed Feb  6 10:49:58 2019
From: rick at magicmail.mooo.com (Rick Jarvis)
Date: Wed, 6 Feb 2019 10:49:58 +0000
Subject: [Freeswitch-users] IPv6
In-Reply-To: <CAOrA2UY+meDdHNhG8TK=oL-vZYcNDodWOMKe9QAPoNApKU-jng@mail.gmail.com>
References: <etPan.5c59ea5c.59c3ee71.2c3@magicmail.mooo.com>
 <CAEJMVkDgoajr3bkB_jWz58C0wrLhkVQ4RBNzKNvZKdQNSrY9JA@mail.gmail.com>
 <etPan.5c59f7e0.77702f5c.2c3@magicmail.mooo.com>
 <CAEJMVkD1wbJLXogQj5L9u1XWRZYBLBGq6wusoX8HZxAeXUXe8Q@mail.gmail.com>
 <CAOrA2UY+meDdHNhG8TK=oL-vZYcNDodWOMKe9QAPoNApKU-jng@mail.gmail.com>
Message-ID: <etPan.5c5abbd6.210657a1.2c3@magicmail.mooo.com>

At this stage we’re having problems with the registration (not helped by the fact these Polycoms don’t seem to give much indication of what they’re doing, without looking at the logs or actually being there).

Sadly I don’t know anything about IPv6 (apart from an elevator pitch), so I’m only guessing at what NAT64 and 464xlate is!


From: Jurijs Ivolga <jurijs.ivolga at gmail.com>
Reply: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Date: 6 February 2019 at 10:21:20
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject:  Re: [Freeswitch-users] IPv6  

Hi,

Do you see a call arriving to Freeswitch?

I was able to make Freeswitch work with NAT64 and in this case call is arriving to Freeswitch, only there was a problem that in SDP is IPv6 address. But I'm not sure that this is your case.

Jurijs


On Wed, Feb 6, 2019 at 2:18 AM Brian West <brian at freeswitch.com> wrote:
Not currently designed to cross the streams in that manner, 464xlate is probably what you need, I've never had the time to really set that up, I have seen it work on T-Mobile on Bria, where the handset is pure v6 and it arrives the server all v4'ed up.

On Tue, Feb 5, 2019 at 2:54 PM Rick Jarvis <rick at magicmail.mooo.com> wrote:
That’s how it is at the moment, but what do I need to do to fix it? Will putting a v6 IP on FreeSWITCH (and an associated AAAA record) help? Or is the problem that the handsets are on a private v4 network? Or both?!


From: Brian West <brian at freeswitch.com>
Reply: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Date: 5 February 2019 at 20:09:42
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject:  Re: [Freeswitch-users] IPv6

Please say that topology isn't so... really?  Chances are it won't.

On Tue, Feb 5, 2019 at 1:56 PM Rick Jarvis <rick at magicmail.mooo.com> wrote:
Looking for some help with FreeSWITCH in a semi-v6 situation… I’ll explain what I mean:

VoIP handsets on private IPv4 network -> NAT to IPv6 Internet address -> IPv4 public FreeSWITCH server

Not working, and few too many variables to get my head around. I guess whether there is an ALG on the the private router is still an issue as it’s IPv4 at that point. But should the public IPv6 address speak ok to the public IPv4 FreeSWITCH server, or would it help if I give it an IPv6 address too? (Don’t really understand how the 6->4 thing works, it’s voodoo! :)

Thanks, R
_________________________________________________________________________
Professional FreeSWITCH Services
sales at freeswitch.com
https://freeswitch.com

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

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


--


Brian West | Co-founder and
Developer

Need Commercial support? email sales at freeswitch.com

FreeSWITCH Solutions | 17345 Civic Drive
#2531 Brookfield, WI 53045

Email:  
brian at freeswitch.com

Mobile: 918-424-9378

Website:  
https://www.FreeSWITCH.com

  

_________________________________________________________________________
Professional FreeSWITCH Services
sales at freeswitch.com
https://freeswitch.com

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

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
https://freeswitch.com
_________________________________________________________________________
Professional FreeSWITCH Services
sales at freeswitch.com
https://freeswitch.com

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

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


--


Brian West | Co-founder and
Developer

Need Commercial support? email sales at freeswitch.com

FreeSWITCH Solutions | 17345 Civic Drive
#2531 Brookfield, WI 53045

Email:  
brian at freeswitch.com

Mobile: 918-424-9378

Website:  
https://www.FreeSWITCH.com

  

_________________________________________________________________________
Professional FreeSWITCH Services
sales at freeswitch.com
https://freeswitch.com

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

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
https://freeswitch.com
_________________________________________________________________________
Professional FreeSWITCH Services
sales at freeswitch.com
https://freeswitch.com

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

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
https://freeswitch.com
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From jurijs.ivolga at gmail.com  Wed Feb  6 11:44:19 2019
From: jurijs.ivolga at gmail.com (Jurijs Ivolga)
Date: Wed, 6 Feb 2019 13:44:19 +0200
Subject: [Freeswitch-users] IPv6
In-Reply-To: <etPan.5c5abbd6.210657a1.2c3@magicmail.mooo.com>
References: <etPan.5c59ea5c.59c3ee71.2c3@magicmail.mooo.com>
 <CAEJMVkDgoajr3bkB_jWz58C0wrLhkVQ4RBNzKNvZKdQNSrY9JA@mail.gmail.com>
 <etPan.5c59f7e0.77702f5c.2c3@magicmail.mooo.com>
 <CAEJMVkD1wbJLXogQj5L9u1XWRZYBLBGq6wusoX8HZxAeXUXe8Q@mail.gmail.com>
 <CAOrA2UY+meDdHNhG8TK=oL-vZYcNDodWOMKe9QAPoNApKU-jng@mail.gmail.com>
 <etPan.5c5abbd6.210657a1.2c3@magicmail.mooo.com>
Message-ID: <CAOrA2UZp=6H-D6VF0badO+Vdj4zAhSH+W8rkkccmPLNSD8YEhQ@mail.gmail.com>

Hi,

Not sure that I will be able to help you, but have you checked freeswitch
console for logs? What errors you get? A shot in the dark, but TLS might
help you in case if some routers altering your SIP messages on the way.

Jurijs


On Wed, Feb 6, 2019 at 12:50 PM Rick Jarvis <rick at magicmail.mooo.com> wrote:

> At this stage we’re having problems with the registration (not helped by
> the fact these Polycoms don’t seem to give much indication of what they’re
> doing, without looking at the logs or actually being there).
>
> Sadly I don’t know anything about IPv6 (apart from an elevator pitch), so
> I’m only guessing at what NAT64 and 464xlate is!
>
>
> From: Jurijs Ivolga <jurijs.ivolga at gmail.com> <jurijs.ivolga at gmail.com>
> Reply: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> <freeswitch-users at lists.freeswitch.org>
> Date: 6 February 2019 at 10:21:20
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> <freeswitch-users at lists.freeswitch.org>
> Subject:  Re: [Freeswitch-users] IPv6
>
> Hi,
>
> Do you see a call arriving to Freeswitch?
>
> I was able to make Freeswitch work with NAT64 and in this case call is
> arriving to Freeswitch, only there was a problem that in SDP is IPv6
> address. But I'm not sure that this is your case.
>
> Jurijs
>
>
> On Wed, Feb 6, 2019 at 2:18 AM Brian West <brian at freeswitch.com> wrote:
>
>> Not currently designed to cross the streams in that manner, 464xlate is
>> probably what you need, I've never had the time to really set that up, I
>> have seen it work on T-Mobile on Bria, where the handset is pure v6 and it
>> arrives the server all v4'ed up.
>>
>> On Tue, Feb 5, 2019 at 2:54 PM Rick Jarvis <rick at magicmail.mooo.com>
>> wrote:
>>
>>> That’s how it is at the moment, but what do I need to do to fix it? Will
>>> putting a v6 IP on FreeSWITCH (and an associated AAAA record) help? Or is
>>> the problem that the handsets are on a private v4 network? Or both?!
>>>
>>>
>>> From: Brian West <brian at freeswitch.com> <brian at freeswitch.com>
>>> Reply: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>> <freeswitch-users at lists.freeswitch.org>
>>> Date: 5 February 2019 at 20:09:42
>>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>> <freeswitch-users at lists.freeswitch.org>
>>> Subject:  Re: [Freeswitch-users] IPv6
>>>
>>> Please say that topology isn't so... really?  Chances are it won't.
>>>
>>> On Tue, Feb 5, 2019 at 1:56 PM Rick Jarvis <rick at magicmail.mooo.com>
>>> wrote:
>>>
>>>> Looking for some help with FreeSWITCH in a semi-v6 situation… I’ll
>>>> explain what I mean:
>>>>
>>>> VoIP handsets on private IPv4 network -> NAT to IPv6 Internet address
>>>> -> IPv4 public FreeSWITCH server
>>>>
>>>> Not working, and few too many variables to get my head around. I guess
>>>> whether there is an ALG on the the private router is still an issue as it’s
>>>> IPv4 at that point. But should the public IPv6 address speak ok to the
>>>> public IPv4 FreeSWITCH server, or would it help if I give it an IPv6
>>>> address too? (Don’t really understand how the 6->4 thing works, it’s
>>>> voodoo! :)
>>>>
>>>> Thanks, R
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Services
>>>> sales at freeswitch.com
>>>> https://freeswitch.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> https://freeswitch.com/oss
>>>> https://freeswitch.org/confluence
>>>> https://cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> https://freeswitch.com
>>>
>>>
>>>
>>> --
>>>
>>> Brian West | Co-founder and Developer
>>>
>>> Need Commercial support? email sales at freeswitch.com
>>>
>>> FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045
>>> <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g>
>>>
>>> Email: brian at freeswitch.com
>>>
>>> Mobile: 918-424-9378
>>>
>>> Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/>
>>>
>>> [image: https://www.facebook.com/signalwireinc?src=email]
>>> <https://www.facebook.com/freeswitch> [image:
>>> https://twitter.com/freeswitch] <https://twitter.com/freeswitch>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Services
>>> sales at freeswitch.com
>>> https://freeswitch.com
>>>
>>> Official FreeSWITCH Sites
>>> https://freeswitch.com/oss
>>> https://freeswitch.org/confluence
>>> https://cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> https://freeswitch.com
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Services
>>> sales at freeswitch.com
>>> https://freeswitch.com
>>>
>>> Official FreeSWITCH Sites
>>> https://freeswitch.com/oss
>>> https://freeswitch.org/confluence
>>> https://cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> https://freeswitch.com
>>
>>
>>
>> --
>>
>> Brian West | Co-founder and Developer
>>
>> Need Commercial support? email sales at freeswitch.com
>>
>> FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045
>> <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g>
>>
>> Email: brian at freeswitch.com
>>
>> Mobile: 918-424-9378
>>
>> Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/>
>>
>> [image: https://www.facebook.com/signalwireinc?src=email]
>> <https://www.facebook.com/freeswitch> [image:
>> https://twitter.com/freeswitch] <https://twitter.com/freeswitch>
>> _________________________________________________________________________
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From rick at magicmail.mooo.com  Wed Feb  6 12:39:43 2019
From: rick at magicmail.mooo.com (Rick Jarvis)
Date: Wed, 6 Feb 2019 12:39:43 +0000
Subject: [Freeswitch-users] IPv6
In-Reply-To: <CAOrA2UZp=6H-D6VF0badO+Vdj4zAhSH+W8rkkccmPLNSD8YEhQ@mail.gmail.com>
References: <etPan.5c59ea5c.59c3ee71.2c3@magicmail.mooo.com>
 <CAEJMVkDgoajr3bkB_jWz58C0wrLhkVQ4RBNzKNvZKdQNSrY9JA@mail.gmail.com>
 <etPan.5c59f7e0.77702f5c.2c3@magicmail.mooo.com>
 <CAEJMVkD1wbJLXogQj5L9u1XWRZYBLBGq6wusoX8HZxAeXUXe8Q@mail.gmail.com>
 <CAOrA2UY+meDdHNhG8TK=oL-vZYcNDodWOMKe9QAPoNApKU-jng@mail.gmail.com>
 <etPan.5c5abbd6.210657a1.2c3@magicmail.mooo.com>
 <CAOrA2UZp=6H-D6VF0badO+Vdj4zAhSH+W8rkkccmPLNSD8YEhQ@mail.gmail.com>
Message-ID: <etPan.5c5ad58f.4108c483.2c3@magicmail.mooo.com>

Thanks for all your thoughts so far. Diving in at the deep end here, but managed to get their phones to pick up v6 addresses locally (the router was already configured for it), and a v6 address on the FreeSWITCH server. 'eval ${local_ip_v6}' is showing the correct (public) v6 address, so is there anything else I need to do (apart from add a AAAA record for the SIP domain)? Should I be able to see the v6 address in ’sofia status’?


From: Jurijs Ivolga <jurijs.ivolga at gmail.com>
Reply: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Date: 6 February 2019 at 11:45:14
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject:  Re: [Freeswitch-users] IPv6  

Hi,

Not sure that I will be able to help you, but have you checked freeswitch console for logs? What errors you get? A shot in the dark, but TLS might help you in case if some routers altering your SIP messages on the way.

Jurijs


On Wed, Feb 6, 2019 at 12:50 PM Rick Jarvis <rick at magicmail.mooo.com> wrote:
At this stage we’re having problems with the registration (not helped by the fact these Polycoms don’t seem to give much indication of what they’re doing, without looking at the logs or actually being there).

Sadly I don’t know anything about IPv6 (apart from an elevator pitch), so I’m only guessing at what NAT64 and 464xlate is!


From: Jurijs Ivolga <jurijs.ivolga at gmail.com>
Reply: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Date: 6 February 2019 at 10:21:20
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject:  Re: [Freeswitch-users] IPv6

Hi,

Do you see a call arriving to Freeswitch?

I was able to make Freeswitch work with NAT64 and in this case call is arriving to Freeswitch, only there was a problem that in SDP is IPv6 address. But I'm not sure that this is your case.

Jurijs


On Wed, Feb 6, 2019 at 2:18 AM Brian West <brian at freeswitch.com> wrote:
Not currently designed to cross the streams in that manner, 464xlate is probably what you need, I've never had the time to really set that up, I have seen it work on T-Mobile on Bria, where the handset is pure v6 and it arrives the server all v4'ed up.

On Tue, Feb 5, 2019 at 2:54 PM Rick Jarvis <rick at magicmail.mooo.com> wrote:
That’s how it is at the moment, but what do I need to do to fix it? Will putting a v6 IP on FreeSWITCH (and an associated AAAA record) help? Or is the problem that the handsets are on a private v4 network? Or both?!


From: Brian West <brian at freeswitch.com>
Reply: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Date: 5 February 2019 at 20:09:42
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject:  Re: [Freeswitch-users] IPv6

Please say that topology isn't so... really?  Chances are it won't.

On Tue, Feb 5, 2019 at 1:56 PM Rick Jarvis <rick at magicmail.mooo.com> wrote:
Looking for some help with FreeSWITCH in a semi-v6 situation… I’ll explain what I mean:

VoIP handsets on private IPv4 network -> NAT to IPv6 Internet address -> IPv4 public FreeSWITCH server

Not working, and few too many variables to get my head around. I guess whether there is an ALG on the the private router is still an issue as it’s IPv4 at that point. But should the public IPv6 address speak ok to the public IPv4 FreeSWITCH server, or would it help if I give it an IPv6 address too? (Don’t really understand how the 6->4 thing works, it’s voodoo! :)

Thanks, R
_________________________________________________________________________
Professional FreeSWITCH Services
sales at freeswitch.com
https://freeswitch.com

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

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


--


Brian West | Co-founder and
Developer

Need Commercial support? email sales at freeswitch.com

FreeSWITCH Solutions | 17345 Civic Drive
#2531 Brookfield, WI 53045

Email:  
brian at freeswitch.com

Mobile: 918-424-9378

Website:  
https://www.FreeSWITCH.com

  

_________________________________________________________________________
Professional FreeSWITCH Services
sales at freeswitch.com
https://freeswitch.com

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

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
https://freeswitch.com
_________________________________________________________________________
Professional FreeSWITCH Services
sales at freeswitch.com
https://freeswitch.com

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

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


--


Brian West | Co-founder and
Developer

Need Commercial support? email sales at freeswitch.com

FreeSWITCH Solutions | 17345 Civic Drive
#2531 Brookfield, WI 53045

Email:  
brian at freeswitch.com

Mobile: 918-424-9378

Website:  
https://www.FreeSWITCH.com

  

_________________________________________________________________________
Professional FreeSWITCH Services
sales at freeswitch.com
https://freeswitch.com

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

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
https://freeswitch.com
_________________________________________________________________________
Professional FreeSWITCH Services
sales at freeswitch.com
https://freeswitch.com

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

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
https://freeswitch.com
_________________________________________________________________________
Professional FreeSWITCH Services
sales at freeswitch.com
https://freeswitch.com

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

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
https://freeswitch.com
_________________________________________________________________________
Professional FreeSWITCH Services
sales at freeswitch.com
https://freeswitch.com

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

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
https://freeswitch.com
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From tahir at ictinnovations.com  Sat Feb  9 10:20:19 2019
From: tahir at ictinnovations.com (Tahir Almas)
Date: Sat, 9 Feb 2019 05:20:19 -0500
Subject: [Freeswitch-users] ICTFax version 4.0 ,
	Open source Fax over Ip software
Message-ID: <CAOL0t+KOf7bir+G5FOvuqggjirvJ=FJiSikLtF08hLsaRg_QNA@mail.gmail.com>

We are pleased to announce  that  ICTFax 4.0 has been released  developed
over freeswitch , ictcore and angular framework.  New features include  ATA
/ Extension support  and REST API

http://ictfax.org/ictfax-4.0-released-with-ata-api-support

Regards
*Tahir Almas*

Managing Partner
ICT Innovations
http://www.ictinnovations.com
Leveraging open source in ICT
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From ch.chhatra at gmail.com  Sat Feb  9 14:35:01 2019
From: ch.chhatra at gmail.com (Chhorm Chhatra)
Date: Sat, 9 Feb 2019 21:35:01 +0700
Subject: [Freeswitch-users] No Media in SIP Calling from 3G Mobile
 Internet to Freeswitch
In-Reply-To: <CAO5BruACk1uf9bHdf3xreD+yWKDuoATyQ53dXU00ah3260icBw@mail.gmail.com>
References: <CAJssr-iW9F5z4R+apRmu7SvLtR7yCL56wkWgwovzGSygfAtzYQ@mail.gmail.com>
 <CAGJLwFirQfQgx0Gk-O8GChoMhSbNJ_+67e-jSErbvpyNOXagEg@mail.gmail.com>
 <CAO5BruACk1uf9bHdf3xreD+yWKDuoATyQ53dXU00ah3260icBw@mail.gmail.com>
Message-ID: <CAJssr-jEG1=_ND3P=DB1wOJpgo71qKN_fFqtbsMWh9uv2XE3=g@mail.gmail.com>

Dear Alex and Faisal,
Thank you so much for your suggested solutions.
In my case, there were some issues with firewall rules. I had to allow the
UDP and TCP ports for SIP and RTP signals.
For more information to those who might face the same problem, please see
the attached image.

Best regards,
Chhorm Chhatra


On Sun, 3 Feb 2019 at 04:27, Faisal Hanif <imfanee at gmail.com> wrote:

> Adding a stun can help too, if it is Linphone on Android then please add
> stun.linphone.org in stun settings further disable AVFP & ICE
>
> On Sat, Feb 2, 2019, 9:32 AM Alexey Sibyakin <alex at freeswitch.com wrote:
>
>> Hi,
>>
>> Yes, that sounds like a network (NAT/Firewall) issue. Your mobile
>> provider may use a chain of several NATs and not every one of them is
>> working properly. Try VPN or another SIP client.
>>
>> Regards,
>>
>> Alex
>>
>> Alex Sibyakin | Support Engineer SignalWire | 228 Hamilton Ave 3rd Floor,
>> Palo Alto, CA 94303 Email: alex at freeswitch.com Website:
>> https://www.signalwire.com
>>
>>
>>
>> On Sat, Feb 2, 2019 at 2:44 AM Chhorm Chhatra <ch.chhatra at gmail.com>
>> wrote:
>>
>>> Hi,
>>> Currently, I face an issue that I could not hear the audio sound at all
>>> when I try to call to FreeSWITCH (9386) via its public domain name from the
>>> SIP client of my mobile phone with mobile internet.
>>> In this case, SIP registration with TLS and media encryption with SRTP
>>> is working perfectly because I could see the secure symbol in Linphone
>>> *but* I could not hear any audio at all. After 30 seconds of the silent
>>> call, it hanged up automatically.
>>>
>>> *SIP Client on the 3G Mobile Internet > Freeswitch (tls:voip.test.com
>>> <http://voip.test.com>, SRTP) *
>>> *SIP Registration Sucess, Call No Media*
>>>
>>> My question is that is this the problem of NAT? If so, it would be great
>>> if you could help me solve this problem.
>>> Any help would be really appreciated.
>>>
>>> Best regards,
>>> Chhorm Chhatra
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Services
>>> sales at freeswitch.com
>>> https://freeswitch.com
>>>
>>> Official FreeSWITCH Sites
>>> https://freeswitch.com/oss
>>> https://freeswitch.org/confluence
>>> https://cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> https://freeswitch.com
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From aqsyounas at gmail.com  Tue Feb 12 11:42:26 2019
From: aqsyounas at gmail.com (Aqs Younas)
Date: Tue, 12 Feb 2019 16:42:26 +0500
Subject: [Freeswitch-users] streamFile speed deteriorates the voice quality
	badly
Message-ID: <CAGVOjVULmVq95qr2AbK0oQB1rqvZDNF8terOaaQB-g6OA2HG-A@mail.gmail.com>

Greetings list.
I am using streamFile in lua script to control speed/volume of the audio
file.  But i see when i fast ward the audio file(set the speed to +1) it
extremely deteriorates the voice quality.

Does anybody know some parameters/suggestions that might prevent the voice
quality from becoming bad?

Best Regards,

Aqs
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From josedavid at zennio.com  Tue Feb 12 13:44:00 2019
From: josedavid at zennio.com (Jose David Jurado Alonso)
Date: Tue, 12 Feb 2019 14:44:00 +0100
Subject: [Freeswitch-users] Clean expired SIP registrations of FS internal
	database
Message-ID: <CANeCd7rZAPPfCe_f-0ZNDC5Prjey4ispMTniJK4ECKFKs=dbqQ@mail.gmail.com>

Hi,

I have seen that in the FS internal database the account SIP registrations
are stored during a lot of time and it seems that they are not deleted
never.

Is there any mode or parameter to automatically delete these database
records after X time, by  expired time or number of records for example?

To be better explained, I refer to the SIP profile records that appear when
the command is executed:

fs_cli> sofia status profile external reg

Thank a lot.

Regards,

José David Jurado Alonso.
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From gmaruzz at gmail.com  Tue Feb 12 14:31:31 2019
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Tue, 12 Feb 2019 15:31:31 +0100
Subject: [Freeswitch-users] Clean expired SIP registrations of FS
	internal database
In-Reply-To: <CANeCd7rZAPPfCe_f-0ZNDC5Prjey4ispMTniJK4ECKFKs=dbqQ@mail.gmail.com>
References: <CANeCd7rZAPPfCe_f-0ZNDC5Prjey4ispMTniJK4ECKFKs=dbqQ@mail.gmail.com>
Message-ID: <CALXCt0rjxJJZ-ZnW0NoNmaHxRSK_RGQaWNG2+YN1rUtotpZLTg@mail.gmail.com>

On Tue, Feb 12, 2019, 15:13 Jose David Jurado Alonso <josedavid at zennio.com
wrote:

> Hi,
>
> I have seen that in the FS internal database the account SIP registrations
> are stored during a lot of time and it seems that they are not deleted
> never.
>

You are registering on external profile. You are not supposed to. So, maybe
there are not parameters for cleaning up.

Look into internal profile configuration, and copy to external the
relevant  (to registration AND presence AND auth) parameters.

Hth,
-giovanni



> Is there any mode or parameter to automatically delete these database
> records after X time, by  expired time or number of records for example?
>
> To be better explained, I refer to the SIP profile records that appear
> when the command is executed:
>
> fs_cli> sofia status profile external reg
>
> Thank a lot.
>
> Regards,
>
> José David Jurado Alonso.
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From paul.muaddib83 at gmail.com  Wed Feb 13 06:53:15 2019
From: paul.muaddib83 at gmail.com (Paul Muaddib)
Date: Wed, 13 Feb 2019 07:53:15 +0100
Subject: [Freeswitch-users] INCOMPATIBLE_DESTINATION
In-Reply-To: <CAEJMVkC6ZfoAhEj1ur-L2Bnq5mUPhEdfnLcXC_eBdU2g-RwYfQ@mail.gmail.com>
References: <CAEjNe_GwH8S-kojcfWcozSW=+DLoBRMfpcu5ALdxJJTfyz1W=g@mail.gmail.com>
 <e340efbb-7102-0042-24a2-ac36637aee05@level5.de>
 <CAEjNe_HXmHuE10Jd3xaBCcrkfyzc_R=dO720_nOHODhS6ftxQw@mail.gmail.com>
 <2E88B039-C771-4054-A50C-B2996310AEFA@vallimamod.org>
 <CAEjNe_EAzwYMOO3niqa38nm1u1V8GZ=fThHHk4csr+BM1j9jqg@mail.gmail.com>
 <CAEJMVkC6ZfoAhEj1ur-L2Bnq5mUPhEdfnLcXC_eBdU2g-RwYfQ@mail.gmail.com>
Message-ID: <CAEjNe_H=gLKHWoWf_pFrzqO2MZvxx7j_OeBpyUXvLQ38RqkAQQ@mail.gmail.com>

Hi

####################################################
This is the inbound part to freeswitch when they are calling me:

2019/02/12 12:44:00.258755 217.0.15.67:5060 -> 10.0.200.2:49881
INVITE sip:auto_to_user at 87.157.X.X:5080;transport=tcp;gw=
sip-trunk.telekom.de SIP/2.0
Via: SIP/2.0/TCP 217.0.15.67:5060;branch=XYXYXYXYXYXYXYX
Record-Route: <sip:reg.sip-trunk.telekom.de;transport=tcp;lr>
Max-Forwards: 57
To: +49XXXXXX18 <sip:+49XXXXXX18 at TELEKOM.DE;user=phone>
From: <sip:+49XXXXXX11 at sip-trunk.telekom.de
;transport=udp;user=phone>;tag=799d34ab
Call-ID: a6a01cb4d9f46c96 at 62.156.111.70
Contact:
<sip:SxmWXkZ2+Ob+Zli0Jp3jABUfr2GdT4MBP0kzwJgGCuGtkw38S668+tgCxOE6QHU1 at th1>
Supported: 100rel,histinfo,timer
Session-Expires: 1900;refresher=uac
CSeq: 12119032 INVITE
Allow: ACK, BYE, CANCEL, INFO, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK,
REFER, REGISTER, UPDATE
Min-SE: 900
P-Asserted-Identity: <sip:+49XXXXXX11 at sip-trunk.telekom.de;user=phone>
P-Called-Party-ID: <sip:+49XXXXXX18 at sip-trunk.telekom.de;user=phone>
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 499

v=0
o=- 40086765 40086765 IN IP4 217.0.15.67
s=on transit
c=IN IP4 217.0.132.153
t=0 0
m=audio 18570 RTP/AVP 8 118 126 101
a=rtpmap:8 PCMA/8000
a=rtpmap:118 PCMA/8000
a=gpmd:118 vbd=yes
a=rtpmap:126 CLEARMODE/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
m=image 14956 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxMaxBuffer:3000
a=T38FaxMaxDatagram:500
a=T38FaxRateManagement:transferredTCF
a=T38FaxUdpEC:t38UDPRedundancy
a=sendrecv

2019/02/12 12:44:00.258755 10.0.200.2:49881 -> 217.0.15.67:5060
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 217.0.15.67:5060;branch=XYXYXYXYXYXYXYX;rport=5060
Record-Route: <sip:reg.sip-trunk.telekom.de;transport=tcp;lr>
From: <sip:+49XXXXXX11 at sip-trunk.telekom.de
;transport=udp;user=phone>;tag=799d34ab
To: +49XXXXXX18 <sip:+49XXXXXX18 at TELEKOM.DE;user=phone>
Call-ID: a6a01cb4d9f46c96 at 62.156.111.70
CSeq: 12119032 INVITE
Content-Length: 0

2019/02/12 12:44:00.406766 10.0.200.2:49881 -> 217.0.15.67:5060
SIP/2.0 180 Ringing
Via: SIP/2.0/TCP 217.0.15.67:5060;branch=XYXYXYXYXYXYXYX;rport=5060
Record-Route: <sip:reg.sip-trunk.telekom.de;transport=tcp;lr>
From: <sip:+49XXXXXX11 at sip-trunk.telekom.de
;transport=udp;user=phone>;tag=799d34ab
To: +49XXXXXX18 <sip:+49XXXXXX18 at TELEKOM.DE;user=phone>;tag=7K7y11N0937Fj
Call-ID: a6a01cb4d9f46c96 at 62.156.111.70
CSeq: 12119032 INVITE
Contact: <sip:+49XXXXXX18 at 87.157.X.X:5080;transport=tcp>
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
P-Asserted-Identity: "+49XXXXXX18" <sip:+49XXXXXX18 at TELEKOM.DE>


2019/02/12 12:44:02.234905 217.0.15.67:5060 -> 10.0.200.2:49881
BYE sip:+49XXXXXX18 at 87.157.X.X:5080;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 217.0.15.67:5060;branch=ABABABABAB
Record-Route: <sip:reg.sip-trunk.telekom.de;transport=tcp;lr>
Max-Forwards: 61
To: +49XXXXXX18 <sip:+49XXXXXX18 at TELEKOM.DE;user=phone>;tag=7K7y11N0937Fj
From: <sip:+49XXXXXX11 at sip-trunk.telekom.de
;transport=udp;user=phone>;tag=799d34ab
Call-ID: a6a01cb4d9f46c96 at 62.156.111.70
Contact:
<sip:SxmWXkZ2+Ob+Zli0Jp3jABUfr2GdT4MBP0kzwJgGCuGtkw38S668+tgCxOE6QHU1 at th1>
CSeq: 12119033 BYE
Allow: ACK, BYE, CANCEL, INFO, INVITE, MESSAGE, OPTIONS, PRACK, REGISTER,
UPDATE
Warning: 399 MSX_SIPCOM "0 RELEASE FROM CC! ExternalRel:1, PeerType:278,
Call State:9, Reason:25"
Content-Length: 0


2019/02/12 12:44:02.246906 10.0.200.2:49881 -> 217.0.15.67:5060
SIP/2.0 200 OK
Via: SIP/2.0/TCP 217.0.15.67:5060;branch=ABABABABAB;rport=5060
From: <sip:+49XXXXXX11 at sip-trunk.telekom.de
;transport=udp;user=phone>;tag=799d34ab
To: +49XXXXXX18 <sip:+49XXXXXX18 at TELEKOM.DE;user=phone>;tag=7K7y11N0937Fj
Call-ID: a6a01cb4d9f46c96 at 62.156.111.70
CSeq: 12119033 BYE
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
REFER, NOTIFY
Supported: timer, path, replaces
Content-Length: 0


2019/02/12 12:44:02.246906 10.0.200.2:49881 -> 217.0.15.67:5060
SIP/2.0 487 Early Session Terminated
Via: SIP/2.0/TCP 217.0.15.67:5060;branch=XYXYXYXYXYXYXYX;rport=5060
From: <sip:+49XXXXXX11 at sip-trunk.telekom.de
;transport=udp;user=phone>;tag=799d34ab
To: +49XXXXXX18 <sip:+49XXXXXX18 at TELEKOM.DE;user=phone>;tag=7K7y11N0937Fj
Call-ID: a6a01cb4d9f46c96 at 62.156.111.70
CSeq: 12119032 INVITE
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

2019/02/12 12:44:02.266908 217.0.15.67:5060 -> 10.0.200.2:49881
ACK sip:auto_to_user at 87.157.X.X:5080;transport=tcp;gw=sip-trunk.telekom.de
SIP/2.0
Via: SIP/2.0/TCP 217.0.15.67:5060;branch=XYXYXYXYXYXYXYX
Max-Forwards: 57
To: +49XXXXXX18 <sip:+49XXXXXX18 at TELEKOM.DE;user=phone>;tag=7K7y11N0937Fj
From: <sip:+49XXXXXX11 at sip-trunk.telekom.de
;transport=udp;user=phone>;tag=799d34ab
Call-ID: a6a01cb4d9f46c96 at 62.156.111.70
CSeq: 12119032 ACK
Content-Length: 0

########################################
This is the invite to my phone behind freeswitch

2019/02/12 12:44:00.298758 10.0.200.2:5060 -> 10.0.200.18:5060
INVITE sip:18 at 10.0.200.18:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.200.2;rport;branch=z9hG4bKZ4t3F3ZrKrHQc
Max-Forwards: 55
From: "+49XXXXXXXX11" <sip:+49XXXXXXXX11 at 10.0.200.2>;tag=a3mj5XKSHyjjD
To: <sip:18 at 10.0.200.18:5060>
Call-ID: 5466d642-a95e-1237-f598-000024d0e264
CSeq: 432128 INVITE
Contact: <sip:mod_sofia at 10.0.200.2:5060>
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
Privacy: none
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 239
P-Called-Party-ID: <sip:+49XXXXXXXX18 at sip-trunk.telekom.de;user=phone>
X-FS-Support: update_display,send_info
P-Asserted-Identity: "+49XXXXXXXX11" <sip:+49XXXXXXXX11 at 10.0.200.2>

v=0
o=FreeSWITCH 1549964758 1549964759 IN IP4 10.0.200.2
s=FreeSWITCH
c=IN IP4 10.0.200.2
t=0 0
m=audio 7082 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

Am Do., 23. Aug. 2018 um 17:50 Uhr schrieb Brian West <brian at freeswitch.com
>:

> it actually is probably fax related since the initial invite has T.38
> stuff in it, I bet its getting tripped up because T.38 is a NO GO REJECT
> and it probably says something to that in the debug logs.
>
> /b
>
>
> On Wed, Aug 22, 2018 at 3:15 PM, Paul Muaddib <paul.muaddib83 at gmail.com>
> wrote:
>
>> Hi,
>>
>> It is definitely not a Fax. The guys, who are calling us from this
>> number, are actually complaining that we are not reachable. But when we
>> call them back the voice call perfectly works. Other Inbound calls are no
>> problem.
>>
>> Best regards,
>> Paul
>>
>> Am Mi., 22. Aug. 2018 um 21:53 Uhr schrieb Vallimamod Abdullah <
>> vma at vallimamod.org>:
>>
>>> Hi,
>>>
>>> When there is udptl media type in sdp, freeswitch ignores the audio
>>> type. That's why you are getting the codec error (if you don't have t38
>>> enabled.)
>>> The caller looks like a fax machine, which explains why you get a direct
>>> t38 request and not a reinvite.
>>>
>>>
>>> Best Regards,
>>> --
>>> Vallimamod Abdullah
>>> SIP Solutions
>>> vma at sip.solutions
>>> linkedin.com/in/vallimamod
>>> .
>>>
>>> On 22 Aug 2018, at 15:55, Paul Muaddib <paul.muaddib83 at gmail.com> wrote:
>>>
>>> vars.xml
>>>
>>>   <X-PRE-PROCESS cmd="set" data="global_codec_prefs=PCMA,PCMU"/>
>>>   <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMA,PCMU"/>
>>>
>>> My phones, all Grandstream GXP2140, accept PCMA and PCMU
>>>
>>> I don't have trouble with other people calling me
>>>
>>> Am Mi., 22. Aug. 2018 um 15:21 Uhr schrieb Thorsten Göllner <
>>> tg-maillistings at level5.de>:
>>>
>>>> Only PCMA at 8000 is offered. Check device if codec is activated.
>>>>
>>>> Am 21.08.2018 um 15:06 schrieb Paul Muaddib:
>>>>
>>>> Hi,
>>>>
>>>> I have a rarely error where I get this failure message in the log file.
>>>> I am bit wondering because this is suposed to be a normal voice call and
>>>> not a fax.
>>>>
>>>> Best regards,
>>>> Paul.
>>>>
>>>> fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 2018-08-20 14:04:54.056776 [INFO]
>>>> switch_ivr_originate.c:1215 Sending early media
>>>> fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 2018-08-20 14:04:54.056776 [ERR]
>>>> mod_sofia.c:2343 CODEC NEGOTIATION ERROR.  SDP:
>>>> fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 v=0^M
>>>> fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 o=- 20012711 20012711 IN IP4
>>>> 217.0.15.67^M
>>>> fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 s=on transit^M
>>>> fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 c=IN IP4 X.X.X.X^M
>>>> fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 t=0 0^M
>>>> fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 m=audio 13274 RTP/AVP 8 118 126
>>>> 101^M
>>>> fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 a=rtpmap:8 PCMA/8000^M
>>>> fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 a=rtpmap:118 PCMA/8000^M
>>>> fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 a=rtpmap:126 CLEARMODE/8000^M
>>>> fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 a=rtpmap:101 telephone-event/8000^M
>>>> fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 a=fmtp:101 0-15^M
>>>> fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 a=gpmd:118 vbd=yes^M
>>>> fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 a=ptime:20^M
>>>> fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 m=image 11232 udptl t38^M
>>>> fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 a=T38FaxVersion:0^M
>>>> fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 a=T38MaxBitRate:14400^M
>>>> fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 a=T38FaxMaxBuffer:3000^M
>>>> fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 a=T38FaxMaxDatagram:500^M
>>>> fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28
>>>> a=T38FaxRateManagement:transferredTCF^M
>>>> fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 a=T38FaxUdpEC:t38UDPRedundancy^M
>>>> fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28
>>>> fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 2018-08-20 14:04:54.056776
>>>> [NOTICE] switch_channel.c:3515 Hangup sofia/external/+
>>>> 49XXXXXXXXX at sip-trunk.telekom.de [CS_EXECUTE]
>>>> [INCOMPATIBLE_DESTINATION]
>>>>
>>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Services
>>> sales at freeswitch.com
>>> https://freeswitch.com
>>>
>>> Official FreeSWITCH Sites
>>> https://freeswitch.com/oss
>>> https://freeswitch.org/confluence
>>> https://cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> https://freeswitch.com
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>>
>
>
>
> --
>
> Brian West | Co-founder and Developer
>
> Need Commercial support? email sales at freeswitch.com
>
> FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045
> <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g>
>
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>
> Mobile: 918-424-9378
>
> Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/>
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> _________________________________________________________________________
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> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
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> https://freeswitch.org/confluence
> https://cluecon.com
>
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> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From josedavid at zennio.com  Wed Feb 13 09:20:45 2019
From: josedavid at zennio.com (Jose David Jurado Alonso)
Date: Wed, 13 Feb 2019 10:20:45 +0100
Subject: [Freeswitch-users] Clean expired SIP registrations of FS
	internal database
In-Reply-To: <CALXCt0rjxJJZ-ZnW0NoNmaHxRSK_RGQaWNG2+YN1rUtotpZLTg@mail.gmail.com>
References: <CANeCd7rZAPPfCe_f-0ZNDC5Prjey4ispMTniJK4ECKFKs=dbqQ@mail.gmail.com>
 <CALXCt0rjxJJZ-ZnW0NoNmaHxRSK_RGQaWNG2+YN1rUtotpZLTg@mail.gmail.com>
Message-ID: <CANeCd7rjLG9Y38vx7Fv5Z7Ta7Qx2qDVNN=RX+Zo+-56pTKn2DQ@mail.gmail.com>

Yes, I am using the external profile but it is a copy of the internal one.
I imagine that having the desired configuration it is irrelevant to use one
or the other.

Can you tell me what parameters and exactly what values should be
configured for cleaning?

I have the next configuration file:

<profile name="external">
  <!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files -->
  <!-- This profile is only for outbound registrations to providers -->
  <gateways>
    <X-PRE-PROCESS cmd="include" data="external/*.xml"/>
  </gateways>

  <aliases>
    <!--
        <alias name="outbound"/>
        <alias name="nat"/>
    -->
  </aliases>

  <domains>
    <domain name="all" alias="false" parse="true"/>
  </domains>

  <settings>
    <param name="debug" value="0"/>
    <!-- If you want FreeSWITCH to shutdown if this profile fails to load,
uncomment the next line. -->
    <!-- <param name="shutdown-on-fail" value="true"/> -->
    <param name="sip-trace" value="no"/>
    <param name="sip-capture" value="no"/>

<!--
    <param name="watchdog-enabled" value="no"/>
    <param name="watchdog-step-timeout" value="30000"/>
    <param name="watchdog-event-timeout" value="30000"/>

    <param name="forward-unsolicited-mwi-notify" value="false"/>
-->

    <param name="log-auth-failures" value="true"/>
    <param name="rfc2833-pt" value="101"/>
    <!-- RFC 5626 : Send reg-id and sip.instance -->
    <!--<param name="enable-rfc-5626" value="true"/> -->
    <param name="sip-port" value="$${external_sip_port}"/>
    <param name="dialplan" value="XML"/>
    <param name="context" value="public"/>
    <param name="dtmf-duration" value="2000"/>
    <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/>
    <param name="outbound-codec-prefs" value="$${outbound_codec_prefs}"/>
    <param name="hold-music" value="$${hold_music}"/>
    <param name="rtp-timer-name" value="soft"/>
    <!--<param name="enable-100rel" value="true"/>-->
    <!--<param name="disable-srv503" value="true"/>-->
    <param name="local-network-acl" value="rfc1918.auto"/>
    <param name="apply-nat-acl" value="rfc1918"/>
    <!-- <param name="apply-register-acl" value="domains"/>-->

    <!-- This could be set to "passive" -->
    <param name="manage-presence" value="false"/>

    <!-- used to share presence info across sofia profiles
         manage-presence needs to be set to passive on this profile
         if you want it to behave as if it were the internal profile
         for presence.
    -->
    <!-- Name of the db to use for this profile -->
    <!--<param name="dbname" value="share_presence"/>-->
    <param name="presence-hosts" value="$${domain}"/>
    <param name="force-register-domain" value="$${domain}"/>
    <!--all inbound reg will stored in the db using this domain -->
    <!--<param name="force-register-db-domain" value="$${domain}"/>-->
    <!-- ************************************************* -->

    <param name="aggressive-nat-detection" value="true"/>
    <param name="inbound-codec-negotiation" value="generous"/>
    <param name="nonce-ttl" value="60"/>
    <param name="auth-calls" value="true"/>
    <param name="inbound-late-negotiation" value="true"/>
    <param name="inbound-zrtp-passthru" value="true"/> <!-- (also enables
late negotiation) -->
    <!--
        DO NOT USE HOSTNAMES, ONLY IP ADDRESSES IN THESE SETTINGS!
    -->
    <param name="rtp-ip" value="$${local_ip_v4}"/>
    <param name="sip-ip" value="$${local_ip_v4}"/>
    <param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
    <param name="ext-sip-ip" value="$${external_sip_ip}"/>
    <param name="rtp-timeout-sec" value="300"/>
    <param name="rtp-hold-timeout-sec" value="1800"/>
    <!--<param name="enable-3pcc" value="true"/>-->

    <!-- TLS: disabled by default, set to "true" to enable -->
    <param name="tls" value="$${external_ssl_enable}"/>
    <!-- Set to true to not bind on the normal sip-port but only on the TLS
port -->
    <param name="tls-only" value="false"/>
    <!-- additional bind parameters for TLS -->
    <param name="tls-bind-params" value="transport=tls"/>
    <!-- Port to listen on for TLS requests. (5081 will be used if
unspecified) -->
    <param name="tls-sip-port" value="$${external_tls_port}"/>
    <!-- Location of the agent.pem and cafile.pem ssl certificates (needed
for TLS server) -->
    <!--<param name="tls-cert-dir" value=""/>-->
    <!-- Optionally set the passphrase password used by openSSL to
encrypt/decrypt TLS private key files -->
    <param name="tls-passphrase" value=""/>
    <!-- Verify the date on TLS certificates -->
    <param name="tls-verify-date" value="true"/>
    <!-- TLS verify policy, when registering/inviting gateways with other
servers (outbound) or handling inbound registration/invite requests how
should we verify their certificate -->
    <!-- set to 'in' to only verify incoming connections, 'out' to only
verify outgoing connections, 'all' to verify all connections, also
'subjects_in', 'subjects_out' and 'subjects_all' for subject validation.
Multiple policies can be split with a '|' pipe -->
    <param name="tls-verify-policy" value="none"/>
    <!-- Certificate max verify depth to use for validating peer TLS
certificates when the verify policy is not none -->
    <param name="tls-verify-depth" value="2"/>
    <!-- If the tls-verify-policy is set to subjects_all or subjects_in
this sets which subjects are allowed, multiple subjects can be split with a
'|' pipe -->
    <param name="tls-verify-in-subjects" value=""/>
    <!-- TLS version ("sslv23" (default), "tlsv1"). NOTE: Phones may not
work with TLSv1 -->
    <param name="tls-version" value="$${sip_tls_version}"/>

    <param name="aggressive-nat-detection" value="true"/>
    <param name="multiple-registrations" value="true"/>
    <param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
    <param name="ext-sip-ip" value="$${external_sip_ip}"/>
    <param name="NDLB-received-in-nat-reg-contact" value="true"/>
    <param name="NDLB-force-rport" value="safe"/>
    <param name="NDLB-broken-auth-hash" value="true"/>
    <param name="enable-timer" value="false"/>
    <param name="auth-calls" value="true"/>


  </settings>
</profile>


Thanks.
Jose David Jurado Alonso






El mar., 12 feb. 2019 a las 15:46, Giovanni Maruzzelli (<gmaruzz at gmail.com>)
escribió:

> On Tue, Feb 12, 2019, 15:13 Jose David Jurado Alonso <josedavid at zennio.com
> wrote:
>
>> Hi,
>>
>> I have seen that in the FS internal database the account SIP
>> registrations are stored during a lot of time and it seems that they are
>> not deleted never.
>>
>
> You are registering on external profile. You are not supposed to. So,
> maybe there are not parameters for cleaning up.
>
> Look into internal profile configuration, and copy to external the
> relevant  (to registration AND presence AND auth) parameters.
>
> Hth,
> -giovanni
>
>
>
>> Is there any mode or parameter to automatically delete these database
>> records after X time, by  expired time or number of records for example?
>>
>> To be better explained, I refer to the SIP profile records that appear
>> when the command is executed:
>>
>> fs_cli> sofia status profile external reg
>>
>> Thank a lot.
>>
>> Regards,
>>
>> José David Jurado Alonso.
>> _________________________________________________________________________
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From josedavid at zennio.com  Wed Feb 13 09:22:50 2019
From: josedavid at zennio.com (Jose David Jurado Alonso)
Date: Wed, 13 Feb 2019 10:22:50 +0100
Subject: [Freeswitch-users] Clean expired SIP registrations of FS
	internal database
In-Reply-To: <CALXCt0rjxJJZ-ZnW0NoNmaHxRSK_RGQaWNG2+YN1rUtotpZLTg@mail.gmail.com>
References: <CANeCd7rZAPPfCe_f-0ZNDC5Prjey4ispMTniJK4ECKFKs=dbqQ@mail.gmail.com>
 <CALXCt0rjxJJZ-ZnW0NoNmaHxRSK_RGQaWNG2+YN1rUtotpZLTg@mail.gmail.com>
Message-ID: <CANeCd7qbmJRJUykBNeisQHXmN6-4rqxtmNba4Lrkk=B3iKDr5A@mail.gmail.com>

Yes, I am using the external profile but it is a copy of the internal one
and modified by me. I imagine that having the desired configuration it is
irrelevant to use one or the other.

Can you tell me what parameters and exactly what values should be
configured for cleaning?

I have the next configuration file:

<profile name="external">
  <!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files -->
  <!-- This profile is only for outbound registrations to providers -->
  <gateways>
    <X-PRE-PROCESS cmd="include" data="external/*.xml"/>
  </gateways>

  <aliases>
    <!--
        <alias name="outbound"/>
        <alias name="nat"/>
    -->
  </aliases>

  <domains>
    <domain name="all" alias="false" parse="true"/>
  </domains>

  <settings>
    <param name="debug" value="0"/>
    <!-- If you want FreeSWITCH to shutdown if this profile fails to load,
uncomment the next line. -->
    <!-- <param name="shutdown-on-fail" value="true"/> -->
    <param name="sip-trace" value="no"/>
    <param name="sip-capture" value="no"/>

<!--
    <param name="watchdog-enabled" value="no"/>
    <param name="watchdog-step-timeout" value="30000"/>
    <param name="watchdog-event-timeout" value="30000"/>

    <param name="forward-unsolicited-mwi-notify" value="false"/>
-->

    <param name="log-auth-failures" value="true"/>
    <param name="rfc2833-pt" value="101"/>
    <!-- RFC 5626 : Send reg-id and sip.instance -->
    <!--<param name="enable-rfc-5626" value="true"/> -->
    <param name="sip-port" value="$${external_sip_port}"/>
    <param name="dialplan" value="XML"/>
    <param name="context" value="public"/>
    <param name="dtmf-duration" value="2000"/>
    <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/>
    <param name="outbound-codec-prefs" value="$${outbound_codec_prefs}"/>
    <param name="hold-music" value="$${hold_music}"/>
    <param name="rtp-timer-name" value="soft"/>
    <!--<param name="enable-100rel" value="true"/>-->
    <!--<param name="disable-srv503" value="true"/>-->
    <param name="local-network-acl" value="rfc1918.auto"/>
    <param name="apply-nat-acl" value="rfc1918"/>
    <!-- <param name="apply-register-acl" value="domains"/>-->

    <!-- This could be set to "passive" -->
    <param name="manage-presence" value="false"/>

    <!-- used to share presence info across sofia profiles
         manage-presence needs to be set to passive on this profile
         if you want it to behave as if it were the internal profile
         for presence.
    -->
    <!-- Name of the db to use for this profile -->
    <!--<param name="dbname" value="share_presence"/>-->
    <param name="presence-hosts" value="$${domain}"/>
    <param name="force-register-domain" value="$${domain}"/>
    <!--all inbound reg will stored in the db using this domain -->
    <!--<param name="force-register-db-domain" value="$${domain}"/>-->
    <!-- ************************************************* -->

    <param name="aggressive-nat-detection" value="true"/>
    <param name="inbound-codec-negotiation" value="generous"/>
    <param name="nonce-ttl" value="60"/>
    <param name="auth-calls" value="true"/>
    <param name="inbound-late-negotiation" value="true"/>
    <param name="inbound-zrtp-passthru" value="true"/> <!-- (also enables
late negotiation) -->
    <!--
        DO NOT USE HOSTNAMES, ONLY IP ADDRESSES IN THESE SETTINGS!
    -->
    <param name="rtp-ip" value="$${local_ip_v4}"/>
    <param name="sip-ip" value="$${local_ip_v4}"/>
    <param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
    <param name="ext-sip-ip" value="$${external_sip_ip}"/>
    <param name="rtp-timeout-sec" value="300"/>
    <param name="rtp-hold-timeout-sec" value="1800"/>
    <!--<param name="enable-3pcc" value="true"/>-->

    <!-- TLS: disabled by default, set to "true" to enable -->
    <param name="tls" value="$${external_ssl_enable}"/>
    <!-- Set to true to not bind on the normal sip-port but only on the TLS
port -->
    <param name="tls-only" value="false"/>
    <!-- additional bind parameters for TLS -->
    <param name="tls-bind-params" value="transport=tls"/>
    <!-- Port to listen on for TLS requests. (5081 will be used if
unspecified) -->
    <param name="tls-sip-port" value="$${external_tls_port}"/>
    <!-- Location of the agent.pem and cafile.pem ssl certificates (needed
for TLS server) -->
    <!--<param name="tls-cert-dir" value=""/>-->
    <!-- Optionally set the passphrase password used by openSSL to
encrypt/decrypt TLS private key files -->
    <param name="tls-passphrase" value=""/>
    <!-- Verify the date on TLS certificates -->
    <param name="tls-verify-date" value="true"/>
    <!-- TLS verify policy, when registering/inviting gateways with other
servers (outbound) or handling inbound registration/invite requests how
should we verify their certificate -->
    <!-- set to 'in' to only verify incoming connections, 'out' to only
verify outgoing connections, 'all' to verify all connections, also
'subjects_in', 'subjects_out' and 'subjects_all' for subject validation.
Multiple policies can be split with a '|' pipe -->
    <param name="tls-verify-policy" value="none"/>
    <!-- Certificate max verify depth to use for validating peer TLS
certificates when the verify policy is not none -->
    <param name="tls-verify-depth" value="2"/>
    <!-- If the tls-verify-policy is set to subjects_all or subjects_in
this sets which subjects are allowed, multiple subjects can be split with a
'|' pipe -->
    <param name="tls-verify-in-subjects" value=""/>
    <!-- TLS version ("sslv23" (default), "tlsv1"). NOTE: Phones may not
work with TLSv1 -->
    <param name="tls-version" value="$${sip_tls_version}"/>

    <param name="aggressive-nat-detection" value="true"/>
    <param name="multiple-registrations" value="true"/>
    <param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
    <param name="ext-sip-ip" value="$${external_sip_ip}"/>
    <param name="NDLB-received-in-nat-reg-contact" value="true"/>
    <param name="NDLB-force-rport" value="safe"/>
    <param name="NDLB-broken-auth-hash" value="true"/>
    <param name="enable-timer" value="false"/>
    <param name="auth-calls" value="true"/>


  </settings>
</profile>


Thanks.

El mar., 12 feb. 2019 a las 15:46, Giovanni Maruzzelli (<gmaruzz at gmail.com>)
escribió:

> On Tue, Feb 12, 2019, 15:13 Jose David Jurado Alonso <josedavid at zennio.com
> wrote:
>
>> Hi,
>>
>> I have seen that in the FS internal database the account SIP
>> registrations are stored during a lot of time and it seems that they are
>> not deleted never.
>>
>
> You are registering on external profile. You are not supposed to. So,
> maybe there are not parameters for cleaning up.
>
> Look into internal profile configuration, and copy to external the
> relevant  (to registration AND presence AND auth) parameters.
>
> Hth,
> -giovanni
>
>
>
>> Is there any mode or parameter to automatically delete these database
>> records after X time, by  expired time or number of records for example?
>>
>> To be better explained, I refer to the SIP profile records that appear
>> when the command is executed:
>>
>> fs_cli> sofia status profile external reg
>>
>> Thank a lot.
>>
>> Regards,
>>
>> José David Jurado Alonso.
>> _________________________________________________________________________
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From gmaruzz at gmail.com  Wed Feb 13 13:18:40 2019
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Wed, 13 Feb 2019 14:18:40 +0100
Subject: [Freeswitch-users] Clean expired SIP registrations of FS
	internal database
In-Reply-To: <CANeCd7qbmJRJUykBNeisQHXmN6-4rqxtmNba4Lrkk=B3iKDr5A@mail.gmail.com>
References: <CANeCd7rZAPPfCe_f-0ZNDC5Prjey4ispMTniJK4ECKFKs=dbqQ@mail.gmail.com>
 <CALXCt0rjxJJZ-ZnW0NoNmaHxRSK_RGQaWNG2+YN1rUtotpZLTg@mail.gmail.com>
 <CANeCd7qbmJRJUykBNeisQHXmN6-4rqxtmNba4Lrkk=B3iKDr5A@mail.gmail.com>
Message-ID: <CALXCt0r7vuuKdSdLjubP_ef+sdx2C-ib87crdxDV8YLua2Legg@mail.gmail.com>

On Wed, Feb 13, 2019 at 11:04 AM Jose David Jurado Alonso <
josedavid at zennio.com> wrote:

>
>
>
>
>
> Yes, I am using the external profile but it is a copy of the internal one
> and modified by me. I imagine that having the desired configuration it is
> irrelevant to use one or the other.
>
>
Yes, I would think the same.

Maybe you want to check with a sip trace if the registrations are ok and
coherents with what the system expects...

Maybe there is a mismatch of some kind... (force registration domain name?)

Other than this, you probably need a consultant to look into it.

You can write to consulting at freeswitch.org

-giovanni
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From ch.chhatra at gmail.com  Thu Feb 14 07:45:50 2019
From: ch.chhatra at gmail.com (Chhorm Chhatra)
Date: Thu, 14 Feb 2019 14:45:50 +0700
Subject: [Freeswitch-users] NORMAL_TEMPORARY_FAILURE Freeswitch Outbound via
	Gateway to PSTN
Message-ID: <CAJssr-gzzjqKyWOTLJWS1hvR_85k8N21XpOc1_LX8GTU=sc_nw@mail.gmail.com>

Hi,
Currently, I faced the issue of getting NORMAL_TEMPORARY_FAILURE when
FreeSWITCH is originating outbound calls to the PSTN via Dinstar GSM
gateway device.
I follow the guides from FreeSWITCH confluence and the inbound call is
working fine.
My Dinstar Gateway device version is UC2000-VE Business.
Any help would be appreciated.

Best regards,
Chhorm Chhatra
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From gmaruzz at gmail.com  Thu Feb 14 08:21:14 2019
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Thu, 14 Feb 2019 09:21:14 +0100
Subject: [Freeswitch-users] NORMAL_TEMPORARY_FAILURE Freeswitch Outbound
 via Gateway to PSTN
In-Reply-To: <CAJssr-gzzjqKyWOTLJWS1hvR_85k8N21XpOc1_LX8GTU=sc_nw@mail.gmail.com>
References: <CAJssr-gzzjqKyWOTLJWS1hvR_85k8N21XpOc1_LX8GTU=sc_nw@mail.gmail.com>
Message-ID: <CALXCt0rnyduiSTtnpFZ8CGktNLB16dZ+kWhKuSB_0o6+ovjCvw@mail.gmail.com>

On Thu, Feb 14, 2019 at 9:15 AM Chhorm Chhatra <ch.chhatra at gmail.com> wrote:

> Hi,
> Currently, I faced the issue of getting NORMAL_TEMPORARY_FAILURE when
> FreeSWITCH is originating outbound calls to the PSTN via Dinstar GSM
> gateway device.
>

check what number you are sending the call to

probably freeswitch thinks is an internal extension, or such

be sure to have a dialplan extension that sends the number outbound to the
gateway

-giovanni
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From mario_fs at mgtech.com  Thu Feb 14 22:07:33 2019
From: mario_fs at mgtech.com (mario_fs)
Date: Thu, 14 Feb 2019 14:07:33 -0800
Subject: [Freeswitch-users] Option to log registrations/deregistrations?
Message-ID: <B01E1C86-E101-4965-B686-606BBE720E70@mgtech.com>

Does anyone know if there is something I can do to log device registrations, and more importantly when devices lose registration, without having a ton of other log data. I know I can turn on a higher level of diagnostics, but it may be several days before I find what I need and don’t want to flood the log.

I am working with Counterpath on Bria Push service and need to find when their push server stops registering. Since they don’t send a notification for that and it’s not in the normal FS log, I may not know for days when this happens. It would be nice to be able to scan the log for lost registrations. Thanks!
Mario G
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From yu at yu-boot.ru  Fri Feb 15 06:17:08 2019
From: yu at yu-boot.ru (Yu Boot)
Date: Fri, 15 Feb 2019 09:17:08 +0300
Subject: [Freeswitch-users] How to find mistake in FS config?
Message-ID: <94b9c25a-78b5-ac8b-42f2-7a74b855e017@yu-boot.ru>

I changed some things in internal.xml and now I got this:

# fs_cli -x reloadxml

+OK [[error near line 10374]: missing >]

Searching in dialplan.xml at line 13074 gives nothing.

How to find where's mistake?



From sebastian_ml at gmx.net  Fri Feb 15 07:11:53 2019
From: sebastian_ml at gmx.net (Sebastian Kemper)
Date: Fri, 15 Feb 2019 07:11:53 +0000
Subject: [Freeswitch-users] How to find mistake in FS config?
In-Reply-To: <94b9c25a-78b5-ac8b-42f2-7a74b855e017@yu-boot.ru>
References: <94b9c25a-78b5-ac8b-42f2-7a74b855e017@yu-boot.ru>
Message-ID: <28CBC1C2-F608-4A6F-B7C9-69A221B845FC@gmx.net>

Hi Yu,

Check your log directory. FS generates an XML file with the complete config and saves it in the log dir. The line should be in there.

Regards,
Seb


From avi at avimarcus.net  Fri Feb 15 08:05:12 2019
From: avi at avimarcus.net (Avi Marcus)
Date: Fri, 15 Feb 2019 08:05:12 +0000
Subject: [Freeswitch-users] Option to log registrations/deregistrations?
In-Reply-To: <B01E1C86-E101-4965-B686-606BBE720E70@mgtech.com>
References: <B01E1C86-E101-4965-B686-606BBE720E70@mgtech.com>
Message-ID: <01000168f0302208-303a2b9c-ffa8-4cad-ac4e-d87797cc2c4d-000000@email.amazonses.com>

Here's some old nodejs code relying on `esl` lib to monitor for
de-registrations and send me an email. I haven't used it in ages.

https://gist.github.com/avimar/9645dd3134476d458e68cca077f4ed16


-Avi Marcus
BestFone


On Fri, Feb 15, 2019 at 12:13 AM mario_fs <mario_fs at mgtech.com> wrote:

> Does anyone know if there is something I can do to log device
> registrations, and more importantly when devices lose registration, *without
> having a ton of other log data*. I know I can turn on a higher level of
> diagnostics, but it may be several days before I find what I need and don’t
> want to flood the log.
>
> I am working with Counterpath on Bria Push service and need to find when
> their push server stops registering. Since they don’t send a notification
> for that and it’s not in the normal FS log, I may not know for days when
> this happens. It would be nice to be able to scan the log for lost
> registrations. Thanks!
> Mario G
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From s.safarov at gmail.com  Fri Feb 15 09:04:24 2019
From: s.safarov at gmail.com (Sergey Safarov)
Date: Fri, 15 Feb 2019 12:04:24 +0300
Subject: [Freeswitch-users] How to find mistake in FS config?
In-Reply-To: <28CBC1C2-F608-4A6F-B7C9-69A221B845FC@gmx.net>
References: <94b9c25a-78b5-ac8b-42f2-7a74b855e017@yu-boot.ru>
 <28CBC1C2-F608-4A6F-B7C9-69A221B845FC@gmx.net>
Message-ID: <CAHtxdDdAOB+5JzKfnRkwe7cokakihAYDKPJh8w5GndigzT4U+w@mail.gmail.com>

Also you you can check content of /var/log/freeswitch/freeswitch.xml.fsxml
using XML validator.
Validator must show XML structure error.

Sergey

пт, 15 февр. 2019 г. в 10:14, Sebastian Kemper <sebastian_ml at gmx.net>:

> Hi Yu,
>
> Check your log directory. FS generates an XML file with the complete
> config and saves it in the log dir. The line should be in there.
>
> Regards,
> Seb
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From mario_fs at mgtech.com  Fri Feb 15 17:46:33 2019
From: mario_fs at mgtech.com (mario_fs)
Date: Fri, 15 Feb 2019 09:46:33 -0800
Subject: [Freeswitch-users] Option to log registrations/deregistrations?
In-Reply-To: <01000168f0302208-303a2b9c-ffa8-4cad-ac4e-d87797cc2c4d-000000@email.amazonses.com>
References: <B01E1C86-E101-4965-B686-606BBE720E70@mgtech.com>
 <01000168f0302208-303a2b9c-ffa8-4cad-ac4e-d87797cc2c4d-000000@email.amazonses.com>
Message-ID: <771A1361-D809-41AF-911F-1A4077DCF898@mgtech.com>

Hmmmm…. Logging de-registrations would be a useful enhancement to FreeSwitch for debugging issues like this. I know a SIP trace does it but that’s a LOT of data when we're talking days. Thanks.
Mario G

> On Feb 15, 2019, at 12:05 AM, Avi Marcus <avi at avimarcus.net> wrote:
> 
> Here's some old nodejs code relying on `esl` lib to monitor for de-registrations and send me an email. I haven't used it in ages.
> 
> https://gist.github.com/avimar/9645dd3134476d458e68cca077f4ed16 <https://gist.github.com/avimar/9645dd3134476d458e68cca077f4ed16>
> 
> 
> -Avi Marcus
> BestFone
> 
> 
> On Fri, Feb 15, 2019 at 12:13 AM mario_fs <mario_fs at mgtech.com <mailto:mario_fs at mgtech.com>> wrote:
> Does anyone know if there is something I can do to log device registrations, and more importantly when devices lose registration, without having a ton of other log data. I know I can turn on a higher level of diagnostics, but it may be several days before I find what I need and don’t want to flood the log.
> 
> I am working with Counterpath on Bria Push service and need to find when their push server stops registering. Since they don’t send a notification for that and it’s not in the normal FS log, I may not know for days when this happens. It would be nice to be able to scan the log for lost registrations. Thanks!
> Mario G
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com <mailto:sales at freeswitch.com>
> https://freeswitch.com <https://freeswitch.com/>
> 
> Official FreeSWITCH Sites
> https://freeswitch.com/oss <https://freeswitch.com/oss>
> https://freeswitch.org/confluence <https://freeswitch.org/confluence>
> https://cluecon.com <https://cluecon.com/>
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> https://freeswitch.com <https://freeswitch.com/>_________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
> 
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com

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From panos at kioski.gr  Mon Feb 18 12:23:12 2019
From: panos at kioski.gr (Panagiotis Voutskidis)
Date: Mon, 18 Feb 2019 14:23:12 +0200
Subject: [Freeswitch-users] Called id name is not displayed - Cisco 7912G
	devices
Message-ID: <CAKv+6K8fGVPBR+hk58e1k6kLELx1Spdniak3Xr6vYgy4jLUKLw@mail.gmail.com>

I run a freeswitch pbx with about 10 cisco 7912G (with SIP firmware) and 2
linksys devices.
When I call a local device from a cisco ip phone, we don't see the called
device id name, during ringing and talking. The only thing displayed is the
dialed number. Linksys devices display called user name as expected.

Is it a problem with the specific device and firmware or there is something
I could do to fix it? Is there anyone else using cisco 7912G phones that
can help with the situation?

Regards,

Panos
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From ch.chhatra at gmail.com  Sat Feb  9 15:50:40 2019
From: ch.chhatra at gmail.com (Chhorm Chhatra)
Date: Sat, 9 Feb 2019 22:50:40 +0700
Subject: [Freeswitch-users] Normal Temporary Failure - Freeswitch Outbout to
	Dinstar GSM Gateway
Message-ID: <CAJssr-hMvrRLFiTpCBW_X4+ESvru9UguzfKkY+K0OF4gF6jEZA@mail.gmail.com>

Hello,
Currently, I faced an issue in setting up the outbound call from Freeswitch
to PSTN via Dinstar GSM Gateway.
My question is there any other configuration I would have to follow to make
it works? Please kindly give advises.
I follow this Freeswitch Dinstar How-To Guide
<https://freeswitch.org/confluence/display/FREESWITCH/Dinstar+GSM+gateway+FreeSwitch+HowTo>.
Since the tutorial is not working for the outbound call to PSTN, I followed
the DWG_WITH_FREESWITCH.pdf instead. Unfortunately, both of them not
working for me with outbound calls.
I tried with the following configuration:
*1. With Dinstar GSM Gateway IP*
 <extension name="dwg2008_gateway">
   <condition field="destination_number" expression="^0(\d+)$">
   <action application="answer"/>
   <action application="set" data="ringback=${us-ring}"/>
   <action application="log" data="INFO calling via gateway to 0$1"/>
   <action application="bridge" data="sofia/internal/0$1 at 192.168.7.137"/>
  </condition>
</extension>
*But this is not working, I got NORMAL_TEMPORARY_FAILURE.*
Please see the dinstar log named* setup_1.log*

*2. With GSM*
<extension name="dwg2008_gateway">
   <condition field="destination_number" expression="^0(\d+)$">
   <action application="answer"/>
   <action application="set" data="ringback=${us-ring}"/>
   <action application="log" data="INFO calling via gateway to 0$1"/>
   <action application="bridge" data="sofia/gateway/gsm/0$1"/>
   </condition>
</extension>
*This is not working as well, I got NORMAL_TEMPORARY_FAILURE.*
Please see the dinstar log named* setup_2.log*
*NB*. The DID from PSTN to Freeswitch via Dinstar GSM gateway is working
fine normal audio.
I am able to provide the configuration of dinstar gateway as well if you
might want to see it.

Any solution would be really appreciated.

Best regards,
Chhorm Chhatra
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From nycphoneservice at gmail.com  Sun Feb 10 14:39:47 2019
From: nycphoneservice at gmail.com (NYCPhoneService)
Date: Sun, 10 Feb 2019 09:39:47 -0500
Subject: [Freeswitch-users] ICTFax version 4.0 ,
	Open source Fax over Ip software
In-Reply-To: <CAOL0t+KOf7bir+G5FOvuqggjirvJ=FJiSikLtF08hLsaRg_QNA@mail.gmail.com>
References: <CAOL0t+KOf7bir+G5FOvuqggjirvJ=FJiSikLtF08hLsaRg_QNA@mail.gmail.com>
Message-ID: <00dc01d4c14e$7902e630$6b08b290$@gmail.com>

Tahir – thank you for great product!

I installed it (although some rpms/deps were missing from ictfax package and had to do some tweaking) and was able to successfully send faxes.

There are some issues exist in the new gui and I have a few questions:

1.       Is there a user support forum or irc channel?  The ictfax.com/forum is not active.

2.       If we find some bugs, suggest improvements – should we post on the github?

3.       Github version of ictfax is completely different from the rpm version on your site. Which one should we use?

Thanks again for the great effort!

Dennis

 

From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Tahir Almas
Sent: Saturday, February 9, 2019 5:20 AM
To: FreeSWITCH Users Help
Cc: freeswitch-biz at lists.freeswitch.org
Subject: [Freeswitch-users] ICTFax version 4.0 , Open source Fax over Ip software

 

We are pleased to announce  that  ICTFax 4.0 has been released  developed over freeswitch , ictcore and angular framework.  New features include  ATA / Extension support  and REST API 

 

http://ictfax.org/ictfax-4.0-released-with-ata-api-support

 

Regards

Tahir Almas

Managing Partner
ICT Innovations
http://www.ictinnovations.com
Leveraging open source in ICT

  

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From ynasida at gmail.com  Tue Feb 12 17:07:19 2019
From: ynasida at gmail.com (Yuriy Nasida)
Date: Tue, 12 Feb 2019 20:07:19 +0300
Subject: [Freeswitch-users] where g729 fs-latest-installer script now ?
Message-ID: <CADihe3K9Cvhjtpdox-QXO-KUSHew-aey16zirECudPUQ6kNYpA@mail.gmail.com>

Hi guys,

Does anybody know why site http://files.freeswitch.org/g729/ contain only
install instrunction now ?  Where fs-latest-installer-v1.6 script or
something like this ?

Please advice.
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From kbhackshall at gmail.com  Wed Feb 13 06:07:27 2019
From: kbhackshall at gmail.com (Kieran Hackshall)
Date: Wed, 13 Feb 2019 16:07:27 +1000
Subject: [Freeswitch-users] Count active calls on individual gateways
Message-ID: <CAOXJZEmrxq_PmA79biw2hQL4qpHkASWJxQJjm28rL0k9wv1Xsw@mail.gmail.com>

Looking to get a count of active calls on a particular gateway for outbound
calls..

Running "show calls" and "show channels" I cannot see anything that would
allow me to determine what gateway is being used.

Is there another way to do this?

Cheers,
Kieran
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From ynasida at gmail.com  Wed Feb 13 09:25:54 2019
From: ynasida at gmail.com (Yuriy Nasida)
Date: Wed, 13 Feb 2019 12:25:54 +0300
Subject: [Freeswitch-users] where g729 fs-latest-installer script now ?
In-Reply-To: <CADihe3K9Cvhjtpdox-QXO-KUSHew-aey16zirECudPUQ6kNYpA@mail.gmail.com>
References: <CADihe3K9Cvhjtpdox-QXO-KUSHew-aey16zirECudPUQ6kNYpA@mail.gmail.com>
Message-ID: <CADihe3+UAjHwHJyz2+Co1+oDnvu=79PhBNfVZM5jh+Vy3OOzNA@mail.gmail.com>

Um... Nobody uses g729 licences anymore ? :)

On Tue, 12 Feb 2019 at 20:07, Yuriy Nasida <ynasida at gmail.com> wrote:

> Hi guys,
>
> Does anybody know why site http://files.freeswitch.org/g729/ contain only
> install instrunction now ?  Where fs-latest-installer-v1.6 script or
> something like this ?
>
> Please advice.
>
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From ynasida at gmail.com  Wed Feb 13 10:36:13 2019
From: ynasida at gmail.com (Yuriy Nasida)
Date: Wed, 13 Feb 2019 13:36:13 +0300
Subject: [Freeswitch-users] where g729 fs-latest-installer script now ?
In-Reply-To: <CADihe3+UAjHwHJyz2+Co1+oDnvu=79PhBNfVZM5jh+Vy3OOzNA@mail.gmail.com>
References: <CADihe3K9Cvhjtpdox-QXO-KUSHew-aey16zirECudPUQ6kNYpA@mail.gmail.com>
 <CADihe3+UAjHwHJyz2+Co1+oDnvu=79PhBNfVZM5jh+Vy3OOzNA@mail.gmail.com>
Message-ID: <CADihe3KvzFs5X62R1EXjhY7k7+MEGjM63u7woyL-BXZaA-BZBw@mail.gmail.com>

Well. Somebody updated http://files.freeswitch.org/g729/ and now it
has fs-latest-installer. Thanks a lot whoever you are!!!
But... Now it has not install instructions  :)

On Wed, 13 Feb 2019 at 12:25, Yuriy Nasida <ynasida at gmail.com> wrote:

> Um... Nobody uses g729 licences anymore ? :)
>
> On Tue, 12 Feb 2019 at 20:07, Yuriy Nasida <ynasida at gmail.com> wrote:
>
>> Hi guys,
>>
>> Does anybody know why site http://files.freeswitch.org/g729/ contain
>> only install instrunction now ?  Where fs-latest-installer-v1.6 script or
>> something like this ?
>>
>> Please advice.
>>
>
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From ynasida at gmail.com  Wed Feb 13 17:10:18 2019
From: ynasida at gmail.com (Yuriy Nasida)
Date: Wed, 13 Feb 2019 20:10:18 +0300
Subject: [Freeswitch-users] where g729 fs-latest-installer script now ?
In-Reply-To: <CADihe3KvzFs5X62R1EXjhY7k7+MEGjM63u7woyL-BXZaA-BZBw@mail.gmail.com>
References: <CADihe3K9Cvhjtpdox-QXO-KUSHew-aey16zirECudPUQ6kNYpA@mail.gmail.com>
 <CADihe3+UAjHwHJyz2+Co1+oDnvu=79PhBNfVZM5jh+Vy3OOzNA@mail.gmail.com>
 <CADihe3KvzFs5X62R1EXjhY7k7+MEGjM63u7woyL-BXZaA-BZBw@mail.gmail.com>
Message-ID: <CADihe3LQwqFQ9vLqU34X_taK0HEf2etp=3CO1S9YkQFCy6wSag@mail.gmail.com>

I have installed freeswitch-license-server  using
fs-201902072050-installer  and
got issue below.
FS doesn't listen port 8021 and I can't connect to it and run any command
using fs_cli.
All works fine in case I kill process freeswitch-license-server.

Can anybody comment please all these very weird things with commercial g729
at FS ?
Thanks.

On Wed, 13 Feb 2019 at 13:36, Yuriy Nasida <ynasida at gmail.com> wrote:

> Well. Somebody updated http://files.freeswitch.org/g729/ and now it
> has fs-latest-installer. Thanks a lot whoever you are!!!
> But... Now it has not install instructions  :)
>
> On Wed, 13 Feb 2019 at 12:25, Yuriy Nasida <ynasida at gmail.com> wrote:
>
>> Um... Nobody uses g729 licences anymore ? :)
>>
>> On Tue, 12 Feb 2019 at 20:07, Yuriy Nasida <ynasida at gmail.com> wrote:
>>
>>> Hi guys,
>>>
>>> Does anybody know why site http://files.freeswitch.org/g729/ contain
>>> only install instrunction now ?  Where fs-latest-installer-v1.6 script or
>>> something like this ?
>>>
>>> Please advice.
>>>
>>
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From m.shirazi at gmail.com  Thu Feb 14 05:36:29 2019
From: m.shirazi at gmail.com (Mehdi Shirazi)
Date: Thu, 14 Feb 2019 09:06:29 +0330
Subject: [Freeswitch-users] Freeswitch As SBC for sip_phones
Message-ID: <CAG2rjpvK5xo6uToUrJ3uF83wk+64CbqKORgjmW-ntT8zmNky2g@mail.gmail.com>

Hi
Thanks for suggestions, If we want to comply strictly with rfc 5853 in our
SBC so we should have a B2BUA also int it.
What is your recommendation?
1-Using Kamailio/Opensips for forwarding REGISTER/SUBSCRIBE/NOTIFY to
external Registrar and using a Freeswitch for media+B2BUA  ?
2-Using Kamailio/Opensips + rtpengine and just using Freeswitch as B2BUA in
media bypass mode
Regards.
M.shirazi
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From ch.chhatra at gmail.com  Thu Feb 14 08:28:49 2019
From: ch.chhatra at gmail.com (Chhorm Chhatra)
Date: Thu, 14 Feb 2019 15:28:49 +0700
Subject: [Freeswitch-users] NORMAL_TEMPORARY_FAILURE Freeswitch Outbound
 via Gateway to PSTN
In-Reply-To: <CALXCt0rnyduiSTtnpFZ8CGktNLB16dZ+kWhKuSB_0o6+ovjCvw@mail.gmail.com>
References: <CAJssr-gzzjqKyWOTLJWS1hvR_85k8N21XpOc1_LX8GTU=sc_nw@mail.gmail.com>
 <CALXCt0rnyduiSTtnpFZ8CGktNLB16dZ+kWhKuSB_0o6+ovjCvw@mail.gmail.com>
Message-ID: <CAJssr-i3Nj4oOnp=5CCrKzh7jpXX6V9g=cdGBtsESLfJYGtYYA@mail.gmail.com>

Dear Giovanni,
Thank you for your response.
Could you please let me know if this issue relates to Dinstar Gateway
device or is there any configuration left that I have to do with Freeswitch
itself?
Please kindly see the following configuration.
I follow this Freeswitch Dinstar How-To Guide
<https://freeswitch.org/confluence/display/FREESWITCH/Dinstar+GSM+gateway+FreeSwitch+HowTo>.
Since the tutorial is not working for the outbound call to PSTN, I followed
the DWG_WITH_FREESWITCH.pdf instead. Unfortunately, both of them not
working for me with outbound calls.
I tried with the following configuration:
*1. With Dinstar GSM Gateway IP*
 <extension name="dwg2008_gateway">
   <condition field="destination_number" expression="^0(\d+)$">
   <action application="answer"/>
   <action application="set" data="ringback=${us-ring}"/>
   <action application="log" data="INFO calling via gateway to 0$1"/>
   <action application="bridge" data="sofia/internal/0$1 at 192.168.7.137"/>
  </condition>
</extension>
*But this is not working, I got NORMAL_TEMPORARY_FAILURE.*
Please see the dinstar log named* setup_1.log*

*2. With GSM*
<extension name="dwg2008_gateway">
   <condition field="destination_number" expression="^0(\d+)$">
   <action application="answer"/>
   <action application="set" data="ringback=${us-ring}"/>
   <action application="log" data="INFO calling via gateway to 0$1"/>
   <action application="bridge" data="sofia/gateway/gsm/0$1"/>
   </condition>
</extension>
*This is not working as well, I got NORMAL_TEMPORARY_FAILURE.*
Please see the dinstar log named* setup_2.log*

Best regards,
Chhorm Chhatra



On Thu, 14 Feb 2019 at 15:22, Giovanni Maruzzelli <gmaruzz at gmail.com> wrote:

> On Thu, Feb 14, 2019 at 9:15 AM Chhorm Chhatra <ch.chhatra at gmail.com>
> wrote:
>
>> Hi,
>> Currently, I faced the issue of getting NORMAL_TEMPORARY_FAILURE when
>> FreeSWITCH is originating outbound calls to the PSTN via Dinstar GSM
>> gateway device.
>>
>
> check what number you are sending the call to
>
> probably freeswitch thinks is an internal extension, or such
>
> be sure to have a dialplan extension that sends the number outbound to the
> gateway
>
> -giovanni
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From brandon.youngdale at gmail.com  Thu Feb 14 21:50:33 2019
From: brandon.youngdale at gmail.com (Brandon Youngdale)
Date: Thu, 14 Feb 2019 13:50:33 -0800
Subject: [Freeswitch-users] missing allison/digits/40.wav audio file
Message-ID: <CAB3JyNw=NkPsuW76fDtHBe9j+9iFtPG4s129UqM1=Nq=77eQsw@mail.gmail.com>

Hi,

Is there a reason there is no '40.wav' for the allison recordings? I
noticed on my install when I got an error that it was not in the digits
directory, and when I looked it is not in the repo either.

https://freeswitch.org/stash/projects/FS/repos/freeswitch-sounds/browse/en/us/
allison/digits/48000

Thank you,
-- 
Brandon Youngdale
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From tafali at gmail.com  Fri Feb 15 08:17:32 2019
From: tafali at gmail.com (Mustafa Ali Kahraman)
Date: Fri, 15 Feb 2019 11:17:32 +0300
Subject: [Freeswitch-users] How to find mistake in FS config?
In-Reply-To: <28CBC1C2-F608-4A6F-B7C9-69A221B845FC@gmx.net>
References: <94b9c25a-78b5-ac8b-42f2-7a74b855e017@yu-boot.ru>
 <28CBC1C2-F608-4A6F-B7C9-69A221B845FC@gmx.net>
Message-ID: <CAGyNz7scRn_owFEPP9udX+muFU3E53is51WUzoA6kaYnnDNORw@mail.gmail.com>

freeswitch.xml.fsxml

Sebastian Kemper <sebastian_ml at gmx.net>, 15 Şub 2019 Cum, 10:24 tarihinde
şunu yazdı:

> Hi Yu,
>
> Check your log directory. FS generates an XML file with the complete
> config and saves it in the log dir. The line should be in there.
>
> Regards,
> Seb
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com



-- 
-tafali-
www.tafali.net
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From nathan at robotics.net  Mon Feb 11 17:37:57 2019
From: nathan at robotics.net (Nathan Stratton)
Date: Mon, 11 Feb 2019 12:37:57 -0500
Subject: [Freeswitch-users] Hold BUG
Message-ID: <CAHubkyM_+0m6PEth6mD+Yj+8sUzMyZ9d4fHJ=BDWhh_8SVvvFg@mail.gmail.com>

Larry calls Bob
Bob puts Larry on hold this sends UPDATE to FreeSWITCH with setup:actpass
media attribute
200 OK from FreeSWITCH responds correctly with setup:active everything
works correctly

Larry calls Bob
Larry puts Bob on hold this send UPDATE to FreeSWITCH with
setup:actpass media attribute
200 OK from FreeSWITCH responds incorrectly with setup:actpass this causes
chrome to kill the call.

The client is lastest JsSIP

><>
nathan stratton
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From ingjfnc at gmail.com  Mon Feb 18 21:46:13 2019
From: ingjfnc at gmail.com (Juan Nino)
Date: Tue, 19 Feb 2019 08:46:13 +1100
Subject: [Freeswitch-users] disable MOH
In-Reply-To: <CAPr4L2-a7B5_w7NBoPP4u5BfS7DRmvCkpWpU5BGssB2Z3L=4Sw@mail.gmail.com>
References: <mailman.171.1550489236.1400.freeswitch-users@lists.freeswitch.org>
 <CAPr4L2_g-PD+XU3bWWkzwEkG5OwNTnTjib_xtWy9vaAD_fWDmQ@mail.gmail.com>
 <CAPr4L2-a7B5_w7NBoPP4u5BfS7DRmvCkpWpU5BGssB2Z3L=4Sw@mail.gmail.com>
Message-ID: <CAPr4L2_WjubN=NfZb4o-bDXG+e8d9kFm3jxfJPk_p_Tt=T-zdA@mail.gmail.com>

I am trying to get freeswitch to ignore re invites and don't play any MOH as
>>> the MOH already comes from the PBX, even when disabling it I still see in
>>> the logs that is trying to play MOH
>>> 2019-02-18 17:24:49.427916 [DEBUG] switch_ivr.c:623
>>> sofia/rtp-dirty/%2B12345 at x.x.x.x:5060 Command Execute
>>> playback(local_stream://moh)
>>>
>>> EXECUTE sofia/rtp-dirty/%2B12345 at x.x.x.x:5060
>>> playback(local_stream://moh)
>>>
>>> any idea what has to be done to get freeswitch to continue passing RTP
>>> without trying to impose its own?
>>>
>>> thanks
>>>
>>
>>
>
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From social at bohboh.info  Mon Feb 18 22:46:53 2019
From: social at bohboh.info (Social Boh)
Date: Mon, 18 Feb 2019 17:46:53 -0500
Subject: [Freeswitch-users] where g729 fs-latest-installer script now ?
In-Reply-To: <CADihe3KvzFs5X62R1EXjhY7k7+MEGjM63u7woyL-BXZaA-BZBw@mail.gmail.com>
References: <CADihe3K9Cvhjtpdox-QXO-KUSHew-aey16zirECudPUQ6kNYpA@mail.gmail.com>
 <CADihe3+UAjHwHJyz2+Co1+oDnvu=79PhBNfVZM5jh+Vy3OOzNA@mail.gmail.com>
 <CADihe3KvzFs5X62R1EXjhY7k7+MEGjM63u7woyL-BXZaA-BZBw@mail.gmail.com>
Message-ID: <22578eaa-ca5d-8dde-c1ed-0d21f7c07129@bohboh.info>

Spanish instructions about mod_bcg729:

*cd /usr/src/*

Se descarga el programa para la instalación de bcg729:

*git clone https://github.com/xadhoom/mod_bcg729.git*

se entra en la carpeta creada:

*cd mod_bcg729/*

*nano Makefile*

las dos lineas que siguen se modifican si no corresponden a las carpetas 
donde se encuentran los datos que necesita el programa para compilar y 
luego instalar el modulo en la ubicación correcta:

/*FS_INCLUDES=/usr/include/freeswitch*/

/*FS_MODULES=/usr/lib/freeswitch/mod*/

En nuestro caso las carpetas indicadas son las correctas pues no hace 
falta realizar cambios. Se cierra el archivo y se continua con:

*make*

por ultimo se instala el modulo

*make install*

Ahora se modifica el archivo de configuración que contiene la lista de 
módulos que FS carga cuando se inicia:

*nano /etc/freeswitch/autoload_configs/modules.conf.xml*

se modifica esta linea (Linea 94):

/*<load module="mod_g729"/>*/

para que quede:

/*<load module="mod_bcg729"/>*/

Como estos cambios aplicarán solamente cuando se reinicie FS para 
aplicarlos en seguida, desde la consola de FreeSWITCH primero se 
deshabilita el Codec G729 que viene con la instalación de FS:

freeswitch at voztovoice.org> *unload mod_g729*

y luego se carga el nuevo:

freeswitch at voztovoice.org> *load mod_bcg729*

---
I'm SoCIaL, MayBe

El 13/02/2019 a las 05:36, Yuriy Nasida escribió:
> Well. Somebody updated http://files.freeswitch.org/g729/ and now it 
> has fs-latest-installer. Thanks a lot whoever you are!!!
> But... Now it has not install instructions  :)
>
> On Wed, 13 Feb 2019 at 12:25, Yuriy Nasida <ynasida at gmail.com 
> <mailto:ynasida at gmail.com>> wrote:
>
>     Um... Nobody uses g729 licences anymore ? :)
>
>     On Tue, 12 Feb 2019 at 20:07, Yuriy Nasida <ynasida at gmail.com
>     <mailto:ynasida at gmail.com>> wrote:
>
>         Hi guys,
>
>         Does anybody know why site http://files.freeswitch.org/g729/
>         contain only install instrunction now ?
>         Where fs-latest-installer-v1.6 script or something like this ?
>
>         Please advice.
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From krice at freeswitch.org  Tue Feb 19 02:02:37 2019
From: krice at freeswitch.org (Ken Rice)
Date: Mon, 18 Feb 2019 20:02:37 -0600
Subject: [Freeswitch-users] where g729 fs-latest-installer script now ?
In-Reply-To: <22578eaa-ca5d-8dde-c1ed-0d21f7c07129@bohboh.info>
References: <CADihe3K9Cvhjtpdox-QXO-KUSHew-aey16zirECudPUQ6kNYpA@mail.gmail.com>
 <CADihe3+UAjHwHJyz2+Co1+oDnvu=79PhBNfVZM5jh+Vy3OOzNA@mail.gmail.com>
 <CADihe3KvzFs5X62R1EXjhY7k7+MEGjM63u7woyL-BXZaA-BZBw@mail.gmail.com>
 <22578eaa-ca5d-8dde-c1ed-0d21f7c07129@bohboh.info>
Message-ID: <70A62093-42C6-4ACA-A627-6DE65043EF36@freeswitch.org>

just no. and i mean no.

seriously Anthon gives you literally millions of dollars worth of code and 1 module that has a very little cost you want to go out and use illicit code for it? that code is not free.

sure the patents might be done with but developers somewhere still own the copyrights on that specific g729 code. 
they have bills to pay. 
they have food to buy.
they have internet access to pay for.
Many have children that like to eat and wear clothing.

just because some code is free as in doesn't cost hard money, not all code is that way. 



Sent from my iPhone

> On Feb 18, 2019, at 16:46, Social Boh <social at bohboh.info> wrote:
> 
> Spanish instructions about mod_bcg729:
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From krice at freeswitch.org  Tue Feb 19 02:06:30 2019
From: krice at freeswitch.org (Ken Rice)
Date: Mon, 18 Feb 2019 20:06:30 -0600
Subject: [Freeswitch-users] Hold BUG
In-Reply-To: <CAHubkyM_+0m6PEth6mD+Yj+8sUzMyZ9d4fHJ=BDWhh_8SVvvFg@mail.gmail.com>
References: <CAHubkyM_+0m6PEth6mD+Yj+8sUzMyZ9d4fHJ=BDWhh_8SVvvFg@mail.gmail.com>
Message-ID: <A9222D29-58F7-4F22-97F4-FBCAB48CB660@freeswitch.org>

bugs should be reported to jira so that any discussion, logs, patches, etc all remain together

Sent from my iPhone

> On Feb 11, 2019, at 11:37, Nathan Stratton <nathan at robotics.net> wrote:
> 
> 
> Larry calls Bob
> Bob puts Larry on hold this sends UPDATE to FreeSWITCH with setup:actpass media attribute
> 200 OK from FreeSWITCH responds correctly with setup:active everything works correctly
> 
> Larry calls Bob
> Larry puts Bob on hold this send UPDATE to FreeSWITCH with setup:actpass media attribute
> 200 OK from FreeSWITCH responds incorrectly with setup:actpass this causes chrome to kill the call.
> 
> The client is lastest JsSIP
> 
> ><>
> nathan stratton
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
> 
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com


From mike at freeswitch.com  Tue Feb 19 02:08:04 2019
From: mike at freeswitch.com (Mike Jerris)
Date: Mon, 18 Feb 2019 21:08:04 -0500
Subject: [Freeswitch-users] Hold BUG
In-Reply-To: <A9222D29-58F7-4F22-97F4-FBCAB48CB660@freeswitch.org>
References: <CAHubkyM_+0m6PEth6mD+Yj+8sUzMyZ9d4fHJ=BDWhh_8SVvvFg@mail.gmail.com>
 <A9222D29-58F7-4F22-97F4-FBCAB48CB660@freeswitch.org>
Message-ID: <CADrq_yHfPUA9WBfUVtRS=k4pWDo_Ea9T2=dM8S8ahWkQ=WkMvA@mail.gmail.com>

After you try master...

On Mon, Feb 18, 2019 at 9:07 PM Ken Rice <krice at freeswitch.org> wrote:

> bugs should be reported to jira so that any discussion, logs, patches, etc
> all remain together
>
> Sent from my iPhone
>
> > On Feb 11, 2019, at 11:37, Nathan Stratton <nathan at robotics.net> wrote:
> >
> >
> > Larry calls Bob
> > Bob puts Larry on hold this sends UPDATE to FreeSWITCH with
> setup:actpass media attribute
> > 200 OK from FreeSWITCH responds correctly with setup:active everything
> works correctly
> >
> > Larry calls Bob
> > Larry puts Bob on hold this send UPDATE to FreeSWITCH with setup:actpass
> media attribute
> > 200 OK from FreeSWITCH responds incorrectly with setup:actpass this
> causes chrome to kill the call.
> >
> > The client is lastest JsSIP
> >
> > ><>
> > nathan stratton
> > _________________________________________________________________________
> > Professional FreeSWITCH Services
> > sales at freeswitch.com
> > https://freeswitch.com
> >
> > Official FreeSWITCH Sites
> > https://freeswitch.com/oss
> > https://freeswitch.org/confluence
> > https://cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > https://freeswitch.com
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From adrian.worutowicz at esifrance.net  Tue Feb 19 09:40:10 2019
From: adrian.worutowicz at esifrance.net (Adrian Worutowicz)
Date: Tue, 19 Feb 2019 09:40:10 +0000
Subject: [Freeswitch-users] How to disable FS DTMF detection
Message-ID: <96cd241a18314d39b455b3e96fb100e5@SRVEXCHANGE02.esifrance.net>

Hi,

I've developed a couple of endpoints and I use bgapi originate & bridge.
Free Switch intercepts and interprets DTMF and I do not want that.

In the dialplan I tried:

<param name="pass-rfc2833" value="true"/>
<action application="set" data="pass_rfc2833=true"/>

<action application="set" data="exec_after_bridge_app=stop_dtmf"/>
<action application="stop_dtmf">

<action application="spandsp_stop_dtmf"/>

None of those methods works.

Maybe there is a parameter in originate that I should use?

*** How to tell Free Switch to leave alone the DTMF tones? ***

I'll appreciate any help,
Adrian.
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From gmaruzz at gmail.com  Tue Feb 19 11:05:04 2019
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Tue, 19 Feb 2019 12:05:04 +0100
Subject: [Freeswitch-users] Freeswitch As SBC for sip_phones
In-Reply-To: <CAG2rjpvK5xo6uToUrJ3uF83wk+64CbqKORgjmW-ntT8zmNky2g@mail.gmail.com>
References: <CAG2rjpvK5xo6uToUrJ3uF83wk+64CbqKORgjmW-ntT8zmNky2g@mail.gmail.com>
Message-ID: <CALXCt0qjqhODjwdDy3hRex_Z6x7C5XCUfWwKee06==7UauH5Gg@mail.gmail.com>

On Mon, Feb 18, 2019 at 10:44 PM Mehdi Shirazi <m.shirazi at gmail.com> wrote:

> Hi
> Thanks for suggestions, If we want to comply strictly with rfc 5853 in our
> SBC so we should have a B2BUA also int it.
> What is your recommendation?
> 1-Using Kamailio/Opensips for forwarding REGISTER/SUBSCRIBE/NOTIFY to
> external Registrar and using a Freeswitch for media+B2BUA  ?
> 2-Using Kamailio/Opensips + rtpengine and just using Freeswitch as B2BUA
> in media bypass mode
> Regards.
> M.shirazi
>
>
Hello Mehdi,

both solution are (can be built as) compliant.

But, depending on what are other priorities, one can be better suited.
Also, there still be one other possibility, to use the B2BUA
functionalities of the proxy, and use FS only for media active handling,
only when required (if rtpengine cannot do it).

In addition to that, you can also have interaction between proxy and
freeswitch, particularly in case of OpenSIPS, where the integration in
terms of feedback and control at the level of ESL is already made for you.

So, lot of options to choose from :).

Have a nice day,

-giovanni





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



-- 
Sincerely,

Giovanni Maruzzelli
OpenTelecom.IT
cell: +39 347 266 56 18
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From vma at vallimamod.org  Tue Feb 19 11:39:17 2019
From: vma at vallimamod.org (Vallimamod Abdullah)
Date: Tue, 19 Feb 2019 12:39:17 +0100
Subject: [Freeswitch-users] How to disable FS DTMF detection
In-Reply-To: <96cd241a18314d39b455b3e96fb100e5@SRVEXCHANGE02.esifrance.net>
References: <96cd241a18314d39b455b3e96fb100e5@SRVEXCHANGE02.esifrance.net>
Message-ID: <54875FA9-519F-43E5-8E15-F0A6A39B72F8@vallimamod.org>

Hi,

You need to put the 
  <param name="pass-rfc2833" value="true"/> 
in your sip profile like all other params.

It is not necessary however as you can just put
  <action application="export" data="pass_rfc2833=true"/>
in your dialplan, notice the export instead of set.

The spandsp_stop_dtmf only works for inband dtmf, not the rfc2833 one.

Hope this helps.


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


> On 19 Feb 2019, at 10:40, Adrian Worutowicz <adrian.worutowicz at esifrance.net> wrote:
> 
> Hi,
>  
> I’ve developed a couple of endpoints and I use bgapi originate & bridge.
> Free Switch intercepts and interprets DTMF and I do not want that.
>  
> In the dialplan I tried:
>  
> <param name="pass-rfc2833" value="true"/>
> <action application="set" data="pass_rfc2833=true"/>
>  
> <action application="set" data="exec_after_bridge_app=stop_dtmf"/>
> <action application="stop_dtmf">
>  
> <action application="spandsp_stop_dtmf"/>
>  
> None of those methods works.
>  
> Maybe there is a parameter in originate that I should use?
>  
> *** How to tell Free Switch to leave alone the DTMF tones? ***
>  
> I’ll appreciate any help,
> Adrian.
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
> 
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com



From adrian.worutowicz at esifrance.net  Tue Feb 19 13:02:53 2019
From: adrian.worutowicz at esifrance.net (Adrian Worutowicz)
Date: Tue, 19 Feb 2019 13:02:53 +0000
Subject: [Freeswitch-users] How to disable FS DTMF detection
In-Reply-To: <54875FA9-519F-43E5-8E15-F0A6A39B72F8@vallimamod.org>
References: <96cd241a18314d39b455b3e96fb100e5@SRVEXCHANGE02.esifrance.net>
 <54875FA9-519F-43E5-8E15-F0A6A39B72F8@vallimamod.org>
Message-ID: <a4b0993885594e58a446841cc9d25199@SRVEXCHANGE02.esifrance.net>

Thank you very much!

Do you know how to stop Free Switch from intercepting inband DTMFs ?

Adrian.

-----Message d'origine-----
De : FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Vallimamod Abdullah
Envoyé : mardi 19 février 2019 12:39
À : FreeSWITCH Users Help
Objet : Re: [Freeswitch-users] How to disable FS DTMF detection

Hi,

You need to put the
  <param name="pass-rfc2833" value="true"/> in your sip profile like all other params.

It is not necessary however as you can just put
  <action application="export" data="pass_rfc2833=true"/> in your dialplan, notice the export instead of set.

The spandsp_stop_dtmf only works for inband dtmf, not the rfc2833 one.

Hope this helps.


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


> On 19 Feb 2019, at 10:40, Adrian Worutowicz <adrian.worutowicz at esifrance.net> wrote:
> 
> Hi,
>  
> I’ve developed a couple of endpoints and I use bgapi originate & bridge.
> Free Switch intercepts and interprets DTMF and I do not want that.
>  
> In the dialplan I tried:
>  
> <param name="pass-rfc2833" value="true"/>
> <action application="set" data="pass_rfc2833=true"/>
>  
> <action application="set" data="exec_after_bridge_app=stop_dtmf"/>
> <action application="stop_dtmf">
>  
> <action application="spandsp_stop_dtmf"/>
>  
> None of those methods works.
>  
> Maybe there is a parameter in originate that I should use?
>  
> *** How to tell Free Switch to leave alone the DTMF tones? ***
>  
> I’ll appreciate any help,
> Adrian.
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
> 
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com


_________________________________________________________________________
Professional FreeSWITCH Services
sales at freeswitch.com
https://freeswitch.com

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

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

From ceniy.mei at gmail.com  Tue Feb 19 07:32:42 2019
From: ceniy.mei at gmail.com (=?UTF-8?B?5qKF6KGo6YC1?=)
Date: Tue, 19 Feb 2019 15:32:42 +0800
Subject: [Freeswitch-users] [ERR] too many stacked extensions
Message-ID: <CABcV8PHtEAsHXKKKWnsas+u63Wy2FKYgQUWqa-Jys7mMmSpr1g@mail.gmail.com>

Hi,


I’m running freeswitch version 1.8.1. In the log files, I saw a lot of
[ERR] too many stacked extensions. It happens when freeswitch is trying to
bridge  a callcenter member to agent.  I don’t know how to handle it. Could
you please give any advises?



55791bf8-30bb-11e9-a4fd-7125b8df8c7a 2019-02-15 08:47:56.419673 [DEBUG]
switch_ivr_bridge.c:1063 sofia/internal/137095 at 10.20.86.100:5060 CUSTOM HOLD
55791bf8-30bb-11e9-a4fd-7125b8df8c7a 2019-02-15 08:47:56.419673 [DEBUG]
switch_core_state_machine.c:662 (sofia/internal/137095 at 10.20.86.100:5060)
State CONSUME_MEDIA going to sleep
55791bf8-30bb-11e9-a4fd-7125b8df8c7a 2019-02-15 08:47:56.419673 [ERR]
switch_ivr.c:902 Error sofia/internal/137095 at 10.20.86.100:5060 too many
stacked extensions
55791bf8-30bb-11e9-a4fd-7125b8df8c7a 2019-02-15 08:47:56.419673 [ERR]
switch_ivr.c:902 Error sofia/internal/137095 at 10.20.86.100:5060 too many
stacked extensions
55791bf8-30bb-11e9-a4fd-7125b8df8c7a 2019-02-15 08:47:56.419673 [DEBUG]
switch_ivr_bridge.c:1744 (sofia/internal/137095 at 10.20.86.100:5060) State
Change CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA
55791bf8-30bb-11e9-a4fd-7125b8df8c7a 2019-02-15 08:47:56.419673 [ERR]
switch_ivr.c:902 Error sofia/internal/137095 at 10.20.86.100:5060 too many
stacked extensions
55791bf8-30bb-11e9-a4fd-7125b8df8c7a 2019-02-15 08:47:56.419673 [ERR]
switch_ivr.c:902 Error sofia/internal/137095 at 10.20.86.100:5060 too many
stacked extensions
55791bf8-30bb-11e9-a4fd-7125b8df8c7a 2019-02-15 08:47:56.419673 [DEBUG]
switch_core_state_machine.c:584 (sofia/internal/137095 at 10.20.86.100:5060)
Running State Change CS_EXCHANGE_MEDIA (Cur 2 Tot 4919)
55791bf8-30bb-11e9-a4fd-7125b8df8c7a 2019-02-15 08:47:56.419673 [DEBUG]
switch_core_state_machine.c:653 (sofia/internal/137095 at 10.20.86.100:5060)
State EXCHANGE_MEDIA
55791bf8-30bb-11e9-a4fd-7125b8df8c7a 2019-02-15 08:47:56.419673 [DEBUG]
mod_sofia.c:645 SOFIA EXCHANGE_MEDIA


XML:
<condition field="destination_number" expression="^(31\d{4})$" >
   <action application="set" data="dialed_extension=$1"/>
   <!-- hangup after successful bridge to agent -->
   <action application="set" data="recording_follow_transfer=true"/>
   <action application="set" data="hangup_after_bridge=true"/>
   <action application="set" data="say_agent_first=1" />
   <action application="set" data="bridge_pre_execute_aleg_app=lua"/>
   <action application="set"
data="bridge_pre_execute_aleg_data=broadcastAgent.lua"/>
  <action application="set" data="call_type=q" />
  <action application="bind_meta_app" data="4 ab s
execute_extension::att_xfer XML features"/>
  <action application="callcenter" data="$1"/>
</condition>
</extension>


Thanks
Ceniy Mei
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From asilva at wirelessmundi.com  Tue Feb 19 08:41:24 2019
From: asilva at wirelessmundi.com (=?UTF-8?Q?Ant=c3=b3nio_Silva?=)
Date: Tue, 19 Feb 2019 09:41:24 +0100
Subject: [Freeswitch-users] where g729 fs-latest-installer script now ?
In-Reply-To: <CADihe3LQwqFQ9vLqU34X_taK0HEf2etp=3CO1S9YkQFCy6wSag@mail.gmail.com>
References: <CADihe3K9Cvhjtpdox-QXO-KUSHew-aey16zirECudPUQ6kNYpA@mail.gmail.com>
 <CADihe3+UAjHwHJyz2+Co1+oDnvu=79PhBNfVZM5jh+Vy3OOzNA@mail.gmail.com>
 <CADihe3KvzFs5X62R1EXjhY7k7+MEGjM63u7woyL-BXZaA-BZBw@mail.gmail.com>
 <CADihe3LQwqFQ9vLqU34X_taK0HEf2etp=3CO1S9YkQFCy6wSag@mail.gmail.com>
Message-ID: <4af1376d-29ba-ea47-c952-c4883b7dd654@wirelessmundi.com>

try using the .deb and installed then manuall.:

Get the packages from here: 
http://files.freeswitch.org/repo/deb/freeswitch-1.8/pool/main/f/

Installed them manually:

dpkg -x freeswitch-mod-com-g729_1.8.5~7~14d59c97e0~jessie_amd64.deb /
dpkg -x freeswitch-license-server_0.0.1~42~731842be7d~jessie_amd64.deb /


I don't know if the installer will remain, from the jira 
https://freeswitch.org/jira/browse/FS-11641 i got the impression that 
now the solution is to use the debian packages.


On 13/02/2019 18:10, Yuriy Nasida wrote:
> I have installed freeswitch-license-server using 
> fs-201902072050-installer and got issue below.
> FS doesn't listen port 8021 and I can't connect to it and run any 
> command using fs_cli.
> All works fine in case I kill process freeswitch-license-server.
>
> Can anybody comment please all these very weird things with commercial 
> g729 at FS ?
> Thanks.
>
> On Wed, 13 Feb 2019 at 13:36, Yuriy Nasida <ynasida at gmail.com 
> <mailto:ynasida at gmail.com>> wrote:
>
>     Well. Somebody updated http://files.freeswitch.org/g729/ and now
>     it has fs-latest-installer. Thanks a lot whoever you are!!!
>     But... Now it has not install instructions  :)
>
>     On Wed, 13 Feb 2019 at 12:25, Yuriy Nasida <ynasida at gmail.com
>     <mailto:ynasida at gmail.com>> wrote:
>
>         Um... Nobody uses g729 licences anymore ? :)
>
>         On Tue, 12 Feb 2019 at 20:07, Yuriy Nasida <ynasida at gmail.com
>         <mailto:ynasida at gmail.com>> wrote:
>
>             Hi guys,
>
>             Does anybody know why site
>             http://files.freeswitch.org/g729/ contain only install
>             instrunction now ? Where fs-latest-installer-v1.6 script
>             or something like this ?
>
>             Please advice.
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com

-- 
Saludos / Regards / Cumprimentos
António Silva

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From ynasida at gmail.com  Tue Feb 19 12:47:13 2019
From: ynasida at gmail.com (Yuriy Nasida)
Date: Tue, 19 Feb 2019 15:47:13 +0300
Subject: [Freeswitch-users] where g729 fs-latest-installer script now ?
In-Reply-To: <4af1376d-29ba-ea47-c952-c4883b7dd654@wirelessmundi.com>
References: <CADihe3K9Cvhjtpdox-QXO-KUSHew-aey16zirECudPUQ6kNYpA@mail.gmail.com>
 <CADihe3+UAjHwHJyz2+Co1+oDnvu=79PhBNfVZM5jh+Vy3OOzNA@mail.gmail.com>
 <CADihe3KvzFs5X62R1EXjhY7k7+MEGjM63u7woyL-BXZaA-BZBw@mail.gmail.com>
 <CADihe3LQwqFQ9vLqU34X_taK0HEf2etp=3CO1S9YkQFCy6wSag@mail.gmail.com>
 <4af1376d-29ba-ea47-c952-c4883b7dd654@wirelessmundi.com>
Message-ID: <CADihe3K-yyXFF8=w_7CLS3thpV9mbHnZWxbwR3TUOVy_k_nwsA@mail.gmail.com>

Ken, not sure what you talking about. I already bought a lot of g729
licenses for my servers and I use them a lot of years. But now I can't use
them normally at Deb 9 + fs1.8 + new g729 install script because of issues
I described.

Thanks a lot Antonio! I will try this way.

On Tue, 19 Feb 2019 at 11:41, António Silva <asilva at wirelessmundi.com>
wrote:

> try using the .deb and installed then manuall.:
>
> Get the packages from here:
> http://files.freeswitch.org/repo/deb/freeswitch-1.8/pool/main/f/
>
> Installed them manually:
>
> dpkg -x freeswitch-mod-com-g729_1.8.5~7~14d59c97e0~jessie_amd64.deb /
> dpkg -x freeswitch-license-server_0.0.1~42~731842be7d~jessie_amd64.deb /
>
>
> I don't know if the installer will remain, from the jira
> https://freeswitch.org/jira/browse/FS-11641 i got the impression that now
> the solution is to use the debian packages.
>
>
> On 13/02/2019 18:10, Yuriy Nasida wrote:
>
> I have installed freeswitch-license-server  using fs-201902072050-installer  and
> got issue below.
> FS doesn't listen port 8021 and I can't connect to it and run any command
> using fs_cli.
> All works fine in case I kill process freeswitch-license-server.
>
> Can anybody comment please all these very weird things with commercial
> g729 at FS ?
> Thanks.
>
> On Wed, 13 Feb 2019 at 13:36, Yuriy Nasida <ynasida at gmail.com> wrote:
>
>> Well. Somebody updated http://files.freeswitch.org/g729/ and now it
>> has fs-latest-installer. Thanks a lot whoever you are!!!
>> But... Now it has not install instructions  :)
>>
>> On Wed, 13 Feb 2019 at 12:25, Yuriy Nasida <ynasida at gmail.com> wrote:
>>
>>> Um... Nobody uses g729 licences anymore ? :)
>>>
>>> On Tue, 12 Feb 2019 at 20:07, Yuriy Nasida <ynasida at gmail.com> wrote:
>>>
>>>> Hi guys,
>>>>
>>>> Does anybody know why site http://files.freeswitch.org/g729/ contain
>>>> only install instrunction now ?  Where fs-latest-installer-v1.6 script or
>>>> something like this ?
>>>>
>>>> Please advice.
>>>>
>>>
> _________________________________________________________________________
> Professional FreeSWITCH Servicessales at freeswitch.comhttps://freeswitch.com
>
> Official FreeSWITCH Siteshttps://freeswitch.com/osshttps://freeswitch.org/confluencehttps://cluecon.com
>
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttps://freeswitch.com
>
> --
> Saludos / Regards / Cumprimentos
> António Silva
>
>
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From bilaln018 at gmail.com  Tue Feb 19 14:07:40 2019
From: bilaln018 at gmail.com (Bilal Abbasi)
Date: Tue, 19 Feb 2019 19:07:40 +0500
Subject: [Freeswitch-users] [FreeSWITCH Install Issue]
Message-ID: <CADSVerCzDkRK6mkj6D16Ois-ZynE_eTbiteH=Vm_oovvFBNcsQ@mail.gmail.com>

Hi Users,
I am having issues with installing FreeSWITCH 1.8.5 on Debian 9.6,
Here is the error i am getting this error when i run make command.

 CC       src/libfreeswitch_la-switch_msrp.lo

  CC       src/libfreeswitch_la-switch_vad.lo

  CC       src/libfreeswitch_la-switch_vpx.lo

  CXX      src/switch_cpp.lo

  CXXLD    libfreeswitch.la


*** Warning: Linking the shared library libfreeswitch.la against the

*** static library libs/libvpx/libvpx.a is not portable!

ar: `u' modifier ignored since `D' is the default (see `U')

Making all in .

make[2]: Entering directory '/usr/src/freeswitch-1.8.5'

  CC       src/freeswitch-switch.o

  CCLD     freeswitch

/usr/bin/ld: warning: libssl.so.1.0.2, needed by
//usr/lib/x86_64-linux-gnu/libcurl.so.4, may conflict with libssl.so.1.1

/usr/bin/ld: warning: libcrypto.so.1.0.2, needed by
//usr/lib/x86_64-linux-gnu/libcurl.so.4, may conflict with libcrypto.so.1.1

  CC       src/tone2wav-tone2wav.o

  CCLD     tone2wav

/usr/bin/ld: warning: libssl.so.1.0.2, needed by
//usr/lib/x86_64-linux-gnu/libcurl.so.4, may conflict with libssl.so.1.1

/usr/bin/ld: warning: libcrypto.so.1.0.2, needed by
//usr/lib/x86_64-linux-gnu/libcurl.so.4, may conflict with libcrypto.so.1.1

  CC       src/fs_encode-fs_encode.o

  CCLD     fs_encode

/usr/bin/ld: warning: libssl.so.1.0.2, needed by
//usr/lib/x86_64-linux-gnu/libcurl.so.4, may conflict with libssl.so.1.1

/usr/bin/ld: warning: libcrypto.so.1.0.2, needed by
//usr/lib/x86_64-linux-gnu/libcurl.so.4, may conflict with libcrypto.so.1.1

make[2]: Leaving directory '/usr/src/freeswitch-1.8.5'

Making all in src

make[2]: Entering directory '/usr/src/freeswitch-1.8.5/src'

Making all in mod

make[3]: Entering directory '/usr/src/freeswitch-1.8.5/src/mod'

Unknown target

Makefile:721: recipe for target '-all' failed

make[3]: *** [-all] Error 1

make[3]: Leaving directory '/usr/src/freeswitch-1.8.5/src/mod'

Makefile:591: recipe for target 'all-recursive' failed

make[2]: *** [all-recursive] Error 1

make[2]: Leaving directory '/usr/src/freeswitch-1.8.5/src'

Makefile:3494: recipe for target 'all-recursive' failed

make[1]: *** [all-recursive] Error 1

make[1]: Leaving directory '/usr/src/freeswitch-1.8.5'

Makefile:1255: recipe for target 'all' failed

make: *** [all] Error 2


Google did not helped me much on this, can somebody point me the right
direction.


Regards

Abbasi
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From brian at freeswitch.com  Tue Feb 19 14:13:56 2019
From: brian at freeswitch.com (Brian West)
Date: Tue, 19 Feb 2019 08:13:56 -0600
Subject: [Freeswitch-users] where g729 fs-latest-installer script now ?
In-Reply-To: <CADihe3KvzFs5X62R1EXjhY7k7+MEGjM63u7woyL-BXZaA-BZBw@mail.gmail.com>
References: <CADihe3K9Cvhjtpdox-QXO-KUSHew-aey16zirECudPUQ6kNYpA@mail.gmail.com>
 <CADihe3+UAjHwHJyz2+Co1+oDnvu=79PhBNfVZM5jh+Vy3OOzNA@mail.gmail.com>
 <CADihe3KvzFs5X62R1EXjhY7k7+MEGjM63u7woyL-BXZaA-BZBw@mail.gmail.com>
Message-ID: <CAEJMVkAhbja8F-WaD+-N_NQi111SLFuu_FYaf7SZadVes0n3pw@mail.gmail.com>

The one there is the latest, its no symlink, it may or may not work on 1.6,
please download and test.

/b

On Mon, Feb 18, 2019 at 3:44 PM Yuriy Nasida <ynasida at gmail.com> wrote:

> Well. Somebody updated http://files.freeswitch.org/g729/ and now it
> has fs-latest-installer. Thanks a lot whoever you are!!!
> But... Now it has not install instructions  :)
>
> On Wed, 13 Feb 2019 at 12:25, Yuriy Nasida <ynasida at gmail.com> wrote:
>
>> Um... Nobody uses g729 licences anymore ? :)
>>
>> On Tue, 12 Feb 2019 at 20:07, Yuriy Nasida <ynasida at gmail.com> wrote:
>>
>>> Hi guys,
>>>
>>> Does anybody know why site http://files.freeswitch.org/g729/ contain
>>> only install instrunction now ?  Where fs-latest-installer-v1.6 script or
>>> something like this ?
>>>
>>> Please advice.
>>>
>> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com



-- 

Brian West | Co-founder and Developer

Need Commercial support? email sales at freeswitch.com

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

Email: brian at freeswitch.com

Mobile: 918-424-9378

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

[image: https://www.facebook.com/signalwireinc?src=email]
<https://www.facebook.com/freeswitch> [image:
https://twitter.com/freeswitch] <https://twitter.com/freeswitch>
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From alex at freeswitch.com  Tue Feb 19 14:29:11 2019
From: alex at freeswitch.com (Alexey Sibyakin)
Date: Tue, 19 Feb 2019 23:29:11 +0900
Subject: [Freeswitch-users] [FreeSWITCH Install Issue]
In-Reply-To: <CADSVerCzDkRK6mkj6D16Ois-ZynE_eTbiteH=Vm_oovvFBNcsQ@mail.gmail.com>
References: <CADSVerCzDkRK6mkj6D16Ois-ZynE_eTbiteH=Vm_oovvFBNcsQ@mail.gmail.com>
Message-ID: <CAGJLwFhNka3iuVHAAmTy3bWibzpbxqzqyL+4_J858wgUN1gj2A@mail.gmail.com>

Hi,

Try our awesome packages
https://freeswitch.org/confluence/display/FREESWITCH/Debian+9+Stretch

You should compile yourself only if you absolutely know what you are doing.

It looks like you did *make -all* instead of *make *or *make all*.

Regards,

Alex

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



On Tue, Feb 19, 2019 at 11:16 PM Bilal Abbasi <bilaln018 at gmail.com> wrote:

> Hi Users,
> I am having issues with installing FreeSWITCH 1.8.5 on Debian 9.6,
> Here is the error i am getting this error when i run make command.
>
>  CC       src/libfreeswitch_la-switch_msrp.lo
>
>   CC       src/libfreeswitch_la-switch_vad.lo
>
>   CC       src/libfreeswitch_la-switch_vpx.lo
>
>   CXX      src/switch_cpp.lo
>
>   CXXLD    libfreeswitch.la
>
>
> *** Warning: Linking the shared library libfreeswitch.la against the
>
> *** static library libs/libvpx/libvpx.a is not portable!
>
> ar: `u' modifier ignored since `D' is the default (see `U')
>
> Making all in .
>
> make[2]: Entering directory '/usr/src/freeswitch-1.8.5'
>
>   CC       src/freeswitch-switch.o
>
>   CCLD     freeswitch
>
> /usr/bin/ld: warning: libssl.so.1.0.2, needed by
> //usr/lib/x86_64-linux-gnu/libcurl.so.4, may conflict with libssl.so.1.1
>
> /usr/bin/ld: warning: libcrypto.so.1.0.2, needed by
> //usr/lib/x86_64-linux-gnu/libcurl.so.4, may conflict with libcrypto.so.1.1
>
>   CC       src/tone2wav-tone2wav.o
>
>   CCLD     tone2wav
>
> /usr/bin/ld: warning: libssl.so.1.0.2, needed by
> //usr/lib/x86_64-linux-gnu/libcurl.so.4, may conflict with libssl.so.1.1
>
> /usr/bin/ld: warning: libcrypto.so.1.0.2, needed by
> //usr/lib/x86_64-linux-gnu/libcurl.so.4, may conflict with libcrypto.so.1.1
>
>   CC       src/fs_encode-fs_encode.o
>
>   CCLD     fs_encode
>
> /usr/bin/ld: warning: libssl.so.1.0.2, needed by
> //usr/lib/x86_64-linux-gnu/libcurl.so.4, may conflict with libssl.so.1.1
>
> /usr/bin/ld: warning: libcrypto.so.1.0.2, needed by
> //usr/lib/x86_64-linux-gnu/libcurl.so.4, may conflict with libcrypto.so.1.1
>
> make[2]: Leaving directory '/usr/src/freeswitch-1.8.5'
>
> Making all in src
>
> make[2]: Entering directory '/usr/src/freeswitch-1.8.5/src'
>
> Making all in mod
>
> make[3]: Entering directory '/usr/src/freeswitch-1.8.5/src/mod'
>
> Unknown target
>
> Makefile:721: recipe for target '-all' failed
>
> make[3]: *** [-all] Error 1
>
> make[3]: Leaving directory '/usr/src/freeswitch-1.8.5/src/mod'
>
> Makefile:591: recipe for target 'all-recursive' failed
>
> make[2]: *** [all-recursive] Error 1
>
> make[2]: Leaving directory '/usr/src/freeswitch-1.8.5/src'
>
> Makefile:3494: recipe for target 'all-recursive' failed
>
> make[1]: *** [all-recursive] Error 1
>
> make[1]: Leaving directory '/usr/src/freeswitch-1.8.5'
>
> Makefile:1255: recipe for target 'all' failed
>
> make: *** [all] Error 2
>
>
> Google did not helped me much on this, can somebody point me the right
> direction.
>
>
> Regards
>
> Abbasi
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From denis.papes at shishko.eu  Tue Feb 19 14:43:33 2019
From: denis.papes at shishko.eu (Denis Papes)
Date: Tue, 19 Feb 2019 14:43:33 +0000
Subject: [Freeswitch-users] SRTP and multiple endpoints
Message-ID: <DB6PR07MB3429D27D1D64EB635EA67B23857C0@DB6PR07MB3429.eurprd07.prod.outlook.com>

Hi,

I have FreeSWITCH server with two softphones and one Polycom VVX 310 phone. All endpoints use SRTP.
One softphone and Polycom phone are registered as same user. When I make a call to that user, softphone and Polycom both ring as expected.
Problem is that Polycom does early media, so it sends encryption key in 183 message and FreeSWITCH does not update key if I answer on softphone and call breaks after several "SRTP audio unprotect failed with code 7 (auth check failed) 83 bytes 10 errors" errors.

Does anyone know of any solution for that?



Call to Polycom
----------------------------------------------------


   ------------------------------------------------------------------------
recv 1268 bytes from udp/[XXX.XXX.XXX.XXX]:5060 at 13:23:24.192990:
   ------------------------------------------------------------------------
   SIP/2.0 183 Session Progress
   P-Asserted-Identity: "User Name" <sip:User.Name at domain.tld>,<tel:+11111111111;ext=1111>
   Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5062;received=XXX.XXX.XXX.XXX;rport=5062;branch=z9hG4bKH13aN0H9550Ue
   From: "Extension 1000" <sip:1000 at fs01.domain.tld>;tag=vvZQ9m8H03Npp
   To: "User Name" <sip:User.Name at domain.tld>;epid=0004f2853296;tag=BB7B7F4F-B75734BA
   CSeq: 737508 INVITE
   Call-ID: 5eaff2fd-aeec-1237-6aaa-00155d018038
   Contact: <sip:User.Name at domain.tld;opaque=user:epid:eji6PlZNF1Sy2o_kLXZ3sQAA;gruu>
   Record-Route: <sip:ucs.domain.tld:5061;transport=tls;ms-fe=server3.domain.tld;opaque=state:T;lr>, <sip:XXX.XXX.XXX.XXX:5061;transport=tls;r2=on;lr>, <sip:XXX.XXX.XXX.XXX;r2=on;lr>
   User-Agent: Polycom/5.9.0.9373 PolycomVVX-VVX_310-UA/5.9.0.9373
   Accept-Language: en
   Content-Type: application/sdp
   Content-Length: 354

   v=0
   o=- 1550582604 1550582604 IN IP4 YYY.YYY.YYY.YYY
   s=Polycom IP Phone
   c=IN IP4 YYY.YYY.YYY.YYY
   t=0 0
   a=sendrecv
   m=audio 5380 RTP/SAVP 102 101
   a=crypto:5 AES_CM_128_HMAC_SHA1_80 inline:GKS3Pyolq2FNrLsVljrZWO3ziKdaRG+9G8mXDn2F|2^31|1:1
   a=rtpmap:102 G7221/16000
   a=fmtp:102 bitrate=24000
   a=rtpmap:101 telephone-event/8000
   a=sendrecv
   a=rtcp:5381
   ------------------------------------------------------------------------
2019-02-19 13:23:23.731999 [INFO] sofia.c:1356 sofia/internal/User.Name at domain.tld Update Callee ID to "User Name" <User.Name>
2019-02-19 13:23:23.731999 [DEBUG] sofia.c:7291 Channel sofia/internal/User.Name at domain.tld entering state [proceeding][183]
2019-02-19 13:23:23.731999 [DEBUG] sofia.c:7301 Remote SDP:
v=0
o=- 1550582604 1550582604 IN IP4 YYY.YYY.YYY.YYY
s=Polycom IP Phone
c=IN IP4 YYY.YYY.YYY.YYY
t=0 0
a=sendrecv
m=audio 5380 RTP/SAVP 102 101
a=rtpmap:102 G7221/16000
a=fmtp:102 bitrate=24000
a=rtpmap:101 telephone-event/8000
a=crypto:5 AES_CM_128_HMAC_SHA1_80 inline:GKS3Pyolq2FNrLsVljrZWO3ziKdaRG+9G8mXDn2F|2^31|1:1
a=rtcp:5381

2019-02-19 13:23:23.731999 [NOTICE] sofia.c:7304 Pre-Answer sofia/internal/User.Name at domain.tld!
2019-02-19 13:23:23.731999 [DEBUG] switch_channel.c:3482 (sofia/internal/User.Name at domain.tld) Callstate Change RINGING -> EARLY
2019-02-19 13:23:23.731999 [DEBUG] switch_core_media.c:1804 looking for crypto suite [AEAD_AES_256_GCM_8]alias=[] in [5 AES_CM_128_HMAC_SHA1_80 inline:GKS3Pyolq2FNrLsVljrZWO3ziKdaRG+9G8mXDn2F|2^31|1:1]
2019-02-19 13:23:23.731999 [DEBUG] switch_core_media.c:1804 looking for crypto suite [AEAD_AES_128_GCM_8]alias=[] in [5 AES_CM_128_HMAC_SHA1_80 inline:GKS3Pyolq2FNrLsVljrZWO3ziKdaRG+9G8mXDn2F|2^31|1:1]
2019-02-19 13:23:23.731999 [DEBUG] switch_core_media.c:1804 looking for crypto suite [AES_256_CM_HMAC_SHA1_80]alias=[AES_CM_256_HMAC_SHA1_80] in [5 AES_CM_128_HMAC_SHA1_80 inline:GKS3Pyolq2FNrLsVljrZWO3ziKdaRG+9G8mXDn2F|2^31|1:1]
2019-02-19 13:23:23.731999 [DEBUG] switch_core_media.c:1804 looking for crypto suite [AES_192_CM_HMAC_SHA1_80]alias=[AES_CM_192_HMAC_SHA1_80] in [5 AES_CM_128_HMAC_SHA1_80 inline:GKS3Pyolq2FNrLsVljrZWO3ziKdaRG+9G8mXDn2F|2^31|1:1]
2019-02-19 13:23:23.731999 [DEBUG] switch_core_media.c:1804 looking for crypto suite [AES_CM_128_HMAC_SHA1_80]alias=[] in [5 AES_CM_128_HMAC_SHA1_80 inline:GKS3Pyolq2FNrLsVljrZWO3ziKdaRG+9G8mXDn2F|2^31|1:1]
2019-02-19 13:23:23.731999 [DEBUG] switch_core_media.c:1813 Found suite AES_CM_128_HMAC_SHA1_80
2019-02-19 13:23:23.731999 [DEBUG] switch_core_media.c:1883 Set Remote Key [5 AES_CM_128_HMAC_SHA1_80 inline:GKS3Pyolq2FNrLsVljrZWO3ziKdaRG+9G8mXDn2F|2^31|1:1]
2019-02-19 13:23:23.731999 [DEBUG] switch_core_media.c:5478 Audio Codec Compare [G7221:102:16000:20:24000:1]/[G7221:107:16000:20:24000:1]
2019-02-19 13:23:23.731999 [DEBUG] switch_core_media.c:5533 Audio Codec Compare [G7221:107:16000:20:24000:1] ++++ is saved as a match
2019-02-19 13:23:23.731999 [DEBUG] switch_core_media.c:5478 Audio Codec Compare [G7221:102:16000:20:24000:1]/[G722:9:8000:20:64000:1]
2019-02-19 13:23:23.731999 [DEBUG] switch_core_media.c:5478 Audio Codec Compare [G7221:102:16000:20:24000:1]/[PCMU:0:8000:20:64000:1]
2019-02-19 13:23:23.731999 [DEBUG] switch_core_media.c:5478 Audio Codec Compare [G7221:102:16000:20:24000:1]/[PCMA:8:8000:20:64000:1]
2019-02-19 13:23:23.731999 [DEBUG] switch_core_media.c:5394 Set telephone-event payload to 101 at 8000
2019-02-19 13:23:23.731999 [DEBUG] switch_core_media.c:3781 Set Codec sofia/internal/User.Name at domain.tld G7221/16000 20 ms 320 samples 24000 bits 1 channels
2019-02-19 13:23:23.731999 [DEBUG] switch_core_codec.c:111 sofia/internal/User.Name at domain.tld Original read codec set to G7221:107
2019-02-19 13:23:23.731999 [DEBUG] switch_core_media.c:5795 sofia/internal/User.Name at domain.tld Set 2833 dtmf send payload to 101 recv payload to 101
2019-02-19 13:23:23.731999 [DEBUG] switch_core_media.c:8511 AUDIO RTP [sofia/internal/User.Name at domain.tld] XXX.XXX.XXX.XXX port 26804 -> YYY.YYY.YYY.YYY port 5380 codec: 102 ms: 20
2019-02-19 13:23:23.731999 [DEBUG] switch_rtp.c:4300 Starting timer [soft] 320 bytes per 20ms
2019-02-19 13:23:23.731999 [DEBUG] switch_core_media.c:8731 Activating RTCP PORT 5381
2019-02-19 13:23:23.731999 [DEBUG] switch_rtp.c:4696 RTCP send rate is: 500 and packet rate is: 20000 Remote Port: 5381
2019-02-19 13:23:23.731999 [DEBUG] switch_rtp.c:2572 Setting RTCP remote addr to YYY.YYY.YYY.YYY:5381 2
2019-02-19 13:23:23.731999 [DEBUG] switch_core_media.c:8815 sofia/internal/User.Name at domain.tld Set 2833 dtmf send payload to 101
2019-02-19 13:23:23.731999 [DEBUG] switch_core_media.c:8822 sofia/internal/User.Name at domain.tld Set 2833 dtmf receive payload to 101
2019-02-19 13:23:23.731999 [DEBUG] switch_core_media.c:8845 sofia/internal/User.Name at domain.tld Set rtp dtmf delay to 40
2019-02-19 13:23:23.731999 [DEBUG] switch_core_media.c:1495 LIFETIME found in |2^31|1:1, base 2 exp 31
2019-02-19 13:23:23.731999 [DEBUG] switch_core_media.c:1502 MKI found in |2^31|1:1, id 1 size 1
2019-02-19 13:23:23.731999 [DEBUG] switch_core_media.c:1495 LIFETIME found in |2^31|1:1, base 2 exp 31
2019-02-19 13:23:23.731999 [DEBUG] switch_core_media.c:1502 MKI found in |2^31|1:1, id 1 size 1
2019-02-19 13:23:23.731999 [INFO] switch_rtp.c:4104 Activating audio Secure RTP SEND (with MKI)
2019-02-19 13:23:23.731999 [DEBUG] switch_core_sqldb.c:2617 Secure Type: srtp:sdes:AES_CM_128_HMAC_SHA1_80
2019-02-19 13:23:23.731999 [INFO] switch_rtp.c:4082 Activating audio Secure RTP RECV (with MKI)
2019-02-19 13:23:23.731999 [DEBUG] switch_core_sqldb.c:2617 Secure Type: srtp:sdes:AES_CM_128_HMAC_SHA1_80



Call to Softphone
----------------------------------------------------


recv 1860 bytes from udp/[XXX.XXX.XXX.XXX]:5060 at 13:23:25.884591:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5062;received=XXX.XXX.XXX.XXX;rport=5062;branch=z9hG4bKH13aN0H9550Ue
   Content-Length: 428
   P-Asserted-Identity: <sip:User.Name at domain.tld>, <tel:+11111111111;ext=1111>
   From: "Extension 1000" <sip:1000 at fs01.domain.tld>;tag=vvZQ9m8H03Npp
   To: <sip:User.Name at domain.tld>;epid=111ee812f0;tag=1f0cd7c868
   Call-ID: 5eaff2fd-aeec-1237-6aaa-00155d018038
   CSeq: 737508 INVITE
   Record-Route: <sip:ucs.domain.tld:5061;transport=tls;ms-fe=server3.domain.tld;opaque=state:T;lr>
   Record-Route: <sip:XXX.XXX.XXX.XXX:5061;transport=tls;r2=on;lr>
   Record-Route: <sip:XXX.XXX.XXX.XXX;r2=on;lr>
   Contact: <sip:User.Name at domain.tld;opaque=user:epid:Xq6LHtQWXFWBMxOY6SkY-wAA;gruu>
   User-Agent: UCCAPI/15.0.5093.1000 OC/15.0.5111.1000 (Skype for Business)
   Supported: histinfo
   Supported: ms-safe-transfer
   Supported: ms-dialog-route-set-update
   Allow: INVITE, BYE, ACK, CANCEL, INFO, UPDATE, REFER, NOTIFY, BENOTIFY, OPTIONS
   Session-Expires: 720;refresher=uac
   ms-endpoint-location-data: NetworkScope;ms-media-location-type=Intranet
   Supported: ms-bypass
   Supported: replaces
   Content-Type: application/sdp
   ms-application-via: ms-udc.cdr%3D70847bfc9ab1b8a46ead5d92c6ad493a%3A6%3Bconvhist%3D0%3A6;ms-pool=ucs.domain.tld;ms-application=http%3A%2F%2Fwww.microsoft.com%2FLCS%2FUdcAgent;ms-server=server3.domain.tld

   v=0
   o=- 0 1 IN IP4 ZZZ.ZZZ.ZZZ.ZZZ
   s=FreeSWITCH
   c=IN IP4 ZZZ.ZZZ.ZZZ.ZZZ
   b=CT:99980
   t=0 0
   m=audio 5407 RTP/SAVP 102 9 0 8 101
   a=crypto:5 AES_CM_128_HMAC_SHA1_80 inline:Pb70Q8VAa0lf1GNvKwjxRjUsK3pu1L28ga4pdn7s|2^31|1:1
   a=maxptime:200
   a=rtpmap:102 G7221/16000
   a=fmtp:102 bitrate=24000
   a=rtpmap:9 G722/8000
   a=rtpmap:0 PCMU/8000
   a=rtpmap:8 PCMA/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=rtcp-mux
   a=ptime:20
   ------------------------------------------------------------------------
2019-02-19 13:23:25.431998 [INFO] sofia.c:1356 sofia/internal/User.Name at domain.tld Update Callee ID to "User.Name" <User.Name>
2019-02-19 13:23:25.431998 [DEBUG] sofia.c:7291 Channel sofia/internal/User.Name at domain.tld entering state [completing][200]
2019-02-19 13:23:25.431998 [DEBUG] sofia.c:7301 Remote SDP:
v=0
o=- 0 1 IN IP4 ZZZ.ZZZ.ZZZ.ZZZ
s=FreeSWITCH
c=IN IP4 ZZZ.ZZZ.ZZZ.ZZZ
b=CT:99980
t=0 0
m=audio 5407 RTP/SAVP 102 9 0 8 101
a=rtpmap:102 G7221/16000
a=fmtp:102 bitrate=24000
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=crypto:5 AES_CM_128_HMAC_SHA1_80 inline:Pb70Q8VAa0lf1GNvKwjxRjUsK3pu1L28ga4pdn7s|2^31|1:1
a=maxptime:200
a=rtcp-mux
a=ptime:20



Thanks,

Denis
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From adrian.worutowicz at esifrance.net  Tue Feb 19 15:01:28 2019
From: adrian.worutowicz at esifrance.net (Adrian Worutowicz)
Date: Tue, 19 Feb 2019 15:01:28 +0000
Subject: [Freeswitch-users] How to disable FS DTMF detection
In-Reply-To: <54875FA9-519F-43E5-8E15-F0A6A39B72F8@vallimamod.org>
References: <96cd241a18314d39b455b3e96fb100e5@SRVEXCHANGE02.esifrance.net>
 <54875FA9-519F-43E5-8E15-F0A6A39B72F8@vallimamod.org>
Message-ID: <b2589f3f8b86493e8c98bbe78a70c0bc@SRVEXCHANGE02.esifrance.net>

Hi,

I've tried and failed.

When I press any key on the phone keyboard, the other end hears some residual DTMF sound and in the console I get:

2019-02-19 15:50:50.070836 [INFO] switch_channel.c:515 RECV DTMF 3:1760
2019-02-19 15:50:51.350964 [INFO] switch_channel.c:515 RECV DTMF 6:1920
2019-02-19 15:50:52.390068 [INFO] switch_channel.c:515 RECV DTMF 9:1760
2019-02-19 15:50:52.470076 [INFO] switch_channel.c:515 RECV DTMF 9:1112

I want the DTMF tones to pass freely from one phone to another with no interference.

Thanks,
Adrian.

-----Message d'origine-----
De : FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Vallimamod Abdullah
Envoyé : mardi 19 février 2019 12:39
À : FreeSWITCH Users Help
Objet : Re: [Freeswitch-users] How to disable FS DTMF detection

Hi,

You need to put the
  <param name="pass-rfc2833" value="true"/> in your sip profile like all other params.

It is not necessary however as you can just put
  <action application="export" data="pass_rfc2833=true"/> in your dialplan, notice the export instead of set.

The spandsp_stop_dtmf only works for inband dtmf, not the rfc2833 one.

Hope this helps.


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


> On 19 Feb 2019, at 10:40, Adrian Worutowicz <adrian.worutowicz at esifrance.net> wrote:
> 
> Hi,
>  
> I’ve developed a couple of endpoints and I use bgapi originate & bridge.
> Free Switch intercepts and interprets DTMF and I do not want that.
>  
> In the dialplan I tried:
>  
> <param name="pass-rfc2833" value="true"/>
> <action application="set" data="pass_rfc2833=true"/>
>  
> <action application="set" data="exec_after_bridge_app=stop_dtmf"/>
> <action application="stop_dtmf">
>  
> <action application="spandsp_stop_dtmf"/>
>  
> None of those methods works.
>  
> Maybe there is a parameter in originate that I should use?
>  
> *** How to tell Free Switch to leave alone the DTMF tones? ***
>  
> I’ll appreciate any help,
> Adrian.
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
> 
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com


_________________________________________________________________________
Professional FreeSWITCH Services
sales at freeswitch.com
https://freeswitch.com

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

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

From david.villasmil.work at gmail.com  Tue Feb 19 15:26:00 2019
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Tue, 19 Feb 2019 15:26:00 +0000
Subject: [Freeswitch-users] How to disable FS DTMF detection
In-Reply-To: <b2589f3f8b86493e8c98bbe78a70c0bc@SRVEXCHANGE02.esifrance.net>
References: <96cd241a18314d39b455b3e96fb100e5@SRVEXCHANGE02.esifrance.net>
 <54875FA9-519F-43E5-8E15-F0A6A39B72F8@vallimamod.org>
 <b2589f3f8b86493e8c98bbe78a70c0bc@SRVEXCHANGE02.esifrance.net>
Message-ID: <CAFGRPVqUAqEo_xhFPXwWFogzk-Nwy=DmG17X4oKGCE1V68V0PQ@mail.gmail.com>

Sorry to ask, but if your endpoint is calling your other endpoint, who is
generating the DTMF?
On Tue, 19 Feb 2019 at 15:18, Adrian Worutowicz <
adrian.worutowicz at esifrance.net> wrote:

> Hi,
>
> I've tried and failed.
>
> When I press any key on the phone keyboard, the other end hears some
> residual DTMF sound and in the console I get:
>
> 2019-02-19 15:50:50.070836 [INFO] switch_channel.c:515 RECV DTMF 3:1760
> 2019-02-19 15:50:51.350964 [INFO] switch_channel.c:515 RECV DTMF 6:1920
> 2019-02-19 15:50:52.390068 [INFO] switch_channel.c:515 RECV DTMF 9:1760
> 2019-02-19 15:50:52.470076 [INFO] switch_channel.c:515 RECV DTMF 9:1112
>
> I want the DTMF tones to pass freely from one phone to another with no
> interference.
>
> Thanks,
> Adrian.
>
> -----Message d'origine-----
> De : FreeSWITCH-users [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] De la part de Vallimamod
> Abdullah
> Envoyé : mardi 19 février 2019 12:39
> À : FreeSWITCH Users Help
> Objet : Re: [Freeswitch-users] How to disable FS DTMF detection
>
> Hi,
>
> You need to put the
>   <param name="pass-rfc2833" value="true"/> in your sip profile like all
> other params.
>
> It is not necessary however as you can just put
>   <action application="export" data="pass_rfc2833=true"/> in your
> dialplan, notice the export instead of set.
>
> The spandsp_stop_dtmf only works for inband dtmf, not the rfc2833 one.
>
> Hope this helps.
>
>
> Best Regards,
> --
> Vallimamod Abdullah
> SIP Solutions
> vma at sip.solutions
> linkedin.com/in/vallimamod
> .
>
>
> > On 19 Feb 2019, at 10:40, Adrian Worutowicz <
> adrian.worutowicz at esifrance.net> wrote:
> >
> > Hi,
> >
> > I’ve developed a couple of endpoints and I use bgapi originate & bridge.
> > Free Switch intercepts and interprets DTMF and I do not want that.
> >
> > In the dialplan I tried:
> >
> > <param name="pass-rfc2833" value="true"/>
> > <action application="set" data="pass_rfc2833=true"/>
> >
> > <action application="set" data="exec_after_bridge_app=stop_dtmf"/>
> > <action application="stop_dtmf">
> >
> > <action application="spandsp_stop_dtmf"/>
> >
> > None of those methods works.
> >
> > Maybe there is a parameter in originate that I should use?
> >
> > *** How to tell Free Switch to leave alone the DTMF tones? ***
> >
> > I’ll appreciate any help,
> > Adrian.
> > _________________________________________________________________________
> > Professional FreeSWITCH Services
> > sales at freeswitch.com
> > https://freeswitch.com
> >
> > Official FreeSWITCH Sites
> > https://freeswitch.com/oss
> > https://freeswitch.org/confluence
> > https://cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > https://freeswitch.com
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From david.villasmil.work at gmail.com  Tue Feb 19 15:27:07 2019
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Tue, 19 Feb 2019 15:27:07 +0000
Subject: [Freeswitch-users] [FreeSWITCH Install Issue]
In-Reply-To: <CAGJLwFhNka3iuVHAAmTy3bWibzpbxqzqyL+4_J858wgUN1gj2A@mail.gmail.com>
References: <CADSVerCzDkRK6mkj6D16Ois-ZynE_eTbiteH=Vm_oovvFBNcsQ@mail.gmail.com>
 <CAGJLwFhNka3iuVHAAmTy3bWibzpbxqzqyL+4_J858wgUN1gj2A@mail.gmail.com>
Message-ID: <CAFGRPVr49biC1HLY2yz6Sh-EhyhqG77te+RVZ65-enaeHpDuZQ@mail.gmail.com>

You should use Debian 8, not 9. At least for the time being.
On Tue, 19 Feb 2019 at 15:15, Alexey Sibyakin <alex at freeswitch.com> wrote:

> Hi,
>
> Try our awesome packages
> https://freeswitch.org/confluence/display/FREESWITCH/Debian+9+Stretch
>
> You should compile yourself only if you absolutely know what you are doing.
>
> It looks like you did *make -all* instead of *make *or *make all*.
>
> Regards,
>
> Alex
>
> Alex Sibyakin | Support Engineer SignalWire | 228 Hamilton Ave 3rd Floor,
> Palo Alto, CA 94303 Email: alex at freeswitch.com Website:
> https://www.signalwire.com
>
>
>
> On Tue, Feb 19, 2019 at 11:16 PM Bilal Abbasi <bilaln018 at gmail.com> wrote:
>
>> Hi Users,
>> I am having issues with installing FreeSWITCH 1.8.5 on Debian 9.6,
>> Here is the error i am getting this error when i run make command.
>>
>>  CC       src/libfreeswitch_la-switch_msrp.lo
>>
>>   CC       src/libfreeswitch_la-switch_vad.lo
>>
>>   CC       src/libfreeswitch_la-switch_vpx.lo
>>
>>   CXX      src/switch_cpp.lo
>>
>>   CXXLD    libfreeswitch.la
>>
>>
>> *** Warning: Linking the shared library libfreeswitch.la against the
>>
>> *** static library libs/libvpx/libvpx.a is not portable!
>>
>> ar: `u' modifier ignored since `D' is the default (see `U')
>>
>> Making all in .
>>
>> make[2]: Entering directory '/usr/src/freeswitch-1.8.5'
>>
>>   CC       src/freeswitch-switch.o
>>
>>   CCLD     freeswitch
>>
>> /usr/bin/ld: warning: libssl.so.1.0.2, needed by
>> //usr/lib/x86_64-linux-gnu/libcurl.so.4, may conflict with libssl.so.1.1
>>
>> /usr/bin/ld: warning: libcrypto.so.1.0.2, needed by
>> //usr/lib/x86_64-linux-gnu/libcurl.so.4, may conflict with libcrypto.so.1.1
>>
>>   CC       src/tone2wav-tone2wav.o
>>
>>   CCLD     tone2wav
>>
>> /usr/bin/ld: warning: libssl.so.1.0.2, needed by
>> //usr/lib/x86_64-linux-gnu/libcurl.so.4, may conflict with libssl.so.1.1
>>
>> /usr/bin/ld: warning: libcrypto.so.1.0.2, needed by
>> //usr/lib/x86_64-linux-gnu/libcurl.so.4, may conflict with libcrypto.so.1.1
>>
>>   CC       src/fs_encode-fs_encode.o
>>
>>   CCLD     fs_encode
>>
>> /usr/bin/ld: warning: libssl.so.1.0.2, needed by
>> //usr/lib/x86_64-linux-gnu/libcurl.so.4, may conflict with libssl.so.1.1
>>
>> /usr/bin/ld: warning: libcrypto.so.1.0.2, needed by
>> //usr/lib/x86_64-linux-gnu/libcurl.so.4, may conflict with libcrypto.so.1.1
>>
>> make[2]: Leaving directory '/usr/src/freeswitch-1.8.5'
>>
>> Making all in src
>>
>> make[2]: Entering directory '/usr/src/freeswitch-1.8.5/src'
>>
>> Making all in mod
>>
>> make[3]: Entering directory '/usr/src/freeswitch-1.8.5/src/mod'
>>
>> Unknown target
>>
>> Makefile:721: recipe for target '-all' failed
>>
>> make[3]: *** [-all] Error 1
>>
>> make[3]: Leaving directory '/usr/src/freeswitch-1.8.5/src/mod'
>>
>> Makefile:591: recipe for target 'all-recursive' failed
>>
>> make[2]: *** [all-recursive] Error 1
>>
>> make[2]: Leaving directory '/usr/src/freeswitch-1.8.5/src'
>>
>> Makefile:3494: recipe for target 'all-recursive' failed
>>
>> make[1]: *** [all-recursive] Error 1
>>
>> make[1]: Leaving directory '/usr/src/freeswitch-1.8.5'
>>
>> Makefile:1255: recipe for target 'all' failed
>>
>> make: *** [all] Error 2
>>
>>
>> Google did not helped me much on this, can somebody point me the right
>> direction.
>>
>>
>> Regards
>>
>> Abbasi
>> _________________________________________________________________________
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From adrian.worutowicz at esifrance.net  Tue Feb 19 15:51:21 2019
From: adrian.worutowicz at esifrance.net (Adrian Worutowicz)
Date: Tue, 19 Feb 2019 15:51:21 +0000
Subject: [Freeswitch-users] How to disable FS DTMF detection
In-Reply-To: <CAFGRPVqUAqEo_xhFPXwWFogzk-Nwy=DmG17X4oKGCE1V68V0PQ@mail.gmail.com>
References: <96cd241a18314d39b455b3e96fb100e5@SRVEXCHANGE02.esifrance.net>
 <54875FA9-519F-43E5-8E15-F0A6A39B72F8@vallimamod.org>
 <b2589f3f8b86493e8c98bbe78a70c0bc@SRVEXCHANGE02.esifrance.net>
 <CAFGRPVqUAqEo_xhFPXwWFogzk-Nwy=DmG17X4oKGCE1V68V0PQ@mail.gmail.com>
Message-ID: <d8dfe969616b4f7d810856fe5502467d@SRVEXCHANGE02.esifrance.net>

There are telephones on both sides.

Roughly: I get a call from ext. 9999:

[NOTICE] switch_channel.c:1104 New Channel sofia/external/9999 at 192.168.1.200 [2052b613-fab8-4508-8c2d-1c3e04c1c598]

I bridge it to ext. 6005:
EXECUTE sofia/external/6005 at 192.168.1.205 bridge(X1_T2_OUT/9999)


9999 talks to 6005. The DTMFs get intercepted and interpreted by FS (see the mail below).

Thank you for any tip on how to stop it.

De : FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de David Villasmil
Envoyé : mardi 19 février 2019 16:26
À : FreeSWITCH Users Help
Objet : Re: [Freeswitch-users] How to disable FS DTMF detection

Sorry to ask, but if your endpoint is calling your other endpoint, who is generating the DTMF?
On Tue, 19 Feb 2019 at 15:18, Adrian Worutowicz <adrian.worutowicz at esifrance.net<mailto:adrian.worutowicz at esifrance.net>> wrote:
Hi,

I've tried and failed.

When I press any key on the phone keyboard, the other end hears some residual DTMF sound and in the console I get:

2019-02-19 15:50:50.070836 [INFO] switch_channel.c:515 RECV DTMF 3:1760
2019-02-19 15:50:51.350964 [INFO] switch_channel.c:515 RECV DTMF 6:1920
2019-02-19 15:50:52.390068 [INFO] switch_channel.c:515 RECV DTMF 9:1760
2019-02-19 15:50:52.470076 [INFO] switch_channel.c:515 RECV DTMF 9:1112

I want the DTMF tones to pass freely from one phone to another with no interference.

Thanks,
Adrian.

-----Message d'origine-----
De : FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] De la part de Vallimamod Abdullah
Envoyé : mardi 19 février 2019 12:39
À : FreeSWITCH Users Help
Objet : Re: [Freeswitch-users] How to disable FS DTMF detection

Hi,

You need to put the
  <param name="pass-rfc2833" value="true"/> in your sip profile like all other params.

It is not necessary however as you can just put
  <action application="export" data="pass_rfc2833=true"/> in your dialplan, notice the export instead of set.

The spandsp_stop_dtmf only works for inband dtmf, not the rfc2833 one.

Hope this helps.


Best Regards,
--
Vallimamod Abdullah
SIP Solutions
vma at sip.solutions<mailto:vma at sip.solutions>
linkedin.com/in/vallimamod<http://linkedin.com/in/vallimamod>
.


> On 19 Feb 2019, at 10:40, Adrian Worutowicz <adrian.worutowicz at esifrance.net<mailto:adrian.worutowicz at esifrance.net>> wrote:
>
> Hi,
>
> I’ve developed a couple of endpoints and I use bgapi originate & bridge.
> Free Switch intercepts and interprets DTMF and I do not want that.
>
> In the dialplan I tried:
>
> <param name="pass-rfc2833" value="true"/>
> <action application="set" data="pass_rfc2833=true"/>
>
> <action application="set" data="exec_after_bridge_app=stop_dtmf"/>
> <action application="stop_dtmf">
>
> <action application="spandsp_stop_dtmf"/>
>
> None of those methods works.
>
> Maybe there is a parameter in originate that I should use?
>
> *** How to tell Free Switch to leave alone the DTMF tones? ***
>
> I’ll appreciate any help,
> Adrian.
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com<mailto:sales at freeswitch.com>
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com


_________________________________________________________________________
Professional FreeSWITCH Services
sales at freeswitch.com<mailto:sales at freeswitch.com>
https://freeswitch.com

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

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

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

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
https://freeswitch.com
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From kaiduanx at yahoo.ca  Tue Feb 19 15:51:55 2019
From: kaiduanx at yahoo.ca (kaiduan xie)
Date: Tue, 19 Feb 2019 15:51:55 +0000 (UTC)
Subject: Opus code change
References: <318977105.1740911.1550591515695.ref@mail.yahoo.com>
Message-ID: <318977105.1740911.1550591515695@mail.yahoo.com>

Hi,
Did Freeswitch team make any source code changes to upstream Opus Opus Codec? If yes, what are the changes?
Thanks,
/Kaiduan

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From bipin at xbipin.com  Tue Feb 19 16:24:20 2019
From: bipin at xbipin.com (Bipin Patel)
Date: Tue, 19 Feb 2019 20:24:20 +0400
Subject: VoIP encryption recommendations
Message-ID: <ea43776d-c02a-7c79-5a93-329b7d5a679f@xbipin.com>

hi,

i would like the ask the community about VoIP encryption, currently in 
few middle east countries VoIP is officially blocked. The isp are so 
aggressive that they use all sorts of fancy tools to block it including 
skype calls, whatsapp calls etc and are very successful in doing it. So 
far companies like voipswitch and recently few others have been 
providing tunneling mechanisms  to get over this but recently UDP 
traffic is heavily filtered and they go to the extreme of checking 
packet length and pattern and artificially introduce delay, jitter or 
simply block it if the number of hits are high. Switching to TLS/SRTP 
also doesnt help, it works with some isp but as soon as you try same 
using mobile data it stops working coz they match packet length and 
block based on the profile. ZRTP doesnt work coz a normal RTP streams 
needs to start and then it starts encrypting it but those initial RTP 
get blocked.

With lack of any more VoIP encryption protocols its almost getting 
impossible to bypass block so has anyone have any ideas of any other 
modern form of encryption which can be used for VoIP (btw VPN are also 
blocked and more over if packet size increases then nothing works on 
mobile data).

The market demand of skype replacements is also extremely high coz 
skype, hangouts, whatsapp video, instagram video, viber etc etc, u name 
it and its blocked.

-- 
Regards,
Bipin


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From s.safarov at gmail.com  Tue Feb 19 17:47:21 2019
From: s.safarov at gmail.com (Sergey Safarov)
Date: Tue, 19 Feb 2019 20:47:21 +0300
Subject: [Freeswitch-users] VoIP encryption recommendations
In-Reply-To: <mailman.222.1550593495.1400.freeswitch-users@lists.freeswitch.org>
References: <mailman.222.1550593495.1400.freeswitch-users@lists.freeswitch.org>
Message-ID: <CAHtxdDfDSpcb72OGFrBzmWRro8OsZki2DU8fU=KZFX98_dpBuw@mail.gmail.com>

In your case need
1) increase ptime value to 40;
2) use codec without compression;
3) need to enable VAD feature;

Also you can route torrent client network traffic via VPN together with
VoIP traffic. This will mask VoIP packets and not allow math your traffic
to VoIP profile on ISP equipment.


вт, 19 февр. 2019 г. в 19:53, Bipin Patel via FreeSWITCH-users <
freeswitch-users at lists.freeswitch.org>:

>
>
>
> ---------- Forwarded message ----------
> From: Bipin Patel <bipin at xbipin.com>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Cc:
> Bcc:
> Date: Tue, 19 Feb 2019 20:24:20 +0400
> Subject: VoIP encryption recommendations
> hi,
>
> i would like the ask the community about VoIP encryption, currently in few
> middle east countries VoIP is officially blocked. The isp are so aggressive
> that they use all sorts of fancy tools to block it including skype calls,
> whatsapp calls etc and are very successful in doing it. So far companies
> like voipswitch and recently few others have been providing tunneling
> mechanisms  to get over this but recently UDP traffic is heavily filtered
> and they go to the extreme of checking packet length and pattern and
> artificially introduce delay, jitter or simply block it if the number of
> hits are high. Switching to TLS/SRTP also doesnt help, it works with some
> isp but as soon as you try same using mobile data it stops working coz they
> match packet length and block based on the profile. ZRTP doesnt work coz a
> normal RTP streams needs to start and then it starts encrypting it but
> those initial RTP get blocked.
>
> With lack of any more VoIP encryption protocols its almost getting
> impossible to bypass block so has anyone have any ideas of any other modern
> form of encryption which can be used for VoIP (btw VPN are also blocked and
> more over if packet size increases then nothing works on mobile data).
>
> The market demand of skype replacements is also extremely high coz skype,
> hangouts, whatsapp video, instagram video, viber etc etc, u name it and its
> blocked.
>
> --
> Regards,
> Bipin
>
>
>
>
>
> ---------- Forwarded message ----------
> From: Bipin Patel via FreeSWITCH-users <
> freeswitch-users at lists.freeswitch.org>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Cc:
> Bcc:
> Date: Tue, 19 Feb 2019 08:53:29 -0800 (PST)
> Subject: [Freeswitch-users] VoIP encryption recommendations
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From bilaln018 at gmail.com  Tue Feb 19 17:55:55 2019
From: bilaln018 at gmail.com (Bilal Abbasi)
Date: Tue, 19 Feb 2019 22:55:55 +0500
Subject: [Freeswitch-users] [FreeSWITCH Install Issue]
In-Reply-To: <CAFGRPVr49biC1HLY2yz6Sh-EhyhqG77te+RVZ65-enaeHpDuZQ@mail.gmail.com>
References: <CADSVerCzDkRK6mkj6D16Ois-ZynE_eTbiteH=Vm_oovvFBNcsQ@mail.gmail.com>
 <CAGJLwFhNka3iuVHAAmTy3bWibzpbxqzqyL+4_J858wgUN1gj2A@mail.gmail.com>
 <CAFGRPVr49biC1HLY2yz6Sh-EhyhqG77te+RVZ65-enaeHpDuZQ@mail.gmail.com>
Message-ID: <CADSVerBZ0vVk6WN+bxP9RvSopsQyxdNFmm3G4_AtdY56VP5yCQ@mail.gmail.com>

I did “make” only, and got this.
I dont want to install via apt-get rather want to build via source.

Regards
Abbasi

On Tue, 19 Feb 2019 at 8:57 PM, David Villasmil <
david.villasmil.work at gmail.com> wrote:

> You should use Debian 8, not 9. At least for the time being.
> On Tue, 19 Feb 2019 at 15:15, Alexey Sibyakin <alex at freeswitch.com> wrote:
>
>> Hi,
>>
>> Try our awesome packages
>> https://freeswitch.org/confluence/display/FREESWITCH/Debian+9+Stretch
>>
>> You should compile yourself only if you absolutely know what you are
>> doing.
>>
>> It looks like you did *make -all* instead of *make *or *make all*.
>>
>> Regards,
>>
>> Alex
>>
>> Alex Sibyakin | Support Engineer SignalWire | 228 Hamilton Ave 3rd Floor,
>> Palo Alto, CA 94303 Email: alex at freeswitch.com Website:
>> https://www.signalwire.com
>>
>>
>>
>> On Tue, Feb 19, 2019 at 11:16 PM Bilal Abbasi <bilaln018 at gmail.com>
>> wrote:
>>
>>> Hi Users,
>>> I am having issues with installing FreeSWITCH 1.8.5 on Debian 9.6,
>>> Here is the error i am getting this error when i run make command.
>>>
>>>  CC       src/libfreeswitch_la-switch_msrp.lo
>>>
>>>   CC       src/libfreeswitch_la-switch_vad.lo
>>>
>>>   CC       src/libfreeswitch_la-switch_vpx.lo
>>>
>>>   CXX      src/switch_cpp.lo
>>>
>>>   CXXLD    libfreeswitch.la
>>>
>>>
>>> *** Warning: Linking the shared library libfreeswitch.la against the
>>>
>>> *** static library libs/libvpx/libvpx.a is not portable!
>>>
>>> ar: `u' modifier ignored since `D' is the default (see `U')
>>>
>>> Making all in .
>>>
>>> make[2]: Entering directory '/usr/src/freeswitch-1.8.5'
>>>
>>>   CC       src/freeswitch-switch.o
>>>
>>>   CCLD     freeswitch
>>>
>>> /usr/bin/ld: warning: libssl.so.1.0.2, needed by
>>> //usr/lib/x86_64-linux-gnu/libcurl.so.4, may conflict with libssl.so.1.1
>>>
>>> /usr/bin/ld: warning: libcrypto.so.1.0.2, needed by
>>> //usr/lib/x86_64-linux-gnu/libcurl.so.4, may conflict with libcrypto.so.1.1
>>>
>>>   CC       src/tone2wav-tone2wav.o
>>>
>>>   CCLD     tone2wav
>>>
>>> /usr/bin/ld: warning: libssl.so.1.0.2, needed by
>>> //usr/lib/x86_64-linux-gnu/libcurl.so.4, may conflict with libssl.so.1.1
>>>
>>> /usr/bin/ld: warning: libcrypto.so.1.0.2, needed by
>>> //usr/lib/x86_64-linux-gnu/libcurl.so.4, may conflict with libcrypto.so.1.1
>>>
>>>   CC       src/fs_encode-fs_encode.o
>>>
>>>   CCLD     fs_encode
>>>
>>> /usr/bin/ld: warning: libssl.so.1.0.2, needed by
>>> //usr/lib/x86_64-linux-gnu/libcurl.so.4, may conflict with libssl.so.1.1
>>>
>>> /usr/bin/ld: warning: libcrypto.so.1.0.2, needed by
>>> //usr/lib/x86_64-linux-gnu/libcurl.so.4, may conflict with libcrypto.so.1.1
>>>
>>> make[2]: Leaving directory '/usr/src/freeswitch-1.8.5'
>>>
>>> Making all in src
>>>
>>> make[2]: Entering directory '/usr/src/freeswitch-1.8.5/src'
>>>
>>> Making all in mod
>>>
>>> make[3]: Entering directory '/usr/src/freeswitch-1.8.5/src/mod'
>>>
>>> Unknown target
>>>
>>> Makefile:721: recipe for target '-all' failed
>>>
>>> make[3]: *** [-all] Error 1
>>>
>>> make[3]: Leaving directory '/usr/src/freeswitch-1.8.5/src/mod'
>>>
>>> Makefile:591: recipe for target 'all-recursive' failed
>>>
>>> make[2]: *** [all-recursive] Error 1
>>>
>>> make[2]: Leaving directory '/usr/src/freeswitch-1.8.5/src'
>>>
>>> Makefile:3494: recipe for target 'all-recursive' failed
>>>
>>> make[1]: *** [all-recursive] Error 1
>>>
>>> make[1]: Leaving directory '/usr/src/freeswitch-1.8.5'
>>>
>>> Makefile:1255: recipe for target 'all' failed
>>>
>>> make: *** [all] Error 2
>>>
>>>
>>> Google did not helped me much on this, can somebody point me the right
>>> direction.
>>>
>>>
>>> Regards
>>>
>>> Abbasi
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Services
>>> sales at freeswitch.com
>>> https://freeswitch.com
>>>
>>> Official FreeSWITCH Sites
>>> https://freeswitch.com/oss
>>> https://freeswitch.org/confluence
>>> https://cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> https://freeswitch.com
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From krice at freeswitch.org  Tue Feb 19 18:04:40 2019
From: krice at freeswitch.org (Ken Rice)
Date: Tue, 19 Feb 2019 12:04:40 -0600
Subject: [Freeswitch-users] where g729 fs-latest-installer script now ?
In-Reply-To: <CADihe3K-yyXFF8=w_7CLS3thpV9mbHnZWxbwR3TUOVy_k_nwsA@mail.gmail.com>
References: <CADihe3K9Cvhjtpdox-QXO-KUSHew-aey16zirECudPUQ6kNYpA@mail.gmail.com>
 <CADihe3+UAjHwHJyz2+Co1+oDnvu=79PhBNfVZM5jh+Vy3OOzNA@mail.gmail.com>
 <CADihe3KvzFs5X62R1EXjhY7k7+MEGjM63u7woyL-BXZaA-BZBw@mail.gmail.com>
 <CADihe3LQwqFQ9vLqU34X_taK0HEf2etp=3CO1S9YkQFCy6wSag@mail.gmail.com>
 <4af1376d-29ba-ea47-c952-c4883b7dd654@wirelessmundi.com>
 <CADihe3K-yyXFF8=w_7CLS3thpV9mbHnZWxbwR3TUOVy_k_nwsA@mail.gmail.com>
Message-ID: <4d7501d4c87d$978a56c0$c69f0440$@freeswitch.org>

My statement wasn’t to you, but to the people that continually want to pirate someone elses G729 code and refuse to support the projects developers. 

 

Its one of those things that grinds my gears.

 

K

 

From: FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> On Behalf Of Yuriy Nasida
Sent: Tuesday, February 19, 2019 6:47 AM
To: António Silva <asilva at wirelessmundi.com>
Cc: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] where g729 fs-latest-installer script now ?

 

Ken, not sure what you talking about. I already bought a lot of g729 licenses for my servers and I use them a lot of years. But now I can't use them normally at Deb 9 + fs1.8 + new g729 install script because of issues I described.

 

Thanks a lot Antonio! I will try this way.

 

On Tue, 19 Feb 2019 at 11:41, António Silva <asilva at wirelessmundi.com <mailto:asilva at wirelessmundi.com> > wrote:

try using the .deb and installed then manuall.:

Get the packages from here: http://files.freeswitch.org/repo/deb/freeswitch-1.8/pool/main/f/

Installed them manually:

dpkg -x freeswitch-mod-com-g729_1.8.5~7~14d59c97e0~jessie_amd64.deb / 
dpkg -x freeswitch-license-server_0.0.1~42~731842be7d~jessie_amd64.deb / 

 

I don't know if the installer will remain, from the jira https://freeswitch.org/jira/browse/FS-11641 i got the impression that now the solution is to use the debian packages.

 

On 13/02/2019 18:10, Yuriy Nasida wrote:

I have installed freeswitch-license-server  using fs-201902072050-installer  and got issue below. 

FS doesn't listen port 8021 and I can't connect to it and run any command using fs_cli.

All works fine in case I kill process freeswitch-license-server.

 

Can anybody comment please all these very weird things with commercial g729 at FS ?

Thanks.

 

On Wed, 13 Feb 2019 at 13:36, Yuriy Nasida <ynasida at gmail.com <mailto:ynasida at gmail.com> > wrote:

Well. Somebody updated http://files.freeswitch.org/g729/ and now it has fs-latest-installer. Thanks a lot whoever you are!!!

But... Now it has not install instructions  :)

 

On Wed, 13 Feb 2019 at 12:25, Yuriy Nasida <ynasida at gmail.com <mailto:ynasida at gmail.com> > wrote:

Um... Nobody uses g729 licences anymore ? :)

 

On Tue, 12 Feb 2019 at 20:07, Yuriy Nasida <ynasida at gmail.com <mailto:ynasida at gmail.com> > wrote:

Hi guys,

 

Does anybody know why site http://files.freeswitch.org/g729/ contain only install instrunction now ?  Where fs-latest-installer-v1.6 script or something like this ?

 

Please advice.

 

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

-- 
Saludos / Regards / Cumprimentos
António Silva

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From bipin at xbipin.com  Tue Feb 19 19:50:45 2019
From: bipin at xbipin.com (Bipin Patel)
Date: Tue, 19 Feb 2019 23:50:45 +0400
Subject: [Freeswitch-users] VoIP encryption recommendations
In-Reply-To: <CAHtxdDfDSpcb72OGFrBzmWRro8OsZki2DU8fU=KZFX98_dpBuw@mail.gmail.com>
References: <mailman.222.1550593495.1400.freeswitch-users@lists.freeswitch.org>
 <CAHtxdDfDSpcb72OGFrBzmWRro8OsZki2DU8fU=KZFX98_dpBuw@mail.gmail.com>
Message-ID: <169074f8208.279b.b07ebdf329620b8089087c7205b03f01@xbipin.com>

Hi,




The whole sip protocol is blocked and udp VPN don't connect and tcp ones 
they delay packets a lot so calls end up heavily choppy. On mobile data the 
restrictions are even heavier and if packets are anywhere close to VoIP or 
VoIP over VPN etc they get filtered. Etisalat is the isp which buys 
blocking equipment from some vendor in UK who specialize in blocking VoIP 
and VPN. Last I was told by some person working there was they use a lot of 
L7 packet inspectors.


Secondly it's not about setting up custom solutions for any company or 
client but we generate a lot of retail traffic so users need something that 
they can run on mobile etc like a customized dialer. Untill now I used to 
give them a openvpn profile which they used to run and then use Zoiper to 
place calls but all that is blocked now.


Webrtc seems to work as of now coz it's new but there isn't a webrtc based 
mobile dialer till now which anyone can install and just use it to place calls




On February 19, 2019 9:47:46 PM Sergey Safarov <s.safarov at gmail.com> wrote:
> In your case need
> 1) increase ptime value to 40;
> 2) use codec without compression;
> 3) need to enable VAD feature;
>
> Also you can route torrent client network traffic via VPN together with 
> VoIP traffic. This will mask VoIP packets and not allow math your traffic 
> to VoIP profile on ISP equipment.
>
> вт, 19 февр. 2019 г. в 19:53, Bipin Patel via FreeSWITCH-users 
> <freeswitch-users at lists.freeswitch.org>:
>
>
>
> ---------- Forwarded message ----------
> From: Bipin Patel <bipin at xbipin.com>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Cc:
> Bcc:
> Date: Tue, 19 Feb 2019 20:24:20 +0400
> Subject: VoIP encryption recommendations
>
> hi,
>
> i would like the ask the community about VoIP encryption, currently in few 
> middle east countries VoIP is officially blocked. The isp are so aggressive 
> that they use all sorts of fancy tools to block it including skype calls, 
> whatsapp calls etc and are very successful in doing it. So far companies 
> like voipswitch and recently few others have been providing tunneling 
> mechanisms  to get over this but recently UDP traffic is heavily filtered 
> and they go to the extreme of checking packet length and pattern and 
> artificially introduce delay, jitter or simply block it if the number of 
> hits are high. Switching to TLS/SRTP also doesnt help, it works with some 
> isp but as soon as you try same using mobile data it stops working coz they 
> match packet length and block based on the profile. ZRTP doesnt work coz a 
> normal RTP streams needs to start and then it starts encrypting it but 
> those initial RTP get blocked.
>
> With lack of any more VoIP encryption protocols its almost getting 
> impossible to bypass block so has anyone have any ideas of any other modern 
> form of encryption which can be used for VoIP (btw VPN are also blocked and 
> more over if packet size increases then nothing works on mobile data).
>
> The market demand of skype replacements is also extremely high coz skype, 
> hangouts, whatsapp video, instagram video, viber etc etc, u name it and its 
> blocked.
>
> --
> Regards,
> Bipin
>
>
>
>
>
> ---------- Forwarded message ----------
> From: Bipin Patel via FreeSWITCH-users <freeswitch-users at lists.freeswitch.org>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Cc:
> Bcc:
> Date: Tue, 19 Feb 2019 08:53:29 -0800 (PST)
> Subject: [Freeswitch-users] VoIP encryption recommendations
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com

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From bipin at xbipin.com  Tue Feb 19 19:58:21 2019
From: bipin at xbipin.com (Bipin Patel)
Date: Tue, 19 Feb 2019 23:58:21 +0400
Subject: [Freeswitch-users] VoIP encryption recommendations
In-Reply-To: <169074f8208.279b.b07ebdf329620b8089087c7205b03f01@xbipin.com>
References: <mailman.222.1550593495.1400.freeswitch-users@lists.freeswitch.org>
 <CAHtxdDfDSpcb72OGFrBzmWRro8OsZki2DU8fU=KZFX98_dpBuw@mail.gmail.com>
 <169074f8208.279b.b07ebdf329620b8089087c7205b03f01@xbipin.com>
Message-ID: <16907567748.279b.b07ebdf329620b8089087c7205b03f01@xbipin.com>


Btw we have tried almost all codecs and currently using g711u but it's just 
impossible to get through easily not to mention the isp even blacklists 
whole data center subnets if they find any data center to be a safe heaven 
for VoIP providers.

They even have this mechanism that home users if attempt to connect to any 
blocked service then their IP ends up on a blacklist and then the filtering 
gets worse for them until they reboot the router which gets them a new IP 
and things get back to normal

No wonder this region is considered a million dollar market where Microsoft 
and Facebook itself can't manage to keep Skype and WhatsApp calls even 
running no matter what.


On February 19, 2019 11:50:45 PM Bipin Patel <bipin at xbipin.com> wrote:
> Hi,
>
>
>
>
> The whole sip protocol is blocked and udp VPN don't connect and tcp ones 
> they delay packets a lot so calls end up heavily choppy. On mobile data the 
> restrictions are even heavier and if packets are anywhere close to VoIP or 
> VoIP over VPN etc they get filtered. Etisalat is the isp which buys 
> blocking equipment from some vendor in UK who specialize in blocking VoIP 
> and VPN. Last I was told by some person working there was they use a lot of 
> L7 packet inspectors.
>
>
> Secondly it's not about setting up custom solutions for any company or 
> client but we generate a lot of retail traffic so users need something that 
> they can run on mobile etc like a customized dialer. Untill now I used to 
> give them a openvpn profile which they used to run and then use Zoiper to 
> place calls but all that is blocked now.
>
>
> Webrtc seems to work as of now coz it's new but there isn't a webrtc based 
> mobile dialer till now which anyone can install and just use it to place calls
>
>
>
>
> On February 19, 2019 9:47:46 PM Sergey Safarov <s.safarov at gmail.com> wrote:
>> In your case need
>> 1) increase ptime value to 40;
>> 2) use codec without compression;
>> 3) need to enable VAD feature;
>>
>> Also you can route torrent client network traffic via VPN together with 
>> VoIP traffic. This will mask VoIP packets and not allow math your traffic 
>> to VoIP profile on ISP equipment.
>>
>>
>>
>>
>> вт, 19 февр. 2019 г. в 19:53, Bipin Patel via FreeSWITCH-users 
>> <freeswitch-users at lists.freeswitch.org>:
>>
>>
>>
>>
>> ---------- Forwarded message ----------
>> From: Bipin Patel <bipin at xbipin.com>
>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> Cc:
>> Bcc:
>> Date: Tue, 19 Feb 2019 20:24:20 +0400
>> Subject: VoIP encryption recommendations
>>
>> hi,
>>
>> i would like the ask the community about VoIP encryption, currently in few 
>> middle east countries VoIP is officially blocked. The isp are so aggressive 
>> that they use all sorts of fancy tools to block it including skype calls, 
>> whatsapp calls etc and are very successful in doing it. So far companies 
>> like voipswitch and recently few others have been providing tunneling 
>> mechanisms  to get over this but recently UDP traffic is heavily filtered 
>> and they go to the extreme of checking packet length and pattern and 
>> artificially introduce delay, jitter or simply block it if the number of 
>> hits are high. Switching to TLS/SRTP also doesnt help, it works with some 
>> isp but as soon as you try same using mobile data it stops working coz they 
>> match packet length and block based on the profile. ZRTP doesnt work coz a 
>> normal RTP streams needs to start and then it starts encrypting it but 
>> those initial RTP get blocked.
>>
>> With lack of any more VoIP encryption protocols its almost getting 
>> impossible to bypass block so has anyone have any ideas of any other modern 
>> form of encryption which can be used for VoIP (btw VPN are also blocked and 
>> more over if packet size increases then nothing works on mobile data).
>>
>> The market demand of skype replacements is also extremely high coz skype, 
>> hangouts, whatsapp video, instagram video, viber etc etc, u name it and its 
>> blocked.
>>
>>
>> --
>> Regards,
>> Bipin
>>
>>
>>
>>
>>
>>
>> ---------- Forwarded message ----------
>> From: Bipin Patel via FreeSWITCH-users <freeswitch-users at lists.freeswitch.org>
>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> Cc:
>> Bcc:
>> Date: Tue, 19 Feb 2019 08:53:29 -0800 (PST)
>> Subject: [Freeswitch-users] VoIP encryption recommendations
>> _________________________________________________________________________
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com

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From social at bohboh.info  Tue Feb 19 20:19:51 2019
From: social at bohboh.info (Social Boh)
Date: Tue, 19 Feb 2019 15:19:51 -0500
Subject: [Freeswitch-users] where g729 fs-latest-installer script now ?
In-Reply-To: <007e01d4c88c$c717cf30$55476d90$@freeswitch.org>
References: <CADihe3K9Cvhjtpdox-QXO-KUSHew-aey16zirECudPUQ6kNYpA@mail.gmail.com>
 <CADihe3+UAjHwHJyz2+Co1+oDnvu=79PhBNfVZM5jh+Vy3OOzNA@mail.gmail.com>
 <CADihe3KvzFs5X62R1EXjhY7k7+MEGjM63u7woyL-BXZaA-BZBw@mail.gmail.com>
 <CADihe3LQwqFQ9vLqU34X_taK0HEf2etp=3CO1S9YkQFCy6wSag@mail.gmail.com>
 <4af1376d-29ba-ea47-c952-c4883b7dd654@wirelessmundi.com>
 <CADihe3K-yyXFF8=w_7CLS3thpV9mbHnZWxbwR3TUOVy_k_nwsA@mail.gmail.com>
 <4d7501d4c87d$978a56c0$c69f0440$@freeswitch.org>
 <918f5e6d-bafb-3e1d-bd2d-26648ca948a7@bohboh.info>
 <007e01d4c88c$c717cf30$55476d90$@freeswitch.org>
Message-ID: <0f4adbf7-4df1-de06-3e3b-e8c7f653712e@bohboh.info>

So you don't update MPL 1.1 with MPL. 2.0 because the 1.1 is not 
compatible with GPL while 2.0 is compatible with GPL.

Regards

---
I'm SoCIaL, MayBe

El 19/02/2019 a las 14:53, Ken Rice escribió:
>
> I was pretty clear on the my statements about taking funding away from 
> the FreeSWITCH Project and the devs by doing such thing
>
> Also the bcg module is GPLv2 license violation waiting to be noticed… 
> you cant combine MPL and GPL code and have a resulting MPL codebase. 
> Its not allowed by the GPL. So you you are also ripping off the GPL 
> coder.
>
> Now you are polluting a MPL code base with the GPL violating the 
> Copyright holder of the MPL’s software rights to determine how his 
> code is licensed.
>
> This happens all the time. People think oh this code is free, and yes 
> most of the code is free, but the devs still have mortgages, children 
> to feed, computers and related hardware to keep updated so that they 
> can continue to support that hardware that everyone is using 
> currently. Website hosting for something like FreeSWITCH isnt free. It 
> requires over 1/2 a rack of colo’d equipment to keep the website, 
> mailing list, build servers, git repos, wiki etc etc etc going.
>
> $10 for a channel is a pretty cheap way to support that seeing as that 
> most of the time you don’t even need G729
>
> *From:*Social Boh <social at bohboh.info>
> *Sent:* Tuesday, February 19, 2019 1:03 PM
> *To:* Ken Rice <krice at freeswitch.org>
> *Subject:* Re: [Freeswitch-users] where g729 fs-latest-installer 
> script now ?
>
> Can you explain me why I can't use the mod_bcg729 with FreeSWITCH?
>
> Thank you
>
> ---
> I'm SoCIaL, MayBe
>
> El 19/02/2019 a las 13:04, Ken Rice escribió:
>
>     My statement wasn’t to you, but to the people that continually
>     want to pirate someone elses G729 code and refuse to support the
>     projects developers.
>
>     Its one of those things that grinds my gears.
>
>     K
>
>     *From:* FreeSWITCH-users
>     <freeswitch-users-bounces at lists.freeswitch.org>
>     <mailto:freeswitch-users-bounces at lists.freeswitch.org> *On Behalf
>     Of *Yuriy Nasida
>     *Sent:* Tuesday, February 19, 2019 6:47 AM
>     *To:* António Silva <asilva at wirelessmundi.com>
>     <mailto:asilva at wirelessmundi.com>
>     *Cc:* freeswitch-users at lists.freeswitch.org
>     <mailto:freeswitch-users at lists.freeswitch.org>
>     *Subject:* Re: [Freeswitch-users] where g729 fs-latest-installer
>     script now ?
>
>     Ken, not sure what you talking about. I already bought a lot of
>     g729 licenses for my servers and I use them a lot of years. But
>     now I can't use them normally at Deb 9 + fs1.8 + new g729 install
>     script because of issues I described.
>
>     Thanks a lot Antonio! I will try this way.
>
>     On Tue, 19 Feb 2019 at 11:41, António Silva
>     <asilva at wirelessmundi.com <mailto:asilva at wirelessmundi.com>> wrote:
>
>         try using the .deb and installed then manuall.:
>
>         Get the packages from here:
>         http://files.freeswitch.org/repo/deb/freeswitch-1.8/pool/main/f/
>
>         Installed them manually:
>
>         dpkg -x
>         freeswitch-mod-com-g729_1.8.5~7~14d59c97e0~jessie_amd64.deb /
>         dpkg -x
>         freeswitch-license-server_0.0.1~42~731842be7d~jessie_amd64.deb /
>
>         I don't know if the installer will remain, from the jira
>         https://freeswitch.org/jira/browse/FS-11641 i got the
>         impression that now the solution is to use the debian packages.
>
>         On 13/02/2019 18:10, Yuriy Nasida wrote:
>
>             I have installed freeswitch-license-server  using
>             fs-201902072050-installer  and got issue below.
>
>             FS doesn't listen port 8021 and I can't connect to it and
>             run any command using fs_cli.
>
>             All works fine in case I kill process
>             freeswitch-license-server.
>
>             Can anybody comment please all these very weird things
>             with commercial g729 at FS ?
>
>             Thanks.
>
>             On Wed, 13 Feb 2019 at 13:36, Yuriy Nasida
>             <ynasida at gmail.com <mailto:ynasida at gmail.com>> wrote:
>
>                 Well. Somebody updated
>                 http://files.freeswitch.org/g729/ and now it
>                 has fs-latest-installer. Thanks a lot whoever you are!!!
>
>                 But... Now it has not install instructions  :)
>
>                 On Wed, 13 Feb 2019 at 12:25, Yuriy Nasida
>                 <ynasida at gmail.com <mailto:ynasida at gmail.com>> wrote:
>
>                     Um... Nobody uses g729 licences anymore ? :)
>
>                     On Tue, 12 Feb 2019 at 20:07, Yuriy Nasida
>                     <ynasida at gmail.com <mailto:ynasida at gmail.com>> wrote:
>
>                         Hi guys,
>
>                         Does anybody know why site
>                         http://files.freeswitch.org/g729/ contain only
>                         install instrunction now ?
>                         Where fs-latest-installer-v1.6 script or
>                         something like this ?
>
>                         Please advice.
>
>             _________________________________________________________________________
>
>             Professional FreeSWITCH Services
>
>             sales at freeswitch.com  <mailto:sales at freeswitch.com>
>
>             https://freeswitch.com
>
>               
>
>             Official FreeSWITCH Sites
>
>             https://freeswitch.com/oss
>
>             https://freeswitch.org/confluence
>
>             https://cluecon.com
>
>               
>
>             FreeSWITCH-users mailing list
>
>             FreeSWITCH-users at lists.freeswitch.org  <mailto:FreeSWITCH-users at lists.freeswitch.org>
>
>             http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
>             UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>
>             https://freeswitch.com
>
>         -- 
>
>         Saludos / Regards / Cumprimentos
>
>         António Silva
>
>
>
>     _________________________________________________________________________
>
>     Professional FreeSWITCH Services
>
>     sales at freeswitch.com  <mailto:sales at freeswitch.com>
>
>     https://freeswitch.com
>
>     Official FreeSWITCH Sites
>
>     https://freeswitch.com/oss
>
>     https://freeswitch.org/confluence
>
>     https://cluecon.com
>
>     FreeSWITCH-users mailing list
>
>     FreeSWITCH-users at lists.freeswitch.org  <mailto:FreeSWITCH-users at lists.freeswitch.org>
>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>
>     https://freeswitch.com
>
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From dragos at freeswitch.org  Tue Feb 19 16:49:51 2019
From: dragos at freeswitch.org (Dragos Oancea)
Date: Tue, 19 Feb 2019 16:49:51 +0000
Subject: [Freeswitch-users] Opus code change
In-Reply-To: <mailman.217.1550591550.1400.freeswitch-users@lists.freeswitch.org>
References: <318977105.1740911.1550591515695.ref@mail.yahoo.com>
 <mailman.217.1550591550.1400.freeswitch-users@lists.freeswitch.org>
Message-ID: <CAGvJV4qbZJ-WOcGsxmVMxLsWY4f1SHtxgS4RrXQj341=U0hO2g@mail.gmail.com>

No, no changes in the library.

On Tue, Feb 19, 2019 at 4:29 PM kaiduan xie via FreeSWITCH-users <
freeswitch-users at lists.freeswitch.org> wrote:

>
>
>
> ---------- Forwarded message ----------
> From: kaiduan xie <kaiduanx at yahoo.ca>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Cc:
> Bcc:
> Date: Tue, 19 Feb 2019 15:51:55 +0000 (UTC)
> Subject: Opus code change
> Hi,
>
> Did Freeswitch team make any source code changes to upstream Opus Opus
> Codec <http://opus-codec.org/>? If yes, what are the changes?
>
> Thanks,
>
> /Kaiduan
>
>
>
>
>
> ---------- Forwarded message ----------
> From: kaiduan xie via FreeSWITCH-users <
> freeswitch-users at lists.freeswitch.org>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Cc:
> Bcc:
> Date: Tue, 19 Feb 2019 08:29:20 -0800 (PST)
> Subject: [Freeswitch-users] Opus code change
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From joel at textplus.com  Tue Feb 19 20:31:55 2019
From: joel at textplus.com (Joel Serrano)
Date: Tue, 19 Feb 2019 12:31:55 -0800
Subject: [Freeswitch-users] VoIP encryption recommendations
In-Reply-To: <mailman.232.1550606337.1400.freeswitch-users@lists.freeswitch.org>
References: <mailman.222.1550593495.1400.freeswitch-users@lists.freeswitch.org>
 <CAHtxdDfDSpcb72OGFrBzmWRro8OsZki2DU8fU=KZFX98_dpBuw@mail.gmail.com>
 <169074f8208.279b.b07ebdf329620b8089087c7205b03f01@xbipin.com>
 <mailman.232.1550606337.1400.freeswitch-users@lists.freeswitch.org>
Message-ID: <CAMtXxQnEpU7YXXWVybvKEie-KEFxA9UrEM2Be4D2=WFiZAe3NQ@mail.gmail.com>

Bipin,

Did you try the linphone tunnel? AFAIR they encrypt SIP+RTP on client
(requires linphone obviously), they also provide a server which receives
such connections and pass over the unencrypted SIP+RTP to the backend.

I think it's worth the try... some years ago it got around most blocks we
tested. We ended up not implementing it but the initial tests did look
good, don't know nowadays though...

On Tue, Feb 19, 2019 at 11:59 AM Bipin Patel via FreeSWITCH-users <
freeswitch-users at lists.freeswitch.org> wrote:

>
>
>
> ---------- Forwarded message ----------
> From: Bipin Patel <bipin at xbipin.com>
> To: <freeswitch-users at lists.freeswitch.org>
> Cc:
> Bcc:
> Date: Tue, 19 Feb 2019 23:58:21 +0400
> Subject: Re: [Freeswitch-users] VoIP encryption recommendations
>
> Btw we have tried almost all codecs and currently using g711u but it's
> just impossible to get through easily not to mention the isp even
> blacklists whole data center subnets if they find any data center to be a
> safe heaven for VoIP providers.
>
> They even have this mechanism that home users if attempt to connect to any
> blocked service then their IP ends up on a blacklist and then the filtering
> gets worse for them until they reboot the router which gets them a new IP
> and things get back to normal
>
> No wonder this region is considered a million dollar market where
> Microsoft and Facebook itself can't manage to keep Skype and WhatsApp calls
> even running no matter what.
>
>
> On February 19, 2019 11:50:45 PM Bipin Patel <bipin at xbipin.com> wrote:
>
>> Hi,
>>
>>
>> The whole sip protocol is blocked and udp VPN don't connect and tcp ones
>> they delay packets a lot so calls end up heavily choppy. On mobile data the
>> restrictions are even heavier and if packets are anywhere close to VoIP or
>> VoIP over VPN etc they get filtered. Etisalat is the isp which buys
>> blocking equipment from some vendor in UK who specialize in blocking VoIP
>> and VPN. Last I was told by some person working there was they use a lot of
>> L7 packet inspectors.
>>
>> Secondly it's not about setting up custom solutions for any company or
>> client but we generate a lot of retail traffic so users need something that
>> they can run on mobile etc like a customized dialer. Untill now I used to
>> give them a openvpn profile which they used to run and then use Zoiper to
>> place calls but all that is blocked now.
>>
>> Webrtc seems to work as of now coz it's new but there isn't a webrtc
>> based mobile dialer till now which anyone can install and just use it to
>> place calls
>>
>>
>>
>> On February 19, 2019 9:47:46 PM Sergey Safarov <s.safarov at gmail.com>
>> wrote:
>>
>>> In your case need
>>> 1) increase ptime value to 40;
>>> 2) use codec without compression;
>>> 3) need to enable VAD feature;
>>>
>>> Also you can route torrent client network traffic via VPN together with
>>> VoIP traffic. This will mask VoIP packets and not allow math your traffic
>>> to VoIP profile on ISP equipment.
>>>
>>>
>>> вт, 19 февр. 2019 г. в 19:53, Bipin Patel via FreeSWITCH-users <
>>> freeswitch-users at lists.freeswitch.org>:
>>>
>>>>
>>>>
>>>>
>>>> ---------- Forwarded message ----------
>>>> From: Bipin Patel <bipin at xbipin.com>
>>>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>>> Cc:
>>>> Bcc:
>>>> Date: Tue, 19 Feb 2019 20:24:20 +0400
>>>> Subject: VoIP encryption recommendations
>>>> hi,
>>>>
>>>> i would like the ask the community about VoIP encryption, currently in
>>>> few middle east countries VoIP is officially blocked. The isp are so
>>>> aggressive that they use all sorts of fancy tools to block it including
>>>> skype calls, whatsapp calls etc and are very successful in doing it. So far
>>>> companies like voipswitch and recently few others have been providing
>>>> tunneling mechanisms  to get over this but recently UDP traffic is heavily
>>>> filtered and they go to the extreme of checking packet length and pattern
>>>> and artificially introduce delay, jitter or simply block it if the number
>>>> of hits are high. Switching to TLS/SRTP also doesnt help, it works with
>>>> some isp but as soon as you try same using mobile data it stops working coz
>>>> they match packet length and block based on the profile. ZRTP doesnt work
>>>> coz a normal RTP streams needs to start and then it starts encrypting it
>>>> but those initial RTP get blocked.
>>>>
>>>> With lack of any more VoIP encryption protocols its almost getting
>>>> impossible to bypass block so has anyone have any ideas of any other modern
>>>> form of encryption which can be used for VoIP (btw VPN are also blocked and
>>>> more over if packet size increases then nothing works on mobile data).
>>>>
>>>> The market demand of skype replacements is also extremely high coz
>>>> skype, hangouts, whatsapp video, instagram video, viber etc etc, u name it
>>>> and its blocked.
>>>>
>>>> --
>>>> Regards,
>>>> Bipin
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> ---------- Forwarded message ----------
>>>> From: Bipin Patel via FreeSWITCH-users <
>>>> freeswitch-users at lists.freeswitch.org>
>>>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>>> Cc:
>>>> Bcc:
>>>> Date: Tue, 19 Feb 2019 08:53:29 -0800 (PST)
>>>> Subject: [Freeswitch-users] VoIP encryption recommendations
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Services
>>>> sales at freeswitch.com
>>>> https://freeswitch.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> https://freeswitch.com/oss
>>>> https://freeswitch.org/confluence
>>>> https://cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> https://freeswitch.com
>>>
>>>
>>
>
>
>
> ---------- Forwarded message ----------
> From: Bipin Patel via FreeSWITCH-users <
> freeswitch-users at lists.freeswitch.org>
> To: <freeswitch-users at lists.freeswitch.org>
> Cc:
> Bcc:
> Date: Tue, 19 Feb 2019 11:59:01 -0800 (PST)
> Subject: Re: [Freeswitch-users] VoIP encryption recommendations
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From joel at textplus.com  Tue Feb 19 20:33:55 2019
From: joel at textplus.com (Joel Serrano)
Date: Tue, 19 Feb 2019 12:33:55 -0800
Subject: [Freeswitch-users] VoIP encryption recommendations
In-Reply-To: <CAMtXxQnEpU7YXXWVybvKEie-KEFxA9UrEM2Be4D2=WFiZAe3NQ@mail.gmail.com>
References: <mailman.222.1550593495.1400.freeswitch-users@lists.freeswitch.org>
 <CAHtxdDfDSpcb72OGFrBzmWRro8OsZki2DU8fU=KZFX98_dpBuw@mail.gmail.com>
 <169074f8208.279b.b07ebdf329620b8089087c7205b03f01@xbipin.com>
 <mailman.232.1550606337.1400.freeswitch-users@lists.freeswitch.org>
 <CAMtXxQnEpU7YXXWVybvKEie-KEFxA9UrEM2Be4D2=WFiZAe3NQ@mail.gmail.com>
Message-ID: <CAMtXxQnqq=KVD6O85W1tGVAquDQiQ8JbKnbiOtdO6OcB0z=fNQ@mail.gmail.com>

Sorry, forgot to paste the link:

https://www.linphone.org/technical-corner/voip-tunnel

If you reach out to them, they have a server which you can use to test it
without having to deploy your own etc etc...

On Tue, Feb 19, 2019 at 12:31 PM Joel Serrano <joel at textplus.com> wrote:

> Bipin,
>
> Did you try the linphone tunnel? AFAIR they encrypt SIP+RTP on client
> (requires linphone obviously), they also provide a server which receives
> such connections and pass over the unencrypted SIP+RTP to the backend.
>
> I think it's worth the try... some years ago it got around most blocks we
> tested. We ended up not implementing it but the initial tests did look
> good, don't know nowadays though...
>
> On Tue, Feb 19, 2019 at 11:59 AM Bipin Patel via FreeSWITCH-users <
> freeswitch-users at lists.freeswitch.org> wrote:
>
>>
>>
>>
>> ---------- Forwarded message ----------
>> From: Bipin Patel <bipin at xbipin.com>
>> To: <freeswitch-users at lists.freeswitch.org>
>> Cc:
>> Bcc:
>> Date: Tue, 19 Feb 2019 23:58:21 +0400
>> Subject: Re: [Freeswitch-users] VoIP encryption recommendations
>>
>> Btw we have tried almost all codecs and currently using g711u but it's
>> just impossible to get through easily not to mention the isp even
>> blacklists whole data center subnets if they find any data center to be a
>> safe heaven for VoIP providers.
>>
>> They even have this mechanism that home users if attempt to connect to
>> any blocked service then their IP ends up on a blacklist and then the
>> filtering gets worse for them until they reboot the router which gets them
>> a new IP and things get back to normal
>>
>> No wonder this region is considered a million dollar market where
>> Microsoft and Facebook itself can't manage to keep Skype and WhatsApp calls
>> even running no matter what.
>>
>>
>> On February 19, 2019 11:50:45 PM Bipin Patel <bipin at xbipin.com> wrote:
>>
>>> Hi,
>>>
>>>
>>> The whole sip protocol is blocked and udp VPN don't connect and tcp ones
>>> they delay packets a lot so calls end up heavily choppy. On mobile data the
>>> restrictions are even heavier and if packets are anywhere close to VoIP or
>>> VoIP over VPN etc they get filtered. Etisalat is the isp which buys
>>> blocking equipment from some vendor in UK who specialize in blocking VoIP
>>> and VPN. Last I was told by some person working there was they use a lot of
>>> L7 packet inspectors.
>>>
>>> Secondly it's not about setting up custom solutions for any company or
>>> client but we generate a lot of retail traffic so users need something that
>>> they can run on mobile etc like a customized dialer. Untill now I used to
>>> give them a openvpn profile which they used to run and then use Zoiper to
>>> place calls but all that is blocked now.
>>>
>>> Webrtc seems to work as of now coz it's new but there isn't a webrtc
>>> based mobile dialer till now which anyone can install and just use it to
>>> place calls
>>>
>>>
>>>
>>> On February 19, 2019 9:47:46 PM Sergey Safarov <s.safarov at gmail.com>
>>> wrote:
>>>
>>>> In your case need
>>>> 1) increase ptime value to 40;
>>>> 2) use codec without compression;
>>>> 3) need to enable VAD feature;
>>>>
>>>> Also you can route torrent client network traffic via VPN together with
>>>> VoIP traffic. This will mask VoIP packets and not allow math your traffic
>>>> to VoIP profile on ISP equipment.
>>>>
>>>>
>>>> вт, 19 февр. 2019 г. в 19:53, Bipin Patel via FreeSWITCH-users <
>>>> freeswitch-users at lists.freeswitch.org>:
>>>>
>>>>>
>>>>>
>>>>>
>>>>> ---------- Forwarded message ----------
>>>>> From: Bipin Patel <bipin at xbipin.com>
>>>>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>>>> Cc:
>>>>> Bcc:
>>>>> Date: Tue, 19 Feb 2019 20:24:20 +0400
>>>>> Subject: VoIP encryption recommendations
>>>>> hi,
>>>>>
>>>>> i would like the ask the community about VoIP encryption, currently in
>>>>> few middle east countries VoIP is officially blocked. The isp are so
>>>>> aggressive that they use all sorts of fancy tools to block it including
>>>>> skype calls, whatsapp calls etc and are very successful in doing it. So far
>>>>> companies like voipswitch and recently few others have been providing
>>>>> tunneling mechanisms  to get over this but recently UDP traffic is heavily
>>>>> filtered and they go to the extreme of checking packet length and pattern
>>>>> and artificially introduce delay, jitter or simply block it if the number
>>>>> of hits are high. Switching to TLS/SRTP also doesnt help, it works with
>>>>> some isp but as soon as you try same using mobile data it stops working coz
>>>>> they match packet length and block based on the profile. ZRTP doesnt work
>>>>> coz a normal RTP streams needs to start and then it starts encrypting it
>>>>> but those initial RTP get blocked.
>>>>>
>>>>> With lack of any more VoIP encryption protocols its almost getting
>>>>> impossible to bypass block so has anyone have any ideas of any other modern
>>>>> form of encryption which can be used for VoIP (btw VPN are also blocked and
>>>>> more over if packet size increases then nothing works on mobile data).
>>>>>
>>>>> The market demand of skype replacements is also extremely high coz
>>>>> skype, hangouts, whatsapp video, instagram video, viber etc etc, u name it
>>>>> and its blocked.
>>>>>
>>>>> --
>>>>> Regards,
>>>>> Bipin
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> ---------- Forwarded message ----------
>>>>> From: Bipin Patel via FreeSWITCH-users <
>>>>> freeswitch-users at lists.freeswitch.org>
>>>>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>>>> Cc:
>>>>> Bcc:
>>>>> Date: Tue, 19 Feb 2019 08:53:29 -0800 (PST)
>>>>> Subject: [Freeswitch-users] VoIP encryption recommendations
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Services
>>>>> sales at freeswitch.com
>>>>> https://freeswitch.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> https://freeswitch.com/oss
>>>>> https://freeswitch.org/confluence
>>>>> https://cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> https://freeswitch.com
>>>>
>>>>
>>>
>>
>>
>>
>> ---------- Forwarded message ----------
>> From: Bipin Patel via FreeSWITCH-users <
>> freeswitch-users at lists.freeswitch.org>
>> To: <freeswitch-users at lists.freeswitch.org>
>> Cc:
>> Bcc:
>> Date: Tue, 19 Feb 2019 11:59:01 -0800 (PST)
>> Subject: Re: [Freeswitch-users] VoIP encryption recommendations
>> _________________________________________________________________________
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
>
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From ryan at comwavetech.com  Tue Feb 19 20:40:58 2019
From: ryan at comwavetech.com (Ryan Hallett)
Date: Tue, 19 Feb 2019 20:40:58 +0000
Subject: [Freeswitch-users] Unsubscribe
Message-ID: <e6add12f-35d1-49d6-88b5-a5bed9ead686@comwavetech.com>



Sent from Nine<http://www.9folders.com/>

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From ingjfnc at gmail.com  Tue Feb 19 21:08:49 2019
From: ingjfnc at gmail.com (Juan Nino)
Date: Wed, 20 Feb 2019 08:08:49 +1100
Subject: [Freeswitch-users] switch_ivr
Message-ID: <CAPr4L28nuPSNQq_fZART7b5w=kWm3Q8vffCx6dJ=yNx1xqqdyA@mail.gmail.com>

HI

chekcing the logs I have seen actions taken by switch_ivr.c
is there any way I can disable it? and/or check how does it work?
I have commented mod_ivr but this does not seem to affect
thanks
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From roman at dissauer.net  Tue Feb 19 22:24:08 2019
From: roman at dissauer.net (Roman Dissauer)
Date: Tue, 19 Feb 2019 23:24:08 +0100
Subject: [Freeswitch-users] setting contact domain in dialplan not working
Message-ID: <F363AB89-018F-4774-941D-74894521E932@dissauer.net>

Is there a way to get the FS public IP replaced by a custom domain name in contact header using the dialplan?

tried:

<action application="export" data="sip_contact_host=test.domain.tld“/>         ...not working
<action application=„set" data="sip_contact_host=test.domain.tld“/>         ...not working
<action application="export" data="sip_invite_domain=test.domain.tld“/>         ...not working
<action application=„set" data="sip_invite_domain=test.domain.tld“/>         ...not working
<action application="export" data=„sip_contact_user=bla at test.domain.tld"/>         ...not working
<action application=„set" data=„sip_invite_domain=bla at test.domain.tld“/>         ...not working


only thing working is setting the parameter in gateway config, 
<param name="contact-host" value=„test.domain.tld“/>

but this is not working for me because I need to do it dynamically in dialplan!

Any hint for me?

thank you!
Roman


From alex at freeswitch.com  Wed Feb 20 00:12:33 2019
From: alex at freeswitch.com (Alexey Sibyakin)
Date: Wed, 20 Feb 2019 09:12:33 +0900
Subject: [Freeswitch-users] [ERR] too many stacked extensions
In-Reply-To: <CABcV8PHtEAsHXKKKWnsas+u63Wy2FKYgQUWqa-Jys7mMmSpr1g@mail.gmail.com>
References: <CABcV8PHtEAsHXKKKWnsas+u63Wy2FKYgQUWqa-Jys7mMmSpr1g@mail.gmail.com>
Message-ID: <CAGJLwFikoeV2FKfwdF+HmCwQJdUZ_AfUscher+LWggxU+Xa8gg@mail.gmail.com>

Hi,

First of all, you need to update to 1.8.5. Second, try without Lua.

Regards,

Alex

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



On Tue, Feb 19, 2019 at 10:43 PM 梅表逵 <ceniy.mei at gmail.com> wrote:

> Hi,
>
>
> I’m running freeswitch version 1.8.1. In the log files, I saw a lot of
> [ERR] too many stacked extensions. It happens when freeswitch is trying to
> bridge  a callcenter member to agent.  I don’t know how to handle it. Could
> you please give any advises?
>
>
>
> 55791bf8-30bb-11e9-a4fd-7125b8df8c7a 2019-02-15 08:47:56.419673 [DEBUG]
> switch_ivr_bridge.c:1063 sofia/internal/137095 at 10.20.86.100:5060 CUSTOM
> HOLD
> 55791bf8-30bb-11e9-a4fd-7125b8df8c7a 2019-02-15 08:47:56.419673 [DEBUG]
> switch_core_state_machine.c:662 (sofia/internal/137095 at 10.20.86.100:5060)
> State CONSUME_MEDIA going to sleep
> 55791bf8-30bb-11e9-a4fd-7125b8df8c7a 2019-02-15 08:47:56.419673 [ERR]
> switch_ivr.c:902 Error sofia/internal/137095 at 10.20.86.100:5060 too many
> stacked extensions
> 55791bf8-30bb-11e9-a4fd-7125b8df8c7a 2019-02-15 08:47:56.419673 [ERR]
> switch_ivr.c:902 Error sofia/internal/137095 at 10.20.86.100:5060 too many
> stacked extensions
> 55791bf8-30bb-11e9-a4fd-7125b8df8c7a 2019-02-15 08:47:56.419673 [DEBUG]
> switch_ivr_bridge.c:1744 (sofia/internal/137095 at 10.20.86.100:5060) State
> Change CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA
> 55791bf8-30bb-11e9-a4fd-7125b8df8c7a 2019-02-15 08:47:56.419673 [ERR]
> switch_ivr.c:902 Error sofia/internal/137095 at 10.20.86.100:5060 too many
> stacked extensions
> 55791bf8-30bb-11e9-a4fd-7125b8df8c7a 2019-02-15 08:47:56.419673 [ERR]
> switch_ivr.c:902 Error sofia/internal/137095 at 10.20.86.100:5060 too many
> stacked extensions
> 55791bf8-30bb-11e9-a4fd-7125b8df8c7a 2019-02-15 08:47:56.419673 [DEBUG]
> switch_core_state_machine.c:584 (sofia/internal/137095 at 10.20.86.100:5060)
> Running State Change CS_EXCHANGE_MEDIA (Cur 2 Tot 4919)
> 55791bf8-30bb-11e9-a4fd-7125b8df8c7a 2019-02-15 08:47:56.419673 [DEBUG]
> switch_core_state_machine.c:653 (sofia/internal/137095 at 10.20.86.100:5060)
> State EXCHANGE_MEDIA
> 55791bf8-30bb-11e9-a4fd-7125b8df8c7a 2019-02-15 08:47:56.419673 [DEBUG]
> mod_sofia.c:645 SOFIA EXCHANGE_MEDIA
>
>
> XML:
> <condition field="destination_number" expression="^(31\d{4})$" >
>    <action application="set" data="dialed_extension=$1"/>
>    <!-- hangup after successful bridge to agent -->
>    <action application="set" data="recording_follow_transfer=true"/>
>    <action application="set" data="hangup_after_bridge=true"/>
>    <action application="set" data="say_agent_first=1" />
>    <action application="set" data="bridge_pre_execute_aleg_app=lua"/>
>    <action application="set"
> data="bridge_pre_execute_aleg_data=broadcastAgent.lua"/>
>   <action application="set" data="call_type=q" />
>   <action application="bind_meta_app" data="4 ab s
> execute_extension::att_xfer XML features"/>
>   <action application="callcenter" data="$1"/>
> </condition>
> </extension>
>
>
> Thanks
> Ceniy Mei
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From koketsom at tluka.co.za  Wed Feb 20 05:14:06 2019
From: koketsom at tluka.co.za (Koketso Mabuse)
Date: Wed, 20 Feb 2019 07:14:06 +0200
Subject: [Freeswitch-users] Connecting endpoint to domain within Free Switch
In-Reply-To: <mailman.229.1550605882.1400.freeswitch-users@lists.freeswitch.org>
References: <mailman.222.1550593495.1400.freeswitch-users@lists.freeswitch.org>
 <CAHtxdDfDSpcb72OGFrBzmWRro8OsZki2DU8fU=KZFX98_dpBuw@mail.gmail.com>
 <mailman.229.1550605882.1400.freeswitch-users@lists.freeswitch.org>
Message-ID: <333d641f-1b8a-4c69-5dcc-bf276e01640d@tluka.co.za>

Hi there

Scenario: Fresh installed FusionPBX:

  * Default domain with local IP,
  * created a 2nd domain as per tutorial,
  * created extensions.
      o tried to register extensions on 2nd domain failure
          + changed domain on extion to point to 1st main domain, end
            point registers.

Issue: how do I create 2 extensions  eg: 1000 for domain 1 and 1000 for 
domain 2 and register them both on main domain?

Domain:mtcc

exmple: Username: 1000

                Password: pass123

                host: 192.168.1.130


Domain: tluka

Username: 1000

                Password: pass123

                host: 192.168.1.130

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From gmaruzz at gmail.com  Wed Feb 20 09:25:25 2019
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Wed, 20 Feb 2019 10:25:25 +0100
Subject: [Freeswitch-users] VoIP encryption recommendations
In-Reply-To: <CAMtXxQnEpU7YXXWVybvKEie-KEFxA9UrEM2Be4D2=WFiZAe3NQ@mail.gmail.com>
References: <mailman.222.1550593495.1400.freeswitch-users@lists.freeswitch.org>
 <CAHtxdDfDSpcb72OGFrBzmWRro8OsZki2DU8fU=KZFX98_dpBuw@mail.gmail.com>
 <169074f8208.279b.b07ebdf329620b8089087c7205b03f01@xbipin.com>
 <mailman.232.1550606337.1400.freeswitch-users@lists.freeswitch.org>
 <CAMtXxQnEpU7YXXWVybvKEie-KEFxA9UrEM2Be4D2=WFiZAe3NQ@mail.gmail.com>
Message-ID: <CALXCt0orQx7AtDhSQtiyc4Q6c-Tq5fgOr+Ra8yxiE2-QaOfw5w@mail.gmail.com>

I would give a try webrtc with STUN/TURN on your server on 443, using
websocket on port 443 on same server that will serve https on same port.
You can use apache reverse websocket proxy, able to discriminate between
plain https and ssl websocket requests.

So, in this example, a total of two servers: one dedicated to
https/webrtc(sip or verto), one to stun/turn, both servers using ssl on
443. You can optionally add a third server for SIP TLS signaling, this too
on 443, with media going through the stun/turn server.

Maybe as codec you want to use a variable rate codec (check your opus
config, or another one) and no comforto noise/rtp waste, so you have a
(relatively) random traffic pattern, instead of a steady rtp flow.

-giovanni

On Tue, Feb 19, 2019 at 10:15 PM Joel Serrano <joel at textplus.com> wrote:

> Bipin,
>
> Did you try the linphone tunnel? AFAIR they encrypt SIP+RTP on client
> (requires linphone obviously), they also provide a server which receives
> such connections and pass over the unencrypted SIP+RTP to the backend.
>
> I think it's worth the try... some years ago it got around most blocks we
> tested. We ended up not implementing it but the initial tests did look
> good, don't know nowadays though...
>
> On Tue, Feb 19, 2019 at 11:59 AM Bipin Patel via FreeSWITCH-users <
> freeswitch-users at lists.freeswitch.org> wrote:
>
>>
>>
>>
>> ---------- Forwarded message ----------
>> From: Bipin Patel <bipin at xbipin.com>
>> To: <freeswitch-users at lists.freeswitch.org>
>> Cc:
>> Bcc:
>> Date: Tue, 19 Feb 2019 23:58:21 +0400
>> Subject: Re: [Freeswitch-users] VoIP encryption recommendations
>>
>> Btw we have tried almost all codecs and currently using g711u but it's
>> just impossible to get through easily not to mention the isp even
>> blacklists whole data center subnets if they find any data center to be a
>> safe heaven for VoIP providers.
>>
>> They even have this mechanism that home users if attempt to connect to
>> any blocked service then their IP ends up on a blacklist and then the
>> filtering gets worse for them until they reboot the router which gets them
>> a new IP and things get back to normal
>>
>> No wonder this region is considered a million dollar market where
>> Microsoft and Facebook itself can't manage to keep Skype and WhatsApp calls
>> even running no matter what.
>>
>>
>> On February 19, 2019 11:50:45 PM Bipin Patel <bipin at xbipin.com> wrote:
>>
>>> Hi,
>>>
>>>
>>> The whole sip protocol is blocked and udp VPN don't connect and tcp ones
>>> they delay packets a lot so calls end up heavily choppy. On mobile data the
>>> restrictions are even heavier and if packets are anywhere close to VoIP or
>>> VoIP over VPN etc they get filtered. Etisalat is the isp which buys
>>> blocking equipment from some vendor in UK who specialize in blocking VoIP
>>> and VPN. Last I was told by some person working there was they use a lot of
>>> L7 packet inspectors.
>>>
>>> Secondly it's not about setting up custom solutions for any company or
>>> client but we generate a lot of retail traffic so users need something that
>>> they can run on mobile etc like a customized dialer. Untill now I used to
>>> give them a openvpn profile which they used to run and then use Zoiper to
>>> place calls but all that is blocked now.
>>>
>>> Webrtc seems to work as of now coz it's new but there isn't a webrtc
>>> based mobile dialer till now which anyone can install and just use it to
>>> place calls
>>>
>>>
>>>
>>> On February 19, 2019 9:47:46 PM Sergey Safarov <s.safarov at gmail.com>
>>> wrote:
>>>
>>>> In your case need
>>>> 1) increase ptime value to 40;
>>>> 2) use codec without compression;
>>>> 3) need to enable VAD feature;
>>>>
>>>> Also you can route torrent client network traffic via VPN together with
>>>> VoIP traffic. This will mask VoIP packets and not allow math your traffic
>>>> to VoIP profile on ISP equipment.
>>>>
>>>>
>>>> вт, 19 февр. 2019 г. в 19:53, Bipin Patel via FreeSWITCH-users <
>>>> freeswitch-users at lists.freeswitch.org>:
>>>>
>>>>>
>>>>>
>>>>>
>>>>> ---------- Forwarded message ----------
>>>>> From: Bipin Patel <bipin at xbipin.com>
>>>>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>>>> Cc:
>>>>> Bcc:
>>>>> Date: Tue, 19 Feb 2019 20:24:20 +0400
>>>>> Subject: VoIP encryption recommendations
>>>>> hi,
>>>>>
>>>>> i would like the ask the community about VoIP encryption, currently in
>>>>> few middle east countries VoIP is officially blocked. The isp are so
>>>>> aggressive that they use all sorts of fancy tools to block it including
>>>>> skype calls, whatsapp calls etc and are very successful in doing it. So far
>>>>> companies like voipswitch and recently few others have been providing
>>>>> tunneling mechanisms  to get over this but recently UDP traffic is heavily
>>>>> filtered and they go to the extreme of checking packet length and pattern
>>>>> and artificially introduce delay, jitter or simply block it if the number
>>>>> of hits are high. Switching to TLS/SRTP also doesnt help, it works with
>>>>> some isp but as soon as you try same using mobile data it stops working coz
>>>>> they match packet length and block based on the profile. ZRTP doesnt work
>>>>> coz a normal RTP streams needs to start and then it starts encrypting it
>>>>> but those initial RTP get blocked.
>>>>>
>>>>> With lack of any more VoIP encryption protocols its almost getting
>>>>> impossible to bypass block so has anyone have any ideas of any other modern
>>>>> form of encryption which can be used for VoIP (btw VPN are also blocked and
>>>>> more over if packet size increases then nothing works on mobile data).
>>>>>
>>>>> The market demand of skype replacements is also extremely high coz
>>>>> skype, hangouts, whatsapp video, instagram video, viber etc etc, u name it
>>>>> and its blocked.
>>>>>
>>>>> --
>>>>> Regards,
>>>>> Bipin
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> ---------- Forwarded message ----------
>>>>> From: Bipin Patel via FreeSWITCH-users <
>>>>> freeswitch-users at lists.freeswitch.org>
>>>>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>>>> Cc:
>>>>> Bcc:
>>>>> Date: Tue, 19 Feb 2019 08:53:29 -0800 (PST)
>>>>> Subject: [Freeswitch-users] VoIP encryption recommendations
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Services
>>>>> sales at freeswitch.com
>>>>> https://freeswitch.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> https://freeswitch.com/oss
>>>>> https://freeswitch.org/confluence
>>>>> https://cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> https://freeswitch.com
>>>>
>>>>
>>>
>>
>>
>>
>> ---------- Forwarded message ----------
>> From: Bipin Patel via FreeSWITCH-users <
>> freeswitch-users at lists.freeswitch.org>
>> To: <freeswitch-users at lists.freeswitch.org>
>> Cc:
>> Bcc:
>> Date: Tue, 19 Feb 2019 11:59:01 -0800 (PST)
>> Subject: Re: [Freeswitch-users] VoIP encryption recommendations
>> _________________________________________________________________________
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com



-- 
Sincerely,

Giovanni Maruzzelli
OpenTelecom.IT
cell: +39 347 266 56 18
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From jurijs.ivolga at gmail.com  Wed Feb 20 11:29:58 2019
From: jurijs.ivolga at gmail.com (Jurijs Ivolga)
Date: Wed, 20 Feb 2019 13:29:58 +0200
Subject: [Freeswitch-users] 1.8 DTLS calls
Message-ID: <CAOrA2UZK=nNhSnm6CR8xieRbrsXHUoFVQnq7sMXT5OxCcm2n0g@mail.gmail.com>

Hi,

I'm trying to migrate from 1.6.20 to 1.8.5, but I'm facing a problem with
DTLS calls(media_webrtc=true).

Does anybody has same issues as me?

I have following setup:

RTP=>Freeswitch=>DTLS

In 1.8.5, when Freeswitch receive 200 from DTLS endpoint, but it never
sends it back to RTP endpoint.

Here some snippet from logs, where call stuck:

2019-02-20 09:50:07.368784 [INFO] switch_rtp.c:3832 Changing audio DTLS
state from OFF to HANDSHAKE
2019-02-20 09:50:07.368784 [DEBUG] switch_core_media.c:8815 sofia/internal/
9999999999999999999 at sip.myapp.net Set 2833 dtmf send payload to 101
2019-02-20 09:50:07.368784 [DEBUG] switch_core_media.c:8822 sofia/internal/
9999999999999999999 at sip.myapp.net Set 2833 dtmf receive payload to 101
2019-02-20 09:50:07.368784 [DEBUG] switch_core_media.c:8845 sofia/internal/
9999999999999999999 at sip.myapp.net Set rtp dtmf delay to 40

In 1.6.20 everything works as expected:

2019-02-20 11:12:40.129989 [DEBUG] switch_core_media.c:7180 sofia/internal/
9999999999999999999 at sip.myapp.net Set 2833 dtmf send payload to 101
2019-02-20 11:12:40.129989 [DEBUG] switch_core_media.c:7187 sofia/internal/
9999999999999999999 at sip.myapp.net Set 2833 dtmf receive payload to 101
2019-02-20 11:12:40.129989 [DEBUG] switch_core_media.c:7210 sofia/internal/
9999999999999999999 at sip.myapp.net Set rtp dtmf delay to 40
2019-02-20 11:12:40.129989 [NOTICE] sofia.c:8218 Channel [sofia/internal/
9999999999999999999 at sip.myapp.net] has been answered
2019-02-20 11:12:40.129989 [DEBUG] switch_channel.c:3773 (sofia/internal/
9999999999999999999 at sip.myapp.net) Callstate Change RINGING -> ACTIVE
2019-02-20 11:12:40.129989 [DEBUG] switch_core_codec.c:248 sofia/internal/
1234567890 at sip.myapp.net Restore previous codec PCMA:8.
2019-02-20 11:12:40.129989 [DEBUG] switch_core_media.c:6861 Audio params
are unchanged for sofia/internal/1234567890 at sip.myapp.net.
2019-02-20 11:12:40.129989 [DEBUG] mod_sofia.c:850 Local SDP sofia/internal/
1234567890 at sip.myapp.net:

Jurijs
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From roman at dissauer.net  Wed Feb 20 11:54:30 2019
From: roman at dissauer.net (Roman Dissauer)
Date: Wed, 20 Feb 2019 12:54:30 +0100
Subject: [Freeswitch-users] setting contact domain in dialplan not
	working
In-Reply-To: <F363AB89-018F-4774-941D-74894521E932@dissauer.net>
References: <F363AB89-018F-4774-941D-74894521E932@dissauer.net>
Message-ID: <2A1FAD07-DB26-4E69-AF70-1B9FC4CAA7D5@dissauer.net>

This is my FreeSWITCH Version:

FreeSWITCH version: 1.8.5-6-31281a0bf1~64bit (-6-31281a0bf1 64bit)

same thing with version 1.6.20

> Am 19.02.2019 um 23:24 schrieb Roman Dissauer <roman at dissauer.net>:
> 
> Is there a way to get the FS public IP replaced by a custom domain name in contact header using the dialplan?
> 
> tried:
> 
> <action application="export" data="sip_contact_host=test.domain.tld“/>         ...not working
> <action application=„set" data="sip_contact_host=test.domain.tld“/>         ...not working
> <action application="export" data="sip_invite_domain=test.domain.tld“/>         ...not working
> <action application=„set" data="sip_invite_domain=test.domain.tld“/>         ...not working
> <action application="export" data=„sip_contact_user=bla at test.domain.tld"/>         ...not working
> <action application=„set" data=„sip_invite_domain=bla at test.domain.tld“/>         ...not working
> 
> 
> only thing working is setting the parameter in gateway config, 
> <param name="contact-host" value=„test.domain.tld“/>
> 
> but this is not working for me because I need to do it dynamically in dialplan!
> 
> Any hint for me?
> 
> thank you!
> Roman
> 
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
> 
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com




From bipin at xbipin.com  Wed Feb 20 12:46:19 2019
From: bipin at xbipin.com (Bipin Patel)
Date: Wed, 20 Feb 2019 16:46:19 +0400
Subject: [Freeswitch-users] VoIP encryption recommendations
In-Reply-To: <CALXCt0orQx7AtDhSQtiyc4Q6c-Tq5fgOr+Ra8yxiE2-QaOfw5w@mail.gmail.com>
References: <mailman.222.1550593495.1400.freeswitch-users@lists.freeswitch.org>
 <CAHtxdDfDSpcb72OGFrBzmWRro8OsZki2DU8fU=KZFX98_dpBuw@mail.gmail.com>
 <169074f8208.279b.b07ebdf329620b8089087c7205b03f01@xbipin.com>
 <mailman.232.1550606337.1400.freeswitch-users@lists.freeswitch.org>
 <CAMtXxQnEpU7YXXWVybvKEie-KEFxA9UrEM2Be4D2=WFiZAe3NQ@mail.gmail.com>
 <CALXCt0orQx7AtDhSQtiyc4Q6c-Tq5fgOr+Ra8yxiE2-QaOfw5w@mail.gmail.com>
Message-ID: <1690af14978.279b.b07ebdf329620b8089087c7205b03f01@xbipin.com>

Well webrtc currently works coz it's fairly new but how would the end user 
place calls? Preferred way is using some app rather than through browser 
and the main aim is to get the port opened on ISP side coz if there is 
traffic flow and port gets opened then the filtering would stop until the 
flow continues and in that duration I'm also able to send normal sip and 
rtp even but if the flow stops for like 5 seconds the filtering begins again.


Some tunnels used this method in the past, they send normal http or https 
traffic both ways and filtering stops and then they send whatever packets 
they want on same port and things start flowing but the isp over time 
figures this out and manages to block it


Earlier within the country VoIP was allowed but since WhatsApp calling and 
Google duo started they apply same filter locally even. The other thing I 
noticed is the closer your are to a ISP Telecom exchange the harder like 
becomes to bypass the block and the farther you are then routers in between 
tend to add their overhead and the packet pattern changes a bit which makes 
life easier to bypass it


On February 20, 2019 1:26:25 PM Giovanni Maruzzelli <gmaruzz at gmail.com> wrote:
> I would give a try webrtc with STUN/TURN on your server on 443, using 
> websocket on port 443 on same server that will serve https on same port. 
> You can use apache reverse websocket proxy, able to discriminate between 
> plain https and ssl websocket requests.
>
> So, in this example, a total of two servers: one dedicated to 
> https/webrtc(sip or verto), one to stun/turn, both servers using ssl on 
> 443. You can optionally add a third server for SIP TLS signaling, this too 
> on 443, with media going through the stun/turn server.
>
> Maybe as codec you want to use a variable rate codec (check your opus 
> config, or another one) and no comforto noise/rtp waste, so you have a 
> (relatively) random traffic pattern, instead of a steady rtp flow.
>
> -giovanni
>
>
> On Tue, Feb 19, 2019 at 10:15 PM Joel Serrano <joel at textplus.com> wrote:
> Bipin,
>
> Did you try the linphone tunnel? AFAIR they encrypt SIP+RTP on client 
> (requires linphone obviously), they also provide a server which receives 
> such connections and pass over the unencrypted SIP+RTP to the backend.
>
> I think it's worth the try... some years ago it got around most blocks we 
> tested. We ended up not implementing it but the initial tests did look 
> good, don't know nowadays though...
>
>
> On Tue, Feb 19, 2019 at 11:59 AM Bipin Patel via FreeSWITCH-users 
> <freeswitch-users at lists.freeswitch.org> wrote:
>
>
>
> ---------- Forwarded message ----------
> From: Bipin Patel <bipin at xbipin.com>
> To: <freeswitch-users at lists.freeswitch.org>
> Cc:
> Bcc:
> Date: Tue, 19 Feb 2019 23:58:21 +0400
> Subject: Re: [Freeswitch-users] VoIP encryption recommendations
>
>
> Btw we have tried almost all codecs and currently using g711u but it's just 
> impossible to get through easily not to mention the isp even blacklists 
> whole data center subnets if they find any data center to be a safe heaven 
> for VoIP providers.
>
> They even have this mechanism that home users if attempt to connect to any 
> blocked service then their IP ends up on a blacklist and then the filtering 
> gets worse for them until they reboot the router which gets them a new IP 
> and things get back to normal
>
> No wonder this region is considered a million dollar market where Microsoft 
> and Facebook itself can't manage to keep Skype and WhatsApp calls even 
> running no matter what.
>
>
> On February 19, 2019 11:50:45 PM Bipin Patel <bipin at xbipin.com> wrote:
>> Hi,
>>
>>
>> The whole sip protocol is blocked and udp VPN don't connect and tcp ones 
>> they delay packets a lot so calls end up heavily choppy. On mobile data the 
>> restrictions are even heavier and if packets are anywhere close to VoIP or 
>> VoIP over VPN etc they get filtered. Etisalat is the isp which buys 
>> blocking equipment from some vendor in UK who specialize in blocking VoIP 
>> and VPN. Last I was told by some person working there was they use a lot of 
>> L7 packet inspectors.
>>
>> Secondly it's not about setting up custom solutions for any company or 
>> client but we generate a lot of retail traffic so users need something that 
>> they can run on mobile etc like a customized dialer. Untill now I used to 
>> give them a openvpn profile which they used to run and then use Zoiper to 
>> place calls but all that is blocked now.
>>
>> Webrtc seems to work as of now coz it's new but there isn't a webrtc based 
>> mobile dialer till now which anyone can install and just use it to place calls
>>
>>
>>
>> On February 19, 2019 9:47:46 PM Sergey Safarov <s.safarov at gmail.com> wrote:
>>> In your case need
>>> 1) increase ptime value to 40;
>>> 2) use codec without compression;
>>> 3) need to enable VAD feature;
>>>
>>> Also you can route torrent client network traffic via VPN together with 
>>> VoIP traffic. This will mask VoIP packets and not allow math your traffic 
>>> to VoIP profile on ISP equipment.
>>>
>>> вт, 19 февр. 2019 г. в 19:53, Bipin Patel via FreeSWITCH-users 
>>> <freeswitch-users at lists.freeswitch.org>:
>>>
>>>
>>>
>>> ---------- Forwarded message ----------
>>> From: Bipin Patel <bipin at xbipin.com>
>>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>> Cc:
>>> Bcc:
>>> Date: Tue, 19 Feb 2019 20:24:20 +0400
>>> Subject: VoIP encryption recommendations
>>>
>>> hi,
>>>
>>> i would like the ask the community about VoIP encryption, currently in few 
>>> middle east countries VoIP is officially blocked. The isp are so aggressive 
>>> that they use all sorts of fancy tools to block it including skype calls, 
>>> whatsapp calls etc and are very successful in doing it. So far companies 
>>> like voipswitch and recently few others have been providing tunneling 
>>> mechanisms  to get over this but recently UDP traffic is heavily filtered 
>>> and they go to the extreme of checking packet length and pattern and 
>>> artificially introduce delay, jitter or simply block it if the number of 
>>> hits are high. Switching to TLS/SRTP also doesnt help, it works with some 
>>> isp but as soon as you try same using mobile data it stops working coz they 
>>> match packet length and block based on the profile. ZRTP doesnt work coz a 
>>> normal RTP streams needs to start and then it starts encrypting it but 
>>> those initial RTP get blocked.
>>>
>>> With lack of any more VoIP encryption protocols its almost getting 
>>> impossible to bypass block so has anyone have any ideas of any other modern 
>>> form of encryption which can be used for VoIP (btw VPN are also blocked and 
>>> more over if packet size increases then nothing works on mobile data).
>>>
>>> The market demand of skype replacements is also extremely high coz skype, 
>>> hangouts, whatsapp video, instagram video, viber etc etc, u name it and its 
>>> blocked.
>>>
>>> --
>>> Regards,
>>> Bipin
>>>
>>>
>>>
>>>
>>>
>>> ---------- Forwarded message ----------
>>> From: Bipin Patel via FreeSWITCH-users <freeswitch-users at lists.freeswitch.org>
>>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>> Cc:
>>> Bcc:
>>> Date: Tue, 19 Feb 2019 08:53:29 -0800 (PST)
>>> Subject: [Freeswitch-users] VoIP encryption recommendations
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Services
>>> sales at freeswitch.com
>>> https://freeswitch.com
>>>
>>> Official FreeSWITCH Sites
>>> https://freeswitch.com/oss
>>> https://freeswitch.org/confluence
>>> https://cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> https://freeswitch.com
>
>
>
>
> ---------- Forwarded message ----------
> From: Bipin Patel via FreeSWITCH-users <freeswitch-users at lists.freeswitch.org>
> To: <freeswitch-users at lists.freeswitch.org>
> Cc:
> Bcc:
> Date: Tue, 19 Feb 2019 11:59:01 -0800 (PST)
> Subject: Re: [Freeswitch-users] VoIP encryption recommendations
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
>
>
> --
>
> Sincerely,
>
> Giovanni Maruzzelli
> OpenTelecom.IT
> cell: +39 347 266 56 18

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From gmaruzz at gmail.com  Wed Feb 20 12:46:16 2019
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Wed, 20 Feb 2019 13:46:16 +0100
Subject: [Freeswitch-users] 1.8 DTLS calls
In-Reply-To: <CAOrA2UZK=nNhSnm6CR8xieRbrsXHUoFVQnq7sMXT5OxCcm2n0g@mail.gmail.com>
References: <CAOrA2UZK=nNhSnm6CR8xieRbrsXHUoFVQnq7sMXT5OxCcm2n0g@mail.gmail.com>
Message-ID: <CALXCt0rL_D4jZq0HCCCssg1vf4QdfGe7U+RxBVBwe3Xc7Kd1Sw@mail.gmail.com>

On Wed, Feb 20, 2019 at 12:59 PM Jurijs Ivolga <jurijs.ivolga at gmail.com>
wrote:

> Hi,
>
> I'm trying to migrate from 1.6.20 to 1.8.5, but I'm facing a problem with
> DTLS calls(media_webrtc=true).
>
> Does anybody has same issues as me?
>
> I have following setup:
>
> RTP=>Freeswitch=>DTLS
>
> In 1.8.5, when Freeswitch receive 200 from DTLS endpoint, but it never
> sends it back to RTP endpoint.
>

maybe you have a network problem, or a configuration problem network
related in 1.8 deployment?

Also, for us to better understand: from fs cli, "sofia global siptrace on",
after this, copy and paste the ENTIRE debug level output from beginning to
end in a pastebin.

Then put here the pastebin link

-giovanni



>
> Here some snippet from logs, where call stuck:
>
> 2019-02-20 09:50:07.368784 [INFO] switch_rtp.c:3832 Changing audio DTLS
> state from OFF to HANDSHAKE
> 2019-02-20 09:50:07.368784 [DEBUG] switch_core_media.c:8815 sofia/internal/
> 9999999999999999999 at sip.myapp.net Set 2833 dtmf send payload to 101
> 2019-02-20 09:50:07.368784 [DEBUG] switch_core_media.c:8822 sofia/internal/
> 9999999999999999999 at sip.myapp.net Set 2833 dtmf receive payload to 101
> 2019-02-20 09:50:07.368784 [DEBUG] switch_core_media.c:8845 sofia/internal/
> 9999999999999999999 at sip.myapp.net Set rtp dtmf delay to 40
>
> In 1.6.20 everything works as expected:
>
> 2019-02-20 11:12:40.129989 [DEBUG] switch_core_media.c:7180 sofia/internal/
> 9999999999999999999 at sip.myapp.net Set 2833 dtmf send payload to 101
> 2019-02-20 11:12:40.129989 [DEBUG] switch_core_media.c:7187 sofia/internal/
> 9999999999999999999 at sip.myapp.net Set 2833 dtmf receive payload to 101
> 2019-02-20 11:12:40.129989 [DEBUG] switch_core_media.c:7210 sofia/internal/
> 9999999999999999999 at sip.myapp.net Set rtp dtmf delay to 40
> 2019-02-20 11:12:40.129989 [NOTICE] sofia.c:8218 Channel [sofia/internal/
> 9999999999999999999 at sip.myapp.net] has been answered
> 2019-02-20 11:12:40.129989 [DEBUG] switch_channel.c:3773 (sofia/internal/
> 9999999999999999999 at sip.myapp.net) Callstate Change RINGING -> ACTIVE
> 2019-02-20 11:12:40.129989 [DEBUG] switch_core_codec.c:248 sofia/internal/
> 1234567890 at sip.myapp.net Restore previous codec PCMA:8.
> 2019-02-20 11:12:40.129989 [DEBUG] switch_core_media.c:6861 Audio params
> are unchanged for sofia/internal/1234567890 at sip.myapp.net.
> 2019-02-20 11:12:40.129989 [DEBUG] mod_sofia.c:850 Local SDP
> sofia/internal/1234567890 at sip.myapp.net:
>
> Jurijs
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com



-- 
Sincerely,

Giovanni Maruzzelli
OpenTelecom.IT
cell: +39 347 266 56 18
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From jurijs.ivolga at gmail.com  Wed Feb 20 13:15:58 2019
From: jurijs.ivolga at gmail.com (Jurijs Ivolga)
Date: Wed, 20 Feb 2019 15:15:58 +0200
Subject: [Freeswitch-users] 1.8 DTLS calls
In-Reply-To: <CALXCt0rL_D4jZq0HCCCssg1vf4QdfGe7U+RxBVBwe3Xc7Kd1Sw@mail.gmail.com>
References: <CAOrA2UZK=nNhSnm6CR8xieRbrsXHUoFVQnq7sMXT5OxCcm2n0g@mail.gmail.com>
 <CALXCt0rL_D4jZq0HCCCssg1vf4QdfGe7U+RxBVBwe3Xc7Kd1Sw@mail.gmail.com>
Message-ID: <CAOrA2UbTpHA3ZAG1WC=DdgUGJ-0A9Fy+BogOs13avR+Zf=4svA@mail.gmail.com>

Hi,

I doubt that this is in network or configuration error. Everything are same
between 1.6 & 1.8. I'm using docker and difference are only in Docker file
where I build freeswitch. If I rebuild with old Docker file(1.6) then it
works fine, but with new Docker file for 1.8, I have problems. Everything
else seems to work, but DTLS calls...

Logs:
https://pastebin.com/g2FqHWWz

Jurijs


On Wed, Feb 20, 2019 at 2:47 PM Giovanni Maruzzelli <gmaruzz at gmail.com>
wrote:

> On Wed, Feb 20, 2019 at 12:59 PM Jurijs Ivolga <jurijs.ivolga at gmail.com>
> wrote:
>
>> Hi,
>>
>> I'm trying to migrate from 1.6.20 to 1.8.5, but I'm facing a problem with
>> DTLS calls(media_webrtc=true).
>>
>> Does anybody has same issues as me?
>>
>> I have following setup:
>>
>> RTP=>Freeswitch=>DTLS
>>
>> In 1.8.5, when Freeswitch receive 200 from DTLS endpoint, but it never
>> sends it back to RTP endpoint.
>>
>
> maybe you have a network problem, or a configuration problem network
> related in 1.8 deployment?
>
> Also, for us to better understand: from fs cli, "sofia global siptrace
> on", after this, copy and paste the ENTIRE debug level output from
> beginning to end in a pastebin.
>
> Then put here the pastebin link
>
> -giovanni
>
>
>
>>
>> Here some snippet from logs, where call stuck:
>>
>> 2019-02-20 09:50:07.368784 [INFO] switch_rtp.c:3832 Changing audio DTLS
>> state from OFF to HANDSHAKE
>> 2019-02-20 09:50:07.368784 [DEBUG] switch_core_media.c:8815
>> sofia/internal/9999999999999999999 at sip.myapp.net Set 2833 dtmf send
>> payload to 101
>> 2019-02-20 09:50:07.368784 [DEBUG] switch_core_media.c:8822
>> sofia/internal/9999999999999999999 at sip.myapp.net Set 2833 dtmf receive
>> payload to 101
>> 2019-02-20 09:50:07.368784 [DEBUG] switch_core_media.c:8845
>> sofia/internal/9999999999999999999 at sip.myapp.net Set rtp dtmf delay to 40
>>
>> In 1.6.20 everything works as expected:
>>
>> 2019-02-20 11:12:40.129989 [DEBUG] switch_core_media.c:7180
>> sofia/internal/9999999999999999999 at sip.myapp.net Set 2833 dtmf send
>> payload to 101
>> 2019-02-20 11:12:40.129989 [DEBUG] switch_core_media.c:7187
>> sofia/internal/9999999999999999999 at sip.myapp.net Set 2833 dtmf receive
>> payload to 101
>> 2019-02-20 11:12:40.129989 [DEBUG] switch_core_media.c:7210
>> sofia/internal/9999999999999999999 at sip.myapp.net Set rtp dtmf delay to 40
>> 2019-02-20 11:12:40.129989 [NOTICE] sofia.c:8218 Channel [sofia/internal/
>> 9999999999999999999 at sip.myapp.net] has been answered
>> 2019-02-20 11:12:40.129989 [DEBUG] switch_channel.c:3773 (sofia/internal/
>> 9999999999999999999 at sip.myapp.net) Callstate Change RINGING -> ACTIVE
>> 2019-02-20 11:12:40.129989 [DEBUG] switch_core_codec.c:248 sofia/internal/
>> 1234567890 at sip.myapp.net Restore previous codec PCMA:8.
>> 2019-02-20 11:12:40.129989 [DEBUG] switch_core_media.c:6861 Audio params
>> are unchanged for sofia/internal/1234567890 at sip.myapp.net.
>> 2019-02-20 11:12:40.129989 [DEBUG] mod_sofia.c:850 Local SDP
>> sofia/internal/1234567890 at sip.myapp.net:
>>
>> Jurijs
>> _________________________________________________________________________
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
>
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> OpenTelecom.IT
> cell: +39 347 266 56 18
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From gmaruzz at gmail.com  Wed Feb 20 14:15:51 2019
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Wed, 20 Feb 2019 15:15:51 +0100
Subject: [Freeswitch-users] 1.8 DTLS calls
In-Reply-To: <CAOrA2UbTpHA3ZAG1WC=DdgUGJ-0A9Fy+BogOs13avR+Zf=4svA@mail.gmail.com>
References: <CAOrA2UZK=nNhSnm6CR8xieRbrsXHUoFVQnq7sMXT5OxCcm2n0g@mail.gmail.com>
 <CALXCt0rL_D4jZq0HCCCssg1vf4QdfGe7U+RxBVBwe3Xc7Kd1Sw@mail.gmail.com>
 <CAOrA2UbTpHA3ZAG1WC=DdgUGJ-0A9Fy+BogOs13avR+Zf=4svA@mail.gmail.com>
Message-ID: <CALXCt0rRvfd_3dAJTrekC7pHqqUK1dfC_Z9H2Z_sUutuN0OzEw@mail.gmail.com>

You stopped the log much before the call ends... Can you pastebin a
complete log, up to DESTROY CS ?

Also, out of memory, do you have the things related to transcode media and
mixing media activated in config? Do FreeSWITCH find a media candidate
during ICE bargaining?

If you pastebin a complete log maybe we can help more

-giovanni

On Wed, Feb 20, 2019 at 2:58 PM Jurijs Ivolga <jurijs.ivolga at gmail.com>
wrote:

> Hi,
>
> I doubt that this is in network or configuration error. Everything are
> same between 1.6 & 1.8. I'm using docker and difference are only in Docker
> file where I build freeswitch. If I rebuild with old Docker file(1.6) then
> it works fine, but with new Docker file for 1.8, I have problems.
> Everything else seems to work, but DTLS calls...
>
> Logs:
> https://pastebin.com/g2FqHWWz
>
> Jurijs
>
>
> On Wed, Feb 20, 2019 at 2:47 PM Giovanni Maruzzelli <gmaruzz at gmail.com>
> wrote:
>
>> On Wed, Feb 20, 2019 at 12:59 PM Jurijs Ivolga <jurijs.ivolga at gmail.com>
>> wrote:
>>
>>> Hi,
>>>
>>> I'm trying to migrate from 1.6.20 to 1.8.5, but I'm facing a problem
>>> with DTLS calls(media_webrtc=true).
>>>
>>> Does anybody has same issues as me?
>>>
>>> I have following setup:
>>>
>>> RTP=>Freeswitch=>DTLS
>>>
>>> In 1.8.5, when Freeswitch receive 200 from DTLS endpoint, but it never
>>> sends it back to RTP endpoint.
>>>
>>
>> maybe you have a network problem, or a configuration problem network
>> related in 1.8 deployment?
>>
>> Also, for us to better understand: from fs cli, "sofia global siptrace
>> on", after this, copy and paste the ENTIRE debug level output from
>> beginning to end in a pastebin.
>>
>> Then put here the pastebin link
>>
>> -giovanni
>>
>>
>>
>>>
>>> Here some snippet from logs, where call stuck:
>>>
>>> 2019-02-20 09:50:07.368784 [INFO] switch_rtp.c:3832 Changing audio DTLS
>>> state from OFF to HANDSHAKE
>>> 2019-02-20 09:50:07.368784 [DEBUG] switch_core_media.c:8815
>>> sofia/internal/9999999999999999999 at sip.myapp.net Set 2833 dtmf send
>>> payload to 101
>>> 2019-02-20 09:50:07.368784 [DEBUG] switch_core_media.c:8822
>>> sofia/internal/9999999999999999999 at sip.myapp.net Set 2833 dtmf receive
>>> payload to 101
>>> 2019-02-20 09:50:07.368784 [DEBUG] switch_core_media.c:8845
>>> sofia/internal/9999999999999999999 at sip.myapp.net Set rtp dtmf delay to
>>> 40
>>>
>>> In 1.6.20 everything works as expected:
>>>
>>> 2019-02-20 11:12:40.129989 [DEBUG] switch_core_media.c:7180
>>> sofia/internal/9999999999999999999 at sip.myapp.net Set 2833 dtmf send
>>> payload to 101
>>> 2019-02-20 11:12:40.129989 [DEBUG] switch_core_media.c:7187
>>> sofia/internal/9999999999999999999 at sip.myapp.net Set 2833 dtmf receive
>>> payload to 101
>>> 2019-02-20 11:12:40.129989 [DEBUG] switch_core_media.c:7210
>>> sofia/internal/9999999999999999999 at sip.myapp.net Set rtp dtmf delay to
>>> 40
>>> 2019-02-20 11:12:40.129989 [NOTICE] sofia.c:8218 Channel [sofia/internal/
>>> 9999999999999999999 at sip.myapp.net] has been answered
>>> 2019-02-20 11:12:40.129989 [DEBUG] switch_channel.c:3773 (sofia/internal/
>>> 9999999999999999999 at sip.myapp.net) Callstate Change RINGING -> ACTIVE
>>> 2019-02-20 11:12:40.129989 [DEBUG] switch_core_codec.c:248
>>> sofia/internal/1234567890 at sip.myapp.net Restore previous codec PCMA:8.
>>> 2019-02-20 11:12:40.129989 [DEBUG] switch_core_media.c:6861 Audio params
>>> are unchanged for sofia/internal/1234567890 at sip.myapp.net.
>>> 2019-02-20 11:12:40.129989 [DEBUG] mod_sofia.c:850 Local SDP
>>> sofia/internal/1234567890 at sip.myapp.net:
>>>
>>> Jurijs
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Services
>>> sales at freeswitch.com
>>> https://freeswitch.com
>>>
>>> Official FreeSWITCH Sites
>>> https://freeswitch.com/oss
>>> https://freeswitch.org/confluence
>>> https://cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> https://freeswitch.com
>>
>>
>>
>> --
>> Sincerely,
>>
>> Giovanni Maruzzelli
>> OpenTelecom.IT
>> cell: +39 347 266 56 18
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com



-- 
Sincerely,

Giovanni Maruzzelli
OpenTelecom.IT
cell: +39 347 266 56 18
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From jurijs.ivolga at gmail.com  Wed Feb 20 14:44:47 2019
From: jurijs.ivolga at gmail.com (Jurijs Ivolga)
Date: Wed, 20 Feb 2019 16:44:47 +0200
Subject: [Freeswitch-users] 1.8 DTLS calls
In-Reply-To: <CALXCt0rRvfd_3dAJTrekC7pHqqUK1dfC_Z9H2Z_sUutuN0OzEw@mail.gmail.com>
References: <CAOrA2UZK=nNhSnm6CR8xieRbrsXHUoFVQnq7sMXT5OxCcm2n0g@mail.gmail.com>
 <CALXCt0rL_D4jZq0HCCCssg1vf4QdfGe7U+RxBVBwe3Xc7Kd1Sw@mail.gmail.com>
 <CAOrA2UbTpHA3ZAG1WC=DdgUGJ-0A9Fy+BogOs13avR+Zf=4svA@mail.gmail.com>
 <CALXCt0rRvfd_3dAJTrekC7pHqqUK1dfC_Z9H2Z_sUutuN0OzEw@mail.gmail.com>
Message-ID: <CAOrA2UYm3WnWqsMRrw+Tae3CvkiCECbLq9NFwK9uwEyCF2PsKw@mail.gmail.com>

Hi Giovanni,

Thank you for tips!

There is nothing happening after what I sent, so call just stuck there.
Only if I hangup on any side.

Only I can to add that after 30 seconds I just get this line:

2019-02-20 14:22:07.195060 [ERR] switch_rtp.c:834 No audio stun for a long
time!

If I hangup calls on both side, call still shows in freeswitch as active if
I do show calls.

I believe I have everything needed for trans-coding in config files, cause
calls where no DTLS involved but with transcoding, calls works fine. In
that case I have RTP(ALAW) => Freeswitch => SRTP(OPUS)

Regarding ICE:

2019-02-20 12:51:21.532911 [INFO] switch_core_media.c:8690 Activating Audio
ICE
2019-02-20 12:51:21.532911 [NOTICE] switch_rtp.c:4799 Activating RTP audio
ICE: 1cf2011d:u4NGJi1myDzrpRoh 10.99.88.122:55465
2019-02-20 12:51:21.532911 [DEBUG] switch_core_media.c:8731 Activating RTCP
PORT 55465
2019-02-20 12:51:21.532911 [DEBUG] switch_rtp.c:4696 RTCP send rate is:
1000 and packet rate is: 20000 Remote Port: 55465
2019-02-20 12:51:21.532911 [INFO] switch_core_media.c:8742 Skipping RTCP
ICE (Same as RTP)

Jurijs


On Wed, Feb 20, 2019 at 4:17 PM Giovanni Maruzzelli <gmaruzz at gmail.com>
wrote:

>
> You stopped the log much before the call ends... Can you pastebin a
> complete log, up to DESTROY CS ?
>
> Also, out of memory, do you have the things related to transcode media and
> mixing media activated in config? Do FreeSWITCH find a media candidate
> during ICE bargaining?
>
> If you pastebin a complete log maybe we can help more
>
> -giovanni
>
> On Wed, Feb 20, 2019 at 2:58 PM Jurijs Ivolga <jurijs.ivolga at gmail.com>
> wrote:
>
>> Hi,
>>
>> I doubt that this is in network or configuration error. Everything are
>> same between 1.6 & 1.8. I'm using docker and difference are only in Docker
>> file where I build freeswitch. If I rebuild with old Docker file(1.6) then
>> it works fine, but with new Docker file for 1.8, I have problems.
>> Everything else seems to work, but DTLS calls...
>>
>> Logs:
>> https://pastebin.com/g2FqHWWz
>>
>> Jurijs
>>
>>
>> On Wed, Feb 20, 2019 at 2:47 PM Giovanni Maruzzelli <gmaruzz at gmail.com>
>> wrote:
>>
>>> On Wed, Feb 20, 2019 at 12:59 PM Jurijs Ivolga <jurijs.ivolga at gmail.com>
>>> wrote:
>>>
>>>> Hi,
>>>>
>>>> I'm trying to migrate from 1.6.20 to 1.8.5, but I'm facing a problem
>>>> with DTLS calls(media_webrtc=true).
>>>>
>>>> Does anybody has same issues as me?
>>>>
>>>> I have following setup:
>>>>
>>>> RTP=>Freeswitch=>DTLS
>>>>
>>>> In 1.8.5, when Freeswitch receive 200 from DTLS endpoint, but it never
>>>> sends it back to RTP endpoint.
>>>>
>>>
>>> maybe you have a network problem, or a configuration problem network
>>> related in 1.8 deployment?
>>>
>>> Also, for us to better understand: from fs cli, "sofia global siptrace
>>> on", after this, copy and paste the ENTIRE debug level output from
>>> beginning to end in a pastebin.
>>>
>>> Then put here the pastebin link
>>>
>>> -giovanni
>>>
>>>
>>>
>>>>
>>>> Here some snippet from logs, where call stuck:
>>>>
>>>> 2019-02-20 09:50:07.368784 [INFO] switch_rtp.c:3832 Changing audio DTLS
>>>> state from OFF to HANDSHAKE
>>>> 2019-02-20 09:50:07.368784 [DEBUG] switch_core_media.c:8815
>>>> sofia/internal/9999999999999999999 at sip.myapp.net Set 2833 dtmf send
>>>> payload to 101
>>>> 2019-02-20 09:50:07.368784 [DEBUG] switch_core_media.c:8822
>>>> sofia/internal/9999999999999999999 at sip.myapp.net Set 2833 dtmf receive
>>>> payload to 101
>>>> 2019-02-20 09:50:07.368784 [DEBUG] switch_core_media.c:8845
>>>> sofia/internal/9999999999999999999 at sip.myapp.net Set rtp dtmf delay to
>>>> 40
>>>>
>>>> In 1.6.20 everything works as expected:
>>>>
>>>> 2019-02-20 11:12:40.129989 [DEBUG] switch_core_media.c:7180
>>>> sofia/internal/9999999999999999999 at sip.myapp.net Set 2833 dtmf send
>>>> payload to 101
>>>> 2019-02-20 11:12:40.129989 [DEBUG] switch_core_media.c:7187
>>>> sofia/internal/9999999999999999999 at sip.myapp.net Set 2833 dtmf receive
>>>> payload to 101
>>>> 2019-02-20 11:12:40.129989 [DEBUG] switch_core_media.c:7210
>>>> sofia/internal/9999999999999999999 at sip.myapp.net Set rtp dtmf delay to
>>>> 40
>>>> 2019-02-20 11:12:40.129989 [NOTICE] sofia.c:8218 Channel
>>>> [sofia/internal/9999999999999999999 at sip.myapp.net] has been answered
>>>> 2019-02-20 11:12:40.129989 [DEBUG] switch_channel.c:3773
>>>> (sofia/internal/9999999999999999999 at sip.myapp.net) Callstate Change
>>>> RINGING -> ACTIVE
>>>> 2019-02-20 11:12:40.129989 [DEBUG] switch_core_codec.c:248
>>>> sofia/internal/1234567890 at sip.myapp.net Restore previous codec PCMA:8.
>>>> 2019-02-20 11:12:40.129989 [DEBUG] switch_core_media.c:6861 Audio
>>>> params are unchanged for sofia/internal/1234567890 at sip.myapp.net.
>>>> 2019-02-20 11:12:40.129989 [DEBUG] mod_sofia.c:850 Local SDP
>>>> sofia/internal/1234567890 at sip.myapp.net:
>>>>
>>>> Jurijs
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Services
>>>> sales at freeswitch.com
>>>> https://freeswitch.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> https://freeswitch.com/oss
>>>> https://freeswitch.org/confluence
>>>> https://cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> https://freeswitch.com
>>>
>>>
>>>
>>> --
>>> Sincerely,
>>>
>>> Giovanni Maruzzelli
>>> OpenTelecom.IT
>>> cell: +39 347 266 56 18
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Services
>>> sales at freeswitch.com
>>> https://freeswitch.com
>>>
>>> Official FreeSWITCH Sites
>>> https://freeswitch.com/oss
>>> https://freeswitch.org/confluence
>>> https://cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> https://freeswitch.com
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
>
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> OpenTelecom.IT
> cell: +39 347 266 56 18
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From imfanee at gmail.com  Wed Feb 20 04:04:08 2019
From: imfanee at gmail.com (Faisal Hanif)
Date: Wed, 20 Feb 2019 09:04:08 +0500
Subject: [Freeswitch-users] switch_ivr
In-Reply-To: <CAPr4L28nuPSNQq_fZART7b5w=kWm3Q8vffCx6dJ=yNx1xqqdyA@mail.gmail.com>
References: <CAPr4L28nuPSNQq_fZART7b5w=kWm3Q8vffCx6dJ=yNx1xqqdyA@mail.gmail.com>
Message-ID: <CAO5BruAmAFZD1neU-zsgS491JO3ZBwi-Sy=uYhRVmYwfJS+wTQ@mail.gmail.com>

You can see all in I've.conf.xml

On Wed, Feb 20, 2019, 2:54 AM Juan Nino <ingjfnc at gmail.com wrote:

> HI
>
> chekcing the logs I have seen actions taken by switch_ivr.c
> is there any way I can disable it? and/or check how does it work?
> I have commented mod_ivr but this does not seem to affect
> thanks
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From ceniy.mei at gmail.com  Wed Feb 20 07:24:58 2019
From: ceniy.mei at gmail.com (=?UTF-8?B?5qKF6KGo6YC1?=)
Date: Wed, 20 Feb 2019 15:24:58 +0800
Subject: [Freeswitch-users] [ERR] too many stacked extensions
In-Reply-To: <CAGJLwFikoeV2FKfwdF+HmCwQJdUZ_AfUscher+LWggxU+Xa8gg@mail.gmail.com>
References: <CABcV8PHtEAsHXKKKWnsas+u63Wy2FKYgQUWqa-Jys7mMmSpr1g@mail.gmail.com>
 <CAGJLwFikoeV2FKfwdF+HmCwQJdUZ_AfUscher+LWggxU+Xa8gg@mail.gmail.com>
Message-ID: <CABcV8PFA9u47y7D3bsFX7oaJakAapB77m61i3aFNO3WUx_-m1g@mail.gmail.com>

Thanks

I will try without Lua first.

Alexey Sibyakin <alex at freeswitch.com> 于2019年2月20日周三 上午8:13写道：

> Hi,
>
> First of all, you need to update to 1.8.5. Second, try without Lua.
>
> Regards,
>
> Alex
>
> Alex Sibyakin | Support Engineer SignalWire | 228 Hamilton Ave 3rd Floor,
> Palo Alto, CA 94303 Email: alex at freeswitch.com Website:
> https://www.signalwire.com
>
>
>
> On Tue, Feb 19, 2019 at 10:43 PM 梅表逵 <ceniy.mei at gmail.com> wrote:
>
>> Hi,
>>
>>
>> I’m running freeswitch version 1.8.1. In the log files, I saw a lot of
>> [ERR] too many stacked extensions. It happens when freeswitch is trying to
>> bridge  a callcenter member to agent.  I don’t know how to handle it. Could
>> you please give any advises?
>>
>>
>>
>> 55791bf8-30bb-11e9-a4fd-7125b8df8c7a 2019-02-15 08:47:56.419673 [DEBUG]
>> switch_ivr_bridge.c:1063 sofia/internal/137095 at 10.20.86.100:5060 CUSTOM
>> HOLD
>> 55791bf8-30bb-11e9-a4fd-7125b8df8c7a 2019-02-15 08:47:56.419673 [DEBUG]
>> switch_core_state_machine.c:662 (sofia/internal/137095 at 10.20.86.100:5060)
>> State CONSUME_MEDIA going to sleep
>> 55791bf8-30bb-11e9-a4fd-7125b8df8c7a 2019-02-15 08:47:56.419673 [ERR]
>> switch_ivr.c:902 Error sofia/internal/137095 at 10.20.86.100:5060 too many
>> stacked extensions
>> 55791bf8-30bb-11e9-a4fd-7125b8df8c7a 2019-02-15 08:47:56.419673 [ERR]
>> switch_ivr.c:902 Error sofia/internal/137095 at 10.20.86.100:5060 too many
>> stacked extensions
>> 55791bf8-30bb-11e9-a4fd-7125b8df8c7a 2019-02-15 08:47:56.419673 [DEBUG]
>> switch_ivr_bridge.c:1744 (sofia/internal/137095 at 10.20.86.100:5060) State
>> Change CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA
>> 55791bf8-30bb-11e9-a4fd-7125b8df8c7a 2019-02-15 08:47:56.419673 [ERR]
>> switch_ivr.c:902 Error sofia/internal/137095 at 10.20.86.100:5060 too many
>> stacked extensions
>> 55791bf8-30bb-11e9-a4fd-7125b8df8c7a 2019-02-15 08:47:56.419673 [ERR]
>> switch_ivr.c:902 Error sofia/internal/137095 at 10.20.86.100:5060 too many
>> stacked extensions
>> 55791bf8-30bb-11e9-a4fd-7125b8df8c7a 2019-02-15 08:47:56.419673 [DEBUG]
>> switch_core_state_machine.c:584 (sofia/internal/137095 at 10.20.86.100:5060)
>> Running State Change CS_EXCHANGE_MEDIA (Cur 2 Tot 4919)
>> 55791bf8-30bb-11e9-a4fd-7125b8df8c7a 2019-02-15 08:47:56.419673 [DEBUG]
>> switch_core_state_machine.c:653 (sofia/internal/137095 at 10.20.86.100:5060)
>> State EXCHANGE_MEDIA
>> 55791bf8-30bb-11e9-a4fd-7125b8df8c7a 2019-02-15 08:47:56.419673 [DEBUG]
>> mod_sofia.c:645 SOFIA EXCHANGE_MEDIA
>>
>>
>> XML:
>> <condition field="destination_number" expression="^(31\d{4})$" >
>>    <action application="set" data="dialed_extension=$1"/>
>>    <!-- hangup after successful bridge to agent -->
>>    <action application="set" data="recording_follow_transfer=true"/>
>>    <action application="set" data="hangup_after_bridge=true"/>
>>    <action application="set" data="say_agent_first=1" />
>>    <action application="set" data="bridge_pre_execute_aleg_app=lua"/>
>>    <action application="set"
>> data="bridge_pre_execute_aleg_data=broadcastAgent.lua"/>
>>   <action application="set" data="call_type=q" />
>>   <action application="bind_meta_app" data="4 ab s
>> execute_extension::att_xfer XML features"/>
>>   <action application="callcenter" data="$1"/>
>> </condition>
>> </extension>
>>
>>
>> Thanks
>> Ceniy Mei
>> _________________________________________________________________________
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From alaadin.abd at gmail.com  Wed Feb 20 07:41:53 2019
From: alaadin.abd at gmail.com (Alaadin Abdurrahman)
Date: Wed, 20 Feb 2019 09:41:53 +0200
Subject: [Freeswitch-users] Connecting endpoint to domain within Free
	Switch
In-Reply-To: <333d641f-1b8a-4c69-5dcc-bf276e01640d@tluka.co.za>
References: <mailman.222.1550593495.1400.freeswitch-users@lists.freeswitch.org>
 <CAHtxdDfDSpcb72OGFrBzmWRro8OsZki2DU8fU=KZFX98_dpBuw@mail.gmail.com>
 <mailman.229.1550605882.1400.freeswitch-users@lists.freeswitch.org>
 <333d641f-1b8a-4c69-5dcc-bf276e01640d@tluka.co.za>
Message-ID: <CAJVaC-uDQX9gxTLV8CpDkAd5T4yS4eqNO7ibAPRVFn14B9NF_w@mail.gmail.com>

Hi,
You need to setup your UA client to use your domain in the domain filed and
the host IP in the proxy filed

Regards
Ala

On Wed, Feb 20, 2019 at 7:45 AM Koketso Mabuse <koketsom at tluka.co.za> wrote:

> Hi there
>
> Scenario: Fresh installed FusionPBX:
>
>    - Default domain with local IP,
>    - created a 2nd domain as per tutorial,
>    - created extensions.
>       - tried to register extensions on 2nd domain failure
>          - changed domain on extion to point to 1st main domain, end
>          point registers.
>
> Issue: how do I create 2 extensions  eg: 1000 for domain 1 and 1000 for
> domain 2 and register them both on main domain?
>
> Domain:mtcc
>
> exmple: Username: 1000
>
>                Password: pass123
>
>                host: 192.168.1.130
>
>
> Domain: tluka
>
> Username: 1000
>
>                Password: pass123
>
>                host: 192.168.1.130
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From mike at freeswitch.com  Wed Feb 20 17:10:55 2019
From: mike at freeswitch.com (Mike Jerris)
Date: Wed, 20 Feb 2019 12:10:55 -0500
Subject: [Freeswitch-users] setting contact domain in dialplan not
	working
In-Reply-To: <F363AB89-018F-4774-941D-74894521E932@dissauer.net>
References: <F363AB89-018F-4774-941D-74894521E932@dissauer.net>
Message-ID: <CADrq_yGu5U40LRUfRwU+zLdyxVN8H_4vbthWoetZURBFtPFpmA@mail.gmail.com>

There is no way to change that

On Tue, Feb 19, 2019 at 5:24 PM Roman Dissauer <roman at dissauer.net> wrote:

> Is there a way to get the FS public IP replaced by a custom domain name in
> contact header using the dialplan?
>
> tried:
>
> <action application="export" data="sip_contact_host=test.domain.tld“/>
>      ...not working
> <action application=„set" data="sip_contact_host=test.domain.tld“/>
>  ...not working
> <action application="export" data="sip_invite_domain=test.domain.tld“/>
>      ...not working
> <action application=„set" data="sip_invite_domain=test.domain.tld“/>
>    ...not working
> <action application="export" data=„sip_contact_user=bla at test.domain.tld"/>
>        ...not working
> <action application=„set" data=„sip_invite_domain=bla at test.domain.tld“/>
>        ...not working
>
>
> only thing working is setting the parameter in gateway config,
> <param name="contact-host" value=„test.domain.tld“/>
>
> but this is not working for me because I need to do it dynamically in
> dialplan!
>
> Any hint for me?
>
> thank you!
> Roman
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From mario_fs at mgtech.com  Wed Feb 20 17:16:20 2019
From: mario_fs at mgtech.com (mario_fs)
Date: Wed, 20 Feb 2019 09:16:20 -0800
Subject: [Freeswitch-users] Option to log registrations/deregistrations?
In-Reply-To: <771A1361-D809-41AF-911F-1A4077DCF898@mgtech.com>
References: <B01E1C86-E101-4965-B686-606BBE720E70@mgtech.com>
 <01000168f0302208-303a2b9c-ffa8-4cad-ac4e-d87797cc2c4d-000000@email.amazonses.com>
 <771A1361-D809-41AF-911F-1A4077DCF898@mgtech.com>
Message-ID: <B5A02804-7121-4841-8866-DC22B2482924@mgtech.com>

I have been searching the wiki and web and still nothing simple to log de-registrations. I find it hard to believe there is not a simple way to do this, anyone have a good idea?
Mario G

> On Feb 15, 2019, at 9:46 AM, mario_fs <mario_fs at mgtech.com> wrote:
> 
> Hmmmm…. Logging de-registrations would be a useful enhancement to FreeSwitch for debugging issues like this. I know a SIP trace does it but that’s a LOT of data when we're talking days. Thanks.
> Mario G
> 
>> On Feb 15, 2019, at 12:05 AM, Avi Marcus <avi at avimarcus.net <mailto:avi at avimarcus.net>> wrote:
>> 
>> Here's some old nodejs code relying on `esl` lib to monitor for de-registrations and send me an email. I haven't used it in ages.
>> 
>> https://gist.github.com/avimar/9645dd3134476d458e68cca077f4ed16 <https://gist.github.com/avimar/9645dd3134476d458e68cca077f4ed16>
>> 
>> 
>> -Avi Marcus
>> BestFone
>> 
>> 
>> On Fri, Feb 15, 2019 at 12:13 AM mario_fs <mario_fs at mgtech.com <mailto:mario_fs at mgtech.com>> wrote:
>> Does anyone know if there is something I can do to log device registrations, and more importantly when devices lose registration, without having a ton of other log data. I know I can turn on a higher level of diagnostics, but it may be several days before I find what I need and don’t want to flood the log.
>> 
>> I am working with Counterpath on Bria Push service and need to find when their push server stops registering. Since they don’t send a notification for that and it’s not in the normal FS log, I may not know for days when this happens. It would be nice to be able to scan the log for lost registrations. Thanks!
>> Mario G
>> _________________________________________________________________________
>> Professional FreeSWITCH Services
>> sales at freeswitch.com <mailto:sales at freeswitch.com>
>> https://freeswitch.com <https://freeswitch.com/>
>> 
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss <https://freeswitch.com/oss>
>> https://freeswitch.org/confluence <https://freeswitch.org/confluence>
>> https://cluecon.com <https://cluecon.com/>
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
>> https://freeswitch.com <https://freeswitch.com/>_________________________________________________________________________
>> Professional FreeSWITCH Services
>> sales at freeswitch.com <mailto:sales at freeswitch.com>
>> https://freeswitch.com
>> 
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
> 
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
> 
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com

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From mike at freeswitch.com  Wed Feb 20 17:13:39 2019
From: mike at freeswitch.com (Mike Jerris)
Date: Wed, 20 Feb 2019 12:13:39 -0500
Subject: [Freeswitch-users] [FreeSWITCH Install Issue]
In-Reply-To: <CAFGRPVr49biC1HLY2yz6Sh-EhyhqG77te+RVZ65-enaeHpDuZQ@mail.gmail.com>
References: <CADSVerCzDkRK6mkj6D16Ois-ZynE_eTbiteH=Vm_oovvFBNcsQ@mail.gmail.com>
 <CAGJLwFhNka3iuVHAAmTy3bWibzpbxqzqyL+4_J858wgUN1gj2A@mail.gmail.com>
 <CAFGRPVr49biC1HLY2yz6Sh-EhyhqG77te+RVZ65-enaeHpDuZQ@mail.gmail.com>
Message-ID: <CADrq_yFGW11f1VbmteRpvput1COC95mDMP6f1pYFxCqUY65xjQ@mail.gmail.com>

This is incorrect.  Primary development is on Debinan 9 and it should work
best of any distro.

On Tue, Feb 19, 2019 at 10:27 AM David Villasmil <
david.villasmil.work at gmail.com> wrote:

> You should use Debian 8, not 9. At least for the time being.
> On Tue, 19 Feb 2019 at 15:15, Alexey Sibyakin <alex at freeswitch.com> wrote:
>
>> Hi,
>>
>> Try our awesome packages
>> https://freeswitch.org/confluence/display/FREESWITCH/Debian+9+Stretch
>>
>> You should compile yourself only if you absolutely know what you are
>> doing.
>>
>> It looks like you did *make -all* instead of *make *or *make all*.
>>
>> Regards,
>>
>> Alex
>>
>> Alex Sibyakin | Support Engineer SignalWire | 228 Hamilton Ave 3rd Floor,
>> Palo Alto, CA 94303 Email: alex at freeswitch.com Website:
>> https://www.signalwire.com
>>
>>
>>
>> On Tue, Feb 19, 2019 at 11:16 PM Bilal Abbasi <bilaln018 at gmail.com>
>> wrote:
>>
>>> Hi Users,
>>> I am having issues with installing FreeSWITCH 1.8.5 on Debian 9.6,
>>> Here is the error i am getting this error when i run make command.
>>>
>>>  CC       src/libfreeswitch_la-switch_msrp.lo
>>>
>>>   CC       src/libfreeswitch_la-switch_vad.lo
>>>
>>>   CC       src/libfreeswitch_la-switch_vpx.lo
>>>
>>>   CXX      src/switch_cpp.lo
>>>
>>>   CXXLD    libfreeswitch.la
>>>
>>>
>>> *** Warning: Linking the shared library libfreeswitch.la against the
>>>
>>> *** static library libs/libvpx/libvpx.a is not portable!
>>>
>>> ar: `u' modifier ignored since `D' is the default (see `U')
>>>
>>> Making all in .
>>>
>>> make[2]: Entering directory '/usr/src/freeswitch-1.8.5'
>>>
>>>   CC       src/freeswitch-switch.o
>>>
>>>   CCLD     freeswitch
>>>
>>> /usr/bin/ld: warning: libssl.so.1.0.2, needed by
>>> //usr/lib/x86_64-linux-gnu/libcurl.so.4, may conflict with libssl.so.1.1
>>>
>>> /usr/bin/ld: warning: libcrypto.so.1.0.2, needed by
>>> //usr/lib/x86_64-linux-gnu/libcurl.so.4, may conflict with libcrypto.so.1.1
>>>
>>>   CC       src/tone2wav-tone2wav.o
>>>
>>>   CCLD     tone2wav
>>>
>>> /usr/bin/ld: warning: libssl.so.1.0.2, needed by
>>> //usr/lib/x86_64-linux-gnu/libcurl.so.4, may conflict with libssl.so.1.1
>>>
>>> /usr/bin/ld: warning: libcrypto.so.1.0.2, needed by
>>> //usr/lib/x86_64-linux-gnu/libcurl.so.4, may conflict with libcrypto.so.1.1
>>>
>>>   CC       src/fs_encode-fs_encode.o
>>>
>>>   CCLD     fs_encode
>>>
>>> /usr/bin/ld: warning: libssl.so.1.0.2, needed by
>>> //usr/lib/x86_64-linux-gnu/libcurl.so.4, may conflict with libssl.so.1.1
>>>
>>> /usr/bin/ld: warning: libcrypto.so.1.0.2, needed by
>>> //usr/lib/x86_64-linux-gnu/libcurl.so.4, may conflict with libcrypto.so.1.1
>>>
>>> make[2]: Leaving directory '/usr/src/freeswitch-1.8.5'
>>>
>>> Making all in src
>>>
>>> make[2]: Entering directory '/usr/src/freeswitch-1.8.5/src'
>>>
>>> Making all in mod
>>>
>>> make[3]: Entering directory '/usr/src/freeswitch-1.8.5/src/mod'
>>>
>>> Unknown target
>>>
>>> Makefile:721: recipe for target '-all' failed
>>>
>>> make[3]: *** [-all] Error 1
>>>
>>> make[3]: Leaving directory '/usr/src/freeswitch-1.8.5/src/mod'
>>>
>>> Makefile:591: recipe for target 'all-recursive' failed
>>>
>>> make[2]: *** [all-recursive] Error 1
>>>
>>> make[2]: Leaving directory '/usr/src/freeswitch-1.8.5/src'
>>>
>>> Makefile:3494: recipe for target 'all-recursive' failed
>>>
>>> make[1]: *** [all-recursive] Error 1
>>>
>>> make[1]: Leaving directory '/usr/src/freeswitch-1.8.5'
>>>
>>> Makefile:1255: recipe for target 'all' failed
>>>
>>> make: *** [all] Error 2
>>>
>>>
>>> Google did not helped me much on this, can somebody point me the right
>>> direction.
>>>
>>>
>>> Regards
>>>
>>> Abbasi
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Services
>>> sales at freeswitch.com
>>> https://freeswitch.com
>>>
>>> Official FreeSWITCH Sites
>>> https://freeswitch.com/oss
>>> https://freeswitch.org/confluence
>>> https://cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> https://freeswitch.com
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From mike at freeswitch.com  Wed Feb 20 17:14:59 2019
From: mike at freeswitch.com (Mike Jerris)
Date: Wed, 20 Feb 2019 12:14:59 -0500
Subject: [Freeswitch-users] [FreeSWITCH Install Issue]
In-Reply-To: <CADSVerBZ0vVk6WN+bxP9RvSopsQyxdNFmm3G4_AtdY56VP5yCQ@mail.gmail.com>
References: <CADSVerCzDkRK6mkj6D16Ois-ZynE_eTbiteH=Vm_oovvFBNcsQ@mail.gmail.com>
 <CAGJLwFhNka3iuVHAAmTy3bWibzpbxqzqyL+4_J858wgUN1gj2A@mail.gmail.com>
 <CAFGRPVr49biC1HLY2yz6Sh-EhyhqG77te+RVZ65-enaeHpDuZQ@mail.gmail.com>
 <CADSVerBZ0vVk6WN+bxP9RvSopsQyxdNFmm3G4_AtdY56VP5yCQ@mail.gmail.com>
Message-ID: <CADrq_yGnRzmesCTqBKcDKQARp79698bHE=CaVce9tBH2tS2udA@mail.gmail.com>

if you follow the instructions on confluence for building like this it will
get the right dependencies from our debian repositories and will fix this
issue.  we have to override a few distro packages to get the deps right.

On Tue, Feb 19, 2019 at 12:56 PM Bilal Abbasi <bilaln018 at gmail.com> wrote:

> I did “make” only, and got this.
> I dont want to install via apt-get rather want to build via source.
>
> Regards
> Abbasi
>
> On Tue, 19 Feb 2019 at 8:57 PM, David Villasmil <
> david.villasmil.work at gmail.com> wrote:
>
>> You should use Debian 8, not 9. At least for the time being.
>> On Tue, 19 Feb 2019 at 15:15, Alexey Sibyakin <alex at freeswitch.com>
>> wrote:
>>
>>> Hi,
>>>
>>> Try our awesome packages
>>> https://freeswitch.org/confluence/display/FREESWITCH/Debian+9+Stretch
>>>
>>> You should compile yourself only if you absolutely know what you are
>>> doing.
>>>
>>> It looks like you did *make -all* instead of *make *or *make all*.
>>>
>>> Regards,
>>>
>>> Alex
>>>
>>> Alex Sibyakin | Support Engineer SignalWire | 228 Hamilton Ave 3rd
>>> Floor, Palo Alto, CA 94303 Email: alex at freeswitch.com Website:
>>> https://www.signalwire.com
>>>
>>>
>>>
>>> On Tue, Feb 19, 2019 at 11:16 PM Bilal Abbasi <bilaln018 at gmail.com>
>>> wrote:
>>>
>>>> Hi Users,
>>>> I am having issues with installing FreeSWITCH 1.8.5 on Debian 9.6,
>>>> Here is the error i am getting this error when i run make command.
>>>>
>>>>  CC       src/libfreeswitch_la-switch_msrp.lo
>>>>
>>>>   CC       src/libfreeswitch_la-switch_vad.lo
>>>>
>>>>   CC       src/libfreeswitch_la-switch_vpx.lo
>>>>
>>>>   CXX      src/switch_cpp.lo
>>>>
>>>>   CXXLD    libfreeswitch.la
>>>>
>>>>
>>>> *** Warning: Linking the shared library libfreeswitch.la against the
>>>>
>>>> *** static library libs/libvpx/libvpx.a is not portable!
>>>>
>>>> ar: `u' modifier ignored since `D' is the default (see `U')
>>>>
>>>> Making all in .
>>>>
>>>> make[2]: Entering directory '/usr/src/freeswitch-1.8.5'
>>>>
>>>>   CC       src/freeswitch-switch.o
>>>>
>>>>   CCLD     freeswitch
>>>>
>>>> /usr/bin/ld: warning: libssl.so.1.0.2, needed by
>>>> //usr/lib/x86_64-linux-gnu/libcurl.so.4, may conflict with libssl.so.1.1
>>>>
>>>> /usr/bin/ld: warning: libcrypto.so.1.0.2, needed by
>>>> //usr/lib/x86_64-linux-gnu/libcurl.so.4, may conflict with libcrypto.so.1.1
>>>>
>>>>   CC       src/tone2wav-tone2wav.o
>>>>
>>>>   CCLD     tone2wav
>>>>
>>>> /usr/bin/ld: warning: libssl.so.1.0.2, needed by
>>>> //usr/lib/x86_64-linux-gnu/libcurl.so.4, may conflict with libssl.so.1.1
>>>>
>>>> /usr/bin/ld: warning: libcrypto.so.1.0.2, needed by
>>>> //usr/lib/x86_64-linux-gnu/libcurl.so.4, may conflict with libcrypto.so.1.1
>>>>
>>>>   CC       src/fs_encode-fs_encode.o
>>>>
>>>>   CCLD     fs_encode
>>>>
>>>> /usr/bin/ld: warning: libssl.so.1.0.2, needed by
>>>> //usr/lib/x86_64-linux-gnu/libcurl.so.4, may conflict with libssl.so.1.1
>>>>
>>>> /usr/bin/ld: warning: libcrypto.so.1.0.2, needed by
>>>> //usr/lib/x86_64-linux-gnu/libcurl.so.4, may conflict with libcrypto.so.1.1
>>>>
>>>> make[2]: Leaving directory '/usr/src/freeswitch-1.8.5'
>>>>
>>>> Making all in src
>>>>
>>>> make[2]: Entering directory '/usr/src/freeswitch-1.8.5/src'
>>>>
>>>> Making all in mod
>>>>
>>>> make[3]: Entering directory '/usr/src/freeswitch-1.8.5/src/mod'
>>>>
>>>> Unknown target
>>>>
>>>> Makefile:721: recipe for target '-all' failed
>>>>
>>>> make[3]: *** [-all] Error 1
>>>>
>>>> make[3]: Leaving directory '/usr/src/freeswitch-1.8.5/src/mod'
>>>>
>>>> Makefile:591: recipe for target 'all-recursive' failed
>>>>
>>>> make[2]: *** [all-recursive] Error 1
>>>>
>>>> make[2]: Leaving directory '/usr/src/freeswitch-1.8.5/src'
>>>>
>>>> Makefile:3494: recipe for target 'all-recursive' failed
>>>>
>>>> make[1]: *** [all-recursive] Error 1
>>>>
>>>> make[1]: Leaving directory '/usr/src/freeswitch-1.8.5'
>>>>
>>>> Makefile:1255: recipe for target 'all' failed
>>>>
>>>> make: *** [all] Error 2
>>>>
>>>>
>>>> Google did not helped me much on this, can somebody point me the right
>>>> direction.
>>>>
>>>>
>>>> Regards
>>>>
>>>> Abbasi
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Services
>>>> sales at freeswitch.com
>>>> https://freeswitch.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> https://freeswitch.com/oss
>>>> https://freeswitch.org/confluence
>>>> https://cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> https://freeswitch.com
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Services
>>> sales at freeswitch.com
>>> https://freeswitch.com
>>>
>>> Official FreeSWITCH Sites
>>> https://freeswitch.com/oss
>>> https://freeswitch.org/confluence
>>> https://cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> https://freeswitch.com
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From gmaruzz at gmail.com  Wed Feb 20 17:34:18 2019
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Wed, 20 Feb 2019 18:34:18 +0100
Subject: [Freeswitch-users] 1.8 DTLS calls
In-Reply-To: <CAOrA2UYm3WnWqsMRrw+Tae3CvkiCECbLq9NFwK9uwEyCF2PsKw@mail.gmail.com>
References: <CAOrA2UZK=nNhSnm6CR8xieRbrsXHUoFVQnq7sMXT5OxCcm2n0g@mail.gmail.com>
 <CALXCt0rL_D4jZq0HCCCssg1vf4QdfGe7U+RxBVBwe3Xc7Kd1Sw@mail.gmail.com>
 <CAOrA2UbTpHA3ZAG1WC=DdgUGJ-0A9Fy+BogOs13avR+Zf=4svA@mail.gmail.com>
 <CALXCt0rRvfd_3dAJTrekC7pHqqUK1dfC_Z9H2Z_sUutuN0OzEw@mail.gmail.com>
 <CAOrA2UYm3WnWqsMRrw+Tae3CvkiCECbLq9NFwK9uwEyCF2PsKw@mail.gmail.com>
Message-ID: <CALXCt0qXU1P5PObbW_JdvCzH8pWhW7+J0db8tS9OuZ8LqcBrgA@mail.gmail.com>

That seems to be a nat problem, or an ice misconfiguration, like using
wrong stun (that results in a nat problem, again).

I assume all is equal in server.

Check to see if you can give different parameters to clients, maybe between
releases Freeswitch changed in subtle modes how it react to subtle client
parameters. I would check all nat/ice/stun client parameters and experiment
with them...

Just a possible line of investigation, no warranties ;)

-giovanni



On Wed, Feb 20, 2019, 16:18 Jurijs Ivolga <jurijs.ivolga at gmail.com wrote:

> Hi Giovanni,
>
> Thank you for tips!
>
> There is nothing happening after what I sent, so call just stuck there.
> Only if I hangup on any side.
>
> Only I can to add that after 30 seconds I just get this line:
>
> 2019-02-20 14:22:07.195060 [ERR] switch_rtp.c:834 No audio stun for a long
> time!
>
> If I hangup calls on both side, call still shows in freeswitch as active
> if I do show calls.
>
> I believe I have everything needed for trans-coding in config files, cause
> calls where no DTLS involved but with transcoding, calls works fine. In
> that case I have RTP(ALAW) => Freeswitch => SRTP(OPUS)
>
> Regarding ICE:
>
> 2019-02-20 12:51:21.532911 [INFO] switch_core_media.c:8690 Activating
> Audio ICE
> 2019-02-20 12:51:21.532911 [NOTICE] switch_rtp.c:4799 Activating RTP audio
> ICE: 1cf2011d:u4NGJi1myDzrpRoh 10.99.88.122:55465
> 2019-02-20 12:51:21.532911 [DEBUG] switch_core_media.c:8731 Activating
> RTCP PORT 55465
> 2019-02-20 12:51:21.532911 [DEBUG] switch_rtp.c:4696 RTCP send rate is:
> 1000 and packet rate is: 20000 Remote Port: 55465
> 2019-02-20 12:51:21.532911 [INFO] switch_core_media.c:8742 Skipping RTCP
> ICE (Same as RTP)
>
> Jurijs
>
>
> On Wed, Feb 20, 2019 at 4:17 PM Giovanni Maruzzelli <gmaruzz at gmail.com>
> wrote:
>
>>
>> You stopped the log much before the call ends... Can you pastebin a
>> complete log, up to DESTROY CS ?
>>
>> Also, out of memory, do you have the things related to transcode media
>> and mixing media activated in config? Do FreeSWITCH find a media candidate
>> during ICE bargaining?
>>
>> If you pastebin a complete log maybe we can help more
>>
>> -giovanni
>>
>> On Wed, Feb 20, 2019 at 2:58 PM Jurijs Ivolga <jurijs.ivolga at gmail.com>
>> wrote:
>>
>>> Hi,
>>>
>>> I doubt that this is in network or configuration error. Everything are
>>> same between 1.6 & 1.8. I'm using docker and difference are only in Docker
>>> file where I build freeswitch. If I rebuild with old Docker file(1.6) then
>>> it works fine, but with new Docker file for 1.8, I have problems.
>>> Everything else seems to work, but DTLS calls...
>>>
>>> Logs:
>>> https://pastebin.com/g2FqHWWz
>>>
>>> Jurijs
>>>
>>>
>>> On Wed, Feb 20, 2019 at 2:47 PM Giovanni Maruzzelli <gmaruzz at gmail.com>
>>> wrote:
>>>
>>>> On Wed, Feb 20, 2019 at 12:59 PM Jurijs Ivolga <jurijs.ivolga at gmail.com>
>>>> wrote:
>>>>
>>>>> Hi,
>>>>>
>>>>> I'm trying to migrate from 1.6.20 to 1.8.5, but I'm facing a problem
>>>>> with DTLS calls(media_webrtc=true).
>>>>>
>>>>> Does anybody has same issues as me?
>>>>>
>>>>> I have following setup:
>>>>>
>>>>> RTP=>Freeswitch=>DTLS
>>>>>
>>>>> In 1.8.5, when Freeswitch receive 200 from DTLS endpoint, but it never
>>>>> sends it back to RTP endpoint.
>>>>>
>>>>
>>>> maybe you have a network problem, or a configuration problem network
>>>> related in 1.8 deployment?
>>>>
>>>> Also, for us to better understand: from fs cli, "sofia global siptrace
>>>> on", after this, copy and paste the ENTIRE debug level output from
>>>> beginning to end in a pastebin.
>>>>
>>>> Then put here the pastebin link
>>>>
>>>> -giovanni
>>>>
>>>>
>>>>
>>>>>
>>>>> Here some snippet from logs, where call stuck:
>>>>>
>>>>> 2019-02-20 09:50:07.368784 [INFO] switch_rtp.c:3832 Changing audio
>>>>> DTLS state from OFF to HANDSHAKE
>>>>> 2019-02-20 09:50:07.368784 [DEBUG] switch_core_media.c:8815
>>>>> sofia/internal/9999999999999999999 at sip.myapp.net Set 2833 dtmf send
>>>>> payload to 101
>>>>> 2019-02-20 09:50:07.368784 [DEBUG] switch_core_media.c:8822
>>>>> sofia/internal/9999999999999999999 at sip.myapp.net Set 2833 dtmf
>>>>> receive payload to 101
>>>>> 2019-02-20 09:50:07.368784 [DEBUG] switch_core_media.c:8845
>>>>> sofia/internal/9999999999999999999 at sip.myapp.net Set rtp dtmf delay
>>>>> to 40
>>>>>
>>>>> In 1.6.20 everything works as expected:
>>>>>
>>>>> 2019-02-20 11:12:40.129989 [DEBUG] switch_core_media.c:7180
>>>>> sofia/internal/9999999999999999999 at sip.myapp.net Set 2833 dtmf send
>>>>> payload to 101
>>>>> 2019-02-20 11:12:40.129989 [DEBUG] switch_core_media.c:7187
>>>>> sofia/internal/9999999999999999999 at sip.myapp.net Set 2833 dtmf
>>>>> receive payload to 101
>>>>> 2019-02-20 11:12:40.129989 [DEBUG] switch_core_media.c:7210
>>>>> sofia/internal/9999999999999999999 at sip.myapp.net Set rtp dtmf delay
>>>>> to 40
>>>>> 2019-02-20 11:12:40.129989 [NOTICE] sofia.c:8218 Channel
>>>>> [sofia/internal/9999999999999999999 at sip.myapp.net] has been answered
>>>>> 2019-02-20 11:12:40.129989 [DEBUG] switch_channel.c:3773
>>>>> (sofia/internal/9999999999999999999 at sip.myapp.net) Callstate Change
>>>>> RINGING -> ACTIVE
>>>>> 2019-02-20 11:12:40.129989 [DEBUG] switch_core_codec.c:248
>>>>> sofia/internal/1234567890 at sip.myapp.net Restore previous codec PCMA:8.
>>>>> 2019-02-20 11:12:40.129989 [DEBUG] switch_core_media.c:6861 Audio
>>>>> params are unchanged for sofia/internal/1234567890 at sip.myapp.net.
>>>>> 2019-02-20 11:12:40.129989 [DEBUG] mod_sofia.c:850 Local SDP
>>>>> sofia/internal/1234567890 at sip.myapp.net:
>>>>>
>>>>> Jurijs
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Services
>>>>> sales at freeswitch.com
>>>>> https://freeswitch.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> https://freeswitch.com/oss
>>>>> https://freeswitch.org/confluence
>>>>> https://cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> https://freeswitch.com
>>>>
>>>>
>>>>
>>>> --
>>>> Sincerely,
>>>>
>>>> Giovanni Maruzzelli
>>>> OpenTelecom.IT
>>>> cell: +39 347 266 56 18
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Services
>>>> sales at freeswitch.com
>>>> https://freeswitch.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> https://freeswitch.com/oss
>>>> https://freeswitch.org/confluence
>>>> https://cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> https://freeswitch.com
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Services
>>> sales at freeswitch.com
>>> https://freeswitch.com
>>>
>>> Official FreeSWITCH Sites
>>> https://freeswitch.com/oss
>>> https://freeswitch.org/confluence
>>> https://cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> https://freeswitch.com
>>
>>
>>
>> --
>> Sincerely,
>>
>> Giovanni Maruzzelli
>> OpenTelecom.IT
>> cell: +39 347 266 56 18
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From roman at dissauer.net  Wed Feb 20 18:34:46 2019
From: roman at dissauer.net (Roman Dissauer)
Date: Wed, 20 Feb 2019 19:34:46 +0100
Subject: [Freeswitch-users] setting contact domain in dialplan not
	working
In-Reply-To: <CADrq_yGu5U40LRUfRwU+zLdyxVN8H_4vbthWoetZURBFtPFpmA@mail.gmail.com>
References: <F363AB89-018F-4774-941D-74894521E932@dissauer.net>
 <CADrq_yGu5U40LRUfRwU+zLdyxVN8H_4vbthWoetZURBFtPFpmA@mail.gmail.com>
Message-ID: <3465EA6E-56C1-48CC-A649-A2A4CA010612@dissauer.net>

that explains a lot :)
I’ll go with the gateway param then...
Thank you!

> Am 20.02.2019 um 18:10 schrieb Mike Jerris <mike at freeswitch.com>:
> 
> There is no way to change that
> 
> On Tue, Feb 19, 2019 at 5:24 PM Roman Dissauer <roman at dissauer.net <mailto:roman at dissauer.net>> wrote:
> Is there a way to get the FS public IP replaced by a custom domain name in contact header using the dialplan?
> 
> tried:
> 
> <action application="export" data="sip_contact_host=test.domain.tld“/>         ...not working
> <action application=„set" data="sip_contact_host=test.domain.tld“/>         ...not working
> <action application="export" data="sip_invite_domain=test.domain.tld“/>         ...not working
> <action application=„set" data="sip_invite_domain=test.domain.tld“/>         ...not working
> <action application="export" data=„sip_contact_user=bla at test.domain.tld"/>         ...not working
> <action application=„set" data=„sip_invite_domain=bla at test.domain.tld“/>         ...not working
> 
> 
> only thing working is setting the parameter in gateway config, 
> <param name="contact-host" value=„test.domain.tld“/>
> 
> but this is not working for me because I need to do it dynamically in dialplan!
> 
> Any hint for me?
> 
> thank you!
> Roman
> 
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com <mailto:sales at freeswitch.com>
> https://freeswitch.com <https://freeswitch.com/>
> 
> Official FreeSWITCH Sites
> https://freeswitch.com/oss <https://freeswitch.com/oss>
> https://freeswitch.org/confluence <https://freeswitch.org/confluence>
> https://cluecon.com <https://cluecon.com/>
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> https://freeswitch.com <https://freeswitch.com/>_________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
> 
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com

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From koketsom at tluka.co.za  Thu Feb 21 04:10:51 2019
From: koketsom at tluka.co.za (Koketso Mabuse)
Date: Thu, 21 Feb 2019 06:10:51 +0200
Subject: [Freeswitch-users] Connecting endpoint to domain within Free
 Switch
In-Reply-To: <CAJVaC-uDQX9gxTLV8CpDkAd5T4yS4eqNO7ibAPRVFn14B9NF_w@mail.gmail.com>
References: <mailman.222.1550593495.1400.freeswitch-users@lists.freeswitch.org>
 <CAHtxdDfDSpcb72OGFrBzmWRro8OsZki2DU8fU=KZFX98_dpBuw@mail.gmail.com>
 <mailman.229.1550605882.1400.freeswitch-users@lists.freeswitch.org>
 <333d641f-1b8a-4c69-5dcc-bf276e01640d@tluka.co.za>
 <CAJVaC-uDQX9gxTLV8CpDkAd5T4yS4eqNO7ibAPRVFn14B9NF_w@mail.gmail.com>
Message-ID: <5cec2c30-5de6-7f8e-5c0b-925fee20b835@tluka.co.za>

Thank you for you advise and it worked like a charm...

to Recap: When using a 2nd domain endpoint looks like this:

Sip server / host: 2nd domain name/number

Proxy is goes to main domain.

On 2019/02/20 09:41, Alaadin Abdurrahman wrote:
> Hi,
> You need to setup your UA client to use your domain in the domain 
> filed and the host IP in the proxy filed
>
> Regards
> Ala
>
> On Wed, Feb 20, 2019 at 7:45 AM Koketso Mabuse <koketsom at tluka.co.za 
> <mailto:koketsom at tluka.co.za>> wrote:
>
>     Hi there
>
>     Scenario: Fresh installed FusionPBX:
>
>       * Default domain with local IP,
>       * created a 2nd domain as per tutorial,
>       * created extensions.
>           o tried to register extensions on 2nd domain failure
>               + changed domain on extion to point to 1st main domain,
>                 end point registers.
>
>     Issue: how do I create 2 extensions  eg: 1000 for domain 1 and
>     1000 for domain 2 and register them both on main domain?
>
>     Domain:mtcc
>
>     exmple: Username: 1000
>
>                    Password: pass123
>
>                    host: 192.168.1.130
>
>
>     Domain: tluka
>
>     Username: 1000
>
>                    Password: pass123
>
>                    host: 192.168.1.130
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Services
>     sales at freeswitch.com <mailto:sales at freeswitch.com>
>     https://freeswitch.com
>
>     Official FreeSWITCH Sites
>     https://freeswitch.com/oss
>     https://freeswitch.org/confluence
>     https://cluecon.com
>
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     https://freeswitch.com
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From jurijs.ivolga at gmail.com  Thu Feb 21 12:19:29 2019
From: jurijs.ivolga at gmail.com (Jurijs Ivolga)
Date: Thu, 21 Feb 2019 14:19:29 +0200
Subject: [Freeswitch-users] 1.8 DTLS calls
In-Reply-To: <CALXCt0qXU1P5PObbW_JdvCzH8pWhW7+J0db8tS9OuZ8LqcBrgA@mail.gmail.com>
References: <CAOrA2UZK=nNhSnm6CR8xieRbrsXHUoFVQnq7sMXT5OxCcm2n0g@mail.gmail.com>
 <CALXCt0rL_D4jZq0HCCCssg1vf4QdfGe7U+RxBVBwe3Xc7Kd1Sw@mail.gmail.com>
 <CAOrA2UbTpHA3ZAG1WC=DdgUGJ-0A9Fy+BogOs13avR+Zf=4svA@mail.gmail.com>
 <CALXCt0rRvfd_3dAJTrekC7pHqqUK1dfC_Z9H2Z_sUutuN0OzEw@mail.gmail.com>
 <CAOrA2UYm3WnWqsMRrw+Tae3CvkiCECbLq9NFwK9uwEyCF2PsKw@mail.gmail.com>
 <CALXCt0qXU1P5PObbW_JdvCzH8pWhW7+J0db8tS9OuZ8LqcBrgA@mail.gmail.com>
Message-ID: <CAOrA2UaaiThWs+6TFYCtDf+bMSJp_KhdWboLXdwSJRdfwneX2Q@mail.gmail.com>

Hi Giovanni,

Thank you for tips!

Lets see if I'll be able to fix this. ;)

Jurijs


On Wed, Feb 20, 2019 at 7:35 PM Giovanni Maruzzelli <gmaruzz at gmail.com>
wrote:

> That seems to be a nat problem, or an ice misconfiguration, like using
> wrong stun (that results in a nat problem, again).
>
> I assume all is equal in server.
>
> Check to see if you can give different parameters to clients, maybe
> between releases Freeswitch changed in subtle modes how it react to subtle
> client parameters. I would check all nat/ice/stun client parameters and
> experiment with them...
>
> Just a possible line of investigation, no warranties ;)
>
> -giovanni
>
>
>
> On Wed, Feb 20, 2019, 16:18 Jurijs Ivolga <jurijs.ivolga at gmail.com wrote:
>
>> Hi Giovanni,
>>
>> Thank you for tips!
>>
>> There is nothing happening after what I sent, so call just stuck there.
>> Only if I hangup on any side.
>>
>> Only I can to add that after 30 seconds I just get this line:
>>
>> 2019-02-20 14:22:07.195060 [ERR] switch_rtp.c:834 No audio stun for a
>> long time!
>>
>> If I hangup calls on both side, call still shows in freeswitch as active
>> if I do show calls.
>>
>> I believe I have everything needed for trans-coding in config files,
>> cause calls where no DTLS involved but with transcoding, calls works fine.
>> In that case I have RTP(ALAW) => Freeswitch => SRTP(OPUS)
>>
>> Regarding ICE:
>>
>> 2019-02-20 12:51:21.532911 [INFO] switch_core_media.c:8690 Activating
>> Audio ICE
>> 2019-02-20 12:51:21.532911 [NOTICE] switch_rtp.c:4799 Activating RTP
>> audio ICE: 1cf2011d:u4NGJi1myDzrpRoh 10.99.88.122:55465
>> 2019-02-20 12:51:21.532911 [DEBUG] switch_core_media.c:8731 Activating
>> RTCP PORT 55465
>> 2019-02-20 12:51:21.532911 [DEBUG] switch_rtp.c:4696 RTCP send rate is:
>> 1000 and packet rate is: 20000 Remote Port: 55465
>> 2019-02-20 12:51:21.532911 [INFO] switch_core_media.c:8742 Skipping RTCP
>> ICE (Same as RTP)
>>
>> Jurijs
>>
>>
>> On Wed, Feb 20, 2019 at 4:17 PM Giovanni Maruzzelli <gmaruzz at gmail.com>
>> wrote:
>>
>>>
>>> You stopped the log much before the call ends... Can you pastebin a
>>> complete log, up to DESTROY CS ?
>>>
>>> Also, out of memory, do you have the things related to transcode media
>>> and mixing media activated in config? Do FreeSWITCH find a media candidate
>>> during ICE bargaining?
>>>
>>> If you pastebin a complete log maybe we can help more
>>>
>>> -giovanni
>>>
>>> On Wed, Feb 20, 2019 at 2:58 PM Jurijs Ivolga <jurijs.ivolga at gmail.com>
>>> wrote:
>>>
>>>> Hi,
>>>>
>>>> I doubt that this is in network or configuration error. Everything are
>>>> same between 1.6 & 1.8. I'm using docker and difference are only in Docker
>>>> file where I build freeswitch. If I rebuild with old Docker file(1.6) then
>>>> it works fine, but with new Docker file for 1.8, I have problems.
>>>> Everything else seems to work, but DTLS calls...
>>>>
>>>> Logs:
>>>> https://pastebin.com/g2FqHWWz
>>>>
>>>> Jurijs
>>>>
>>>>
>>>> On Wed, Feb 20, 2019 at 2:47 PM Giovanni Maruzzelli <gmaruzz at gmail.com>
>>>> wrote:
>>>>
>>>>> On Wed, Feb 20, 2019 at 12:59 PM Jurijs Ivolga <
>>>>> jurijs.ivolga at gmail.com> wrote:
>>>>>
>>>>>> Hi,
>>>>>>
>>>>>> I'm trying to migrate from 1.6.20 to 1.8.5, but I'm facing a problem
>>>>>> with DTLS calls(media_webrtc=true).
>>>>>>
>>>>>> Does anybody has same issues as me?
>>>>>>
>>>>>> I have following setup:
>>>>>>
>>>>>> RTP=>Freeswitch=>DTLS
>>>>>>
>>>>>> In 1.8.5, when Freeswitch receive 200 from DTLS endpoint, but it
>>>>>> never sends it back to RTP endpoint.
>>>>>>
>>>>>
>>>>> maybe you have a network problem, or a configuration problem network
>>>>> related in 1.8 deployment?
>>>>>
>>>>> Also, for us to better understand: from fs cli, "sofia global siptrace
>>>>> on", after this, copy and paste the ENTIRE debug level output from
>>>>> beginning to end in a pastebin.
>>>>>
>>>>> Then put here the pastebin link
>>>>>
>>>>> -giovanni
>>>>>
>>>>>
>>>>>
>>>>>>
>>>>>> Here some snippet from logs, where call stuck:
>>>>>>
>>>>>> 2019-02-20 09:50:07.368784 [INFO] switch_rtp.c:3832 Changing audio
>>>>>> DTLS state from OFF to HANDSHAKE
>>>>>> 2019-02-20 09:50:07.368784 [DEBUG] switch_core_media.c:8815
>>>>>> sofia/internal/9999999999999999999 at sip.myapp.net Set 2833 dtmf send
>>>>>> payload to 101
>>>>>> 2019-02-20 09:50:07.368784 [DEBUG] switch_core_media.c:8822
>>>>>> sofia/internal/9999999999999999999 at sip.myapp.net Set 2833 dtmf
>>>>>> receive payload to 101
>>>>>> 2019-02-20 09:50:07.368784 [DEBUG] switch_core_media.c:8845
>>>>>> sofia/internal/9999999999999999999 at sip.myapp.net Set rtp dtmf delay
>>>>>> to 40
>>>>>>
>>>>>> In 1.6.20 everything works as expected:
>>>>>>
>>>>>> 2019-02-20 11:12:40.129989 [DEBUG] switch_core_media.c:7180
>>>>>> sofia/internal/9999999999999999999 at sip.myapp.net Set 2833 dtmf send
>>>>>> payload to 101
>>>>>> 2019-02-20 11:12:40.129989 [DEBUG] switch_core_media.c:7187
>>>>>> sofia/internal/9999999999999999999 at sip.myapp.net Set 2833 dtmf
>>>>>> receive payload to 101
>>>>>> 2019-02-20 11:12:40.129989 [DEBUG] switch_core_media.c:7210
>>>>>> sofia/internal/9999999999999999999 at sip.myapp.net Set rtp dtmf delay
>>>>>> to 40
>>>>>> 2019-02-20 11:12:40.129989 [NOTICE] sofia.c:8218 Channel
>>>>>> [sofia/internal/9999999999999999999 at sip.myapp.net] has been answered
>>>>>> 2019-02-20 11:12:40.129989 [DEBUG] switch_channel.c:3773
>>>>>> (sofia/internal/9999999999999999999 at sip.myapp.net) Callstate Change
>>>>>> RINGING -> ACTIVE
>>>>>> 2019-02-20 11:12:40.129989 [DEBUG] switch_core_codec.c:248
>>>>>> sofia/internal/1234567890 at sip.myapp.net Restore previous codec
>>>>>> PCMA:8.
>>>>>> 2019-02-20 11:12:40.129989 [DEBUG] switch_core_media.c:6861 Audio
>>>>>> params are unchanged for sofia/internal/1234567890 at sip.myapp.net.
>>>>>> 2019-02-20 11:12:40.129989 [DEBUG] mod_sofia.c:850 Local SDP
>>>>>> sofia/internal/1234567890 at sip.myapp.net:
>>>>>>
>>>>>> Jurijs
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Services
>>>>>> sales at freeswitch.com
>>>>>> https://freeswitch.com
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> https://freeswitch.com/oss
>>>>>> https://freeswitch.org/confluence
>>>>>> https://cluecon.com
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> https://freeswitch.com
>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> Sincerely,
>>>>>
>>>>> Giovanni Maruzzelli
>>>>> OpenTelecom.IT
>>>>> cell: +39 347 266 56 18
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Services
>>>>> sales at freeswitch.com
>>>>> https://freeswitch.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> https://freeswitch.com/oss
>>>>> https://freeswitch.org/confluence
>>>>> https://cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> https://freeswitch.com
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Services
>>>> sales at freeswitch.com
>>>> https://freeswitch.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> https://freeswitch.com/oss
>>>> https://freeswitch.org/confluence
>>>> https://cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> https://freeswitch.com
>>>
>>>
>>>
>>> --
>>> Sincerely,
>>>
>>> Giovanni Maruzzelli
>>> OpenTelecom.IT
>>> cell: +39 347 266 56 18
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Services
>>> sales at freeswitch.com
>>> https://freeswitch.com
>>>
>>> Official FreeSWITCH Sites
>>> https://freeswitch.com/oss
>>> https://freeswitch.org/confluence
>>> https://cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> https://freeswitch.com
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From alaadin.abd at gmail.com  Thu Feb 21 06:58:50 2019
From: alaadin.abd at gmail.com (Alaadin Abdurrahman)
Date: Thu, 21 Feb 2019 08:58:50 +0200
Subject: [Freeswitch-users] Connecting endpoint to domain within Free
	Switch
In-Reply-To: <5cec2c30-5de6-7f8e-5c0b-925fee20b835@tluka.co.za>
References: <mailman.222.1550593495.1400.freeswitch-users@lists.freeswitch.org>
 <CAHtxdDfDSpcb72OGFrBzmWRro8OsZki2DU8fU=KZFX98_dpBuw@mail.gmail.com>
 <mailman.229.1550605882.1400.freeswitch-users@lists.freeswitch.org>
 <333d641f-1b8a-4c69-5dcc-bf276e01640d@tluka.co.za>
 <CAJVaC-uDQX9gxTLV8CpDkAd5T4yS4eqNO7ibAPRVFn14B9NF_w@mail.gmail.com>
 <5cec2c30-5de6-7f8e-5c0b-925fee20b835@tluka.co.za>
Message-ID: <CAJVaC-vq8KUkgqrCQwk1qbhZgO9EraTZPLf2M9JeZ0YL0-70LA@mail.gmail.com>

Yes exactly, domain in freeswitch has nothing to do with internet domain
names it is just to refer to a tenant in
multi tenant environment.

Regards
Ala

On Thu, Feb 21, 2019 at 6:24 AM Koketso Mabuse <koketsom at tluka.co.za> wrote:

> Thank you for you advise and it worked like a charm...
>
> to Recap: When using a 2nd domain endpoint looks like this:
>
> Sip server / host: 2nd domain name/number
>
> Proxy is goes to main domain.
> On 2019/02/20 09:41, Alaadin Abdurrahman wrote:
>
> Hi,
> You need to setup your UA client to use your domain in the domain filed
> and the host IP in the proxy filed
>
> Regards
> Ala
>
> On Wed, Feb 20, 2019 at 7:45 AM Koketso Mabuse <koketsom at tluka.co.za>
> wrote:
>
>> Hi there
>>
>> Scenario: Fresh installed FusionPBX:
>>
>>    - Default domain with local IP,
>>    - created a 2nd domain as per tutorial,
>>    - created extensions.
>>       - tried to register extensions on 2nd domain failure
>>          - changed domain on extion to point to 1st main domain, end
>>          point registers.
>>
>> Issue: how do I create 2 extensions  eg: 1000 for domain 1 and 1000 for
>> domain 2 and register them both on main domain?
>>
>> Domain:mtcc
>>
>> exmple: Username: 1000
>>
>>                Password: pass123
>>
>>                host: 192.168.1.130
>>
>>
>> Domain: tluka
>>
>> Username: 1000
>>
>>                Password: pass123
>>
>>                host: 192.168.1.130
>> _________________________________________________________________________
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Servicessales at freeswitch.comhttps://freeswitch.com
>
> Official FreeSWITCH Siteshttps://freeswitch.com/osshttps://freeswitch.org/confluencehttps://cluecon.com
>
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttps://freeswitch.com
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From rajil.s at gmail.com  Fri Feb 22 02:05:48 2019
From: rajil.s at gmail.com (Rajil Saraswat)
Date: Thu, 21 Feb 2019 20:05:48 -0600
Subject: [Freeswitch-users] Zoiper goes crazy with registrations RPORT bug?
Message-ID: <be1f4cc8-37f0-5af2-abf8-6f04ae3d1314@gmail.com>

   Hello,

I have a Zoiper android client with Freeswitch server version 
1.8.4-5-749a6e108b~64bit. If i select RPORT option in Zoiper along with 
TLS/TCP transport protocol, i get hundreds for registration requests per 
minute. Zoiper customer support thinks that Freeswitch is at fault here. 
The message i received from them is below. Is there any fix for this?

------------

The issue in your case is that the topmost Via header does not have the 
mandatory received parameter.

According to RFC 3581 (An Extension to the Session Initiation Protocol 
(SIP) for Symmetric Response Routing) Section 4. Server Behavior:

"When a server compliant to this specification (which can be a proxy
or UAS) receives a request, it examines the topmost Via header field
value. If this Via header field value contains an "rport" parameter
with no value, it MUST set the value of the parameter to the source
port of the request. This is analogous to the way in which a server
will insert the "received" parameter into the topmost Via header
field value. * In fact, the server MUST insert a "received" parameter*
containing the source IP address that the request came from, even if
it is identical to the value of the "sent-by" component. Note that
this processing takes place independent of the transport protocol."

This is a server side issue and it is related most likely to the version 
of FreeSWITCH that you are using.

----------

Thanks



From vbvbrj at gmail.com  Fri Feb 22 11:08:15 2019
From: vbvbrj at gmail.com (Mimiko)
Date: Fri, 22 Feb 2019 13:08:15 +0200
Subject: [Freeswitch-users] Change SIP Contact header on b-leg
In-Reply-To: <5513E6FD.7010005@wirelessmundi.com>
References: <HE1PR06MB11309B44ECF52155228F7A7B8D080@HE1PR06MB1130.eurprd06.prod.outlook.com>
 <5513E6FD.7010005@wirelessmundi.com>
Message-ID: <ddb3a1cc-e797-73a2-c3f3-5451e8dab0ae@gmail.com>

On 26.03.2015 13:01, Antonio Silva wrote:
> You can force the contact extension in the gateway... instead of having the "gw+ss"
> 
> The parameter is
> 
> extension = your_provider_expected_extension
> extension-in-contact  = true

This is old and I'm facing such bad devices on provider.

I set:

<param name="extension" value="auto_to_user"/>
<param name="extension-in-contact" value="true"/>

But then other problem arise. When provider sends a call to me the destination_number is populated with username from contact instead of from 
${sip_to_user}.

When I set
<param name="extension-in-contact" value="false"/> then destination_number is populated correctly. How can I change contact header another way?


From ynasida at gmail.com  Fri Feb 22 16:59:39 2019
From: ynasida at gmail.com (Yuriy Nasida)
Date: Fri, 22 Feb 2019 19:59:39 +0300
Subject: [Freeswitch-users] where g729 fs-latest-installer script now ?
In-Reply-To: <0f4adbf7-4df1-de06-3e3b-e8c7f653712e@bohboh.info>
References: <CADihe3K9Cvhjtpdox-QXO-KUSHew-aey16zirECudPUQ6kNYpA@mail.gmail.com>
 <CADihe3+UAjHwHJyz2+Co1+oDnvu=79PhBNfVZM5jh+Vy3OOzNA@mail.gmail.com>
 <CADihe3KvzFs5X62R1EXjhY7k7+MEGjM63u7woyL-BXZaA-BZBw@mail.gmail.com>
 <CADihe3LQwqFQ9vLqU34X_taK0HEf2etp=3CO1S9YkQFCy6wSag@mail.gmail.com>
 <4af1376d-29ba-ea47-c952-c4883b7dd654@wirelessmundi.com>
 <CADihe3K-yyXFF8=w_7CLS3thpV9mbHnZWxbwR3TUOVy_k_nwsA@mail.gmail.com>
 <4d7501d4c87d$978a56c0$c69f0440$@freeswitch.org>
 <918f5e6d-bafb-3e1d-bd2d-26648ca948a7@bohboh.info>
 <007e01d4c88c$c717cf30$55476d90$@freeswitch.org>
 <0f4adbf7-4df1-de06-3e3b-e8c7f653712e@bohboh.info>
Message-ID: <CADihe3KOsGs36Bayh8oZKUivJRcnrdZSWsxDqGmC9ifcLK04bg@mail.gmail.com>

Well. I tried Antonio's method and have installed mod_com_g729 and
freeswitch_license_server using deb files.
But... I have got same result.  FS doesn't listen 8021 port in case I have
enabled mod_com_g729 in modules.conf.xml

Also I noted that systemd starts two freeswitch processes/

# ps ax | grep free
22367 ?        Ss     0:04 /usr/sbin/freeswitch-license-server
22776 ?        Ss     0:00 /usr/bin/freeswitch -u freeswitch -g freeswitch
-ncwait -nonat
22777 ?        S<sl   0:01 /usr/bin/freeswitch -u freeswitch -g freeswitch
-ncwait -nonat

# netstat -nlp | grep free
tcp        0      0 0.0.0.0:2855            0.0.0.0:*               LISTEN
    22808/freeswitch
tcp        0      0 0.0.0.0:2856            0.0.0.0:*               LISTEN
    22808/freeswitch
tcp        0      0 my_ip:5060      0.0.0.0:*               LISTEN
22808/freeswitch
tcp        0      0 my_ip:5060      0.0.0.0:*               LISTEN
22808/freeswitch
udp        0      0 my_ip:5060      0.0.0.0:*
 22808/freeswitch
udp        0      0 my_ip:5060      0.0.0.0:*
 22808/freeswitch

FS doesn't listen 8021 like you see.

All things working fine I case I will disable mod_com_g729 in
modules.conf.xml

Please advice

On Tue, 19 Feb 2019 at 23:20, Social Boh <social at bohboh.info> wrote:

> So you don't update MPL 1.1 with MPL. 2.0 because the 1.1 is not
> compatible with GPL while 2.0 is compatible with GPL.
> Regards
>
> ---
> I'm SoCIaL, MayBe
>
>
> El 19/02/2019 a las 14:53, Ken Rice escribió:
>
> I was pretty clear on the my statements about taking funding away from the
> FreeSWITCH Project and the devs by doing such thing
>
>
>
> Also the bcg module is GPLv2 license violation waiting to be noticed… you
> cant combine MPL and GPL code and have a resulting MPL codebase. Its not
> allowed by the GPL. So you you are also ripping off the GPL coder.
>
>
>
> Now you are polluting a MPL code base with the GPL violating the Copyright
> holder of the MPL’s software rights to determine how his code is licensed.
>
>
>
> This happens all the time. People think oh this code is free, and yes most
> of the code is free, but the devs still have mortgages, children to feed,
> computers and related hardware to keep updated so that they can continue to
> support that hardware that everyone is using currently. Website hosting for
> something like FreeSWITCH isnt free. It requires over 1/2 a rack of colo’d
> equipment to keep the website, mailing list, build servers, git repos, wiki
> etc etc etc going.
>
>
>
> $10 for a channel is a pretty cheap way to support that seeing as that
> most of the time you don’t even need G729
>
>
>
> *From:* Social Boh <social at bohboh.info> <social at bohboh.info>
> *Sent:* Tuesday, February 19, 2019 1:03 PM
> *To:* Ken Rice <krice at freeswitch.org> <krice at freeswitch.org>
> *Subject:* Re: [Freeswitch-users] where g729 fs-latest-installer script
> now ?
>
>
>
> Can you explain me why I can't use the mod_bcg729 with FreeSWITCH?
>
> Thank you
>
> ---
>
> I'm SoCIaL, MayBe
>
> El 19/02/2019 a las 13:04, Ken Rice escribió:
>
> My statement wasn’t to you, but to the people that continually want to
> pirate someone elses G729 code and refuse to support the projects
> developers.
>
>
>
> Its one of those things that grinds my gears.
>
>
>
> K
>
>
>
> *From:* FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org>
> <freeswitch-users-bounces at lists.freeswitch.org> *On Behalf Of *Yuriy
> Nasida
> *Sent:* Tuesday, February 19, 2019 6:47 AM
> *To:* António Silva <asilva at wirelessmundi.com> <asilva at wirelessmundi.com>
> *Cc:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] where g729 fs-latest-installer script
> now ?
>
>
>
> Ken, not sure what you talking about. I already bought a lot of g729
> licenses for my servers and I use them a lot of years. But now I can't use
> them normally at Deb 9 + fs1.8 + new g729 install script because of issues
> I described.
>
>
>
> Thanks a lot Antonio! I will try this way.
>
>
>
> On Tue, 19 Feb 2019 at 11:41, António Silva <asilva at wirelessmundi.com>
> wrote:
>
> try using the .deb and installed then manuall.:
>
> Get the packages from here:
> http://files.freeswitch.org/repo/deb/freeswitch-1.8/pool/main/f/
>
> Installed them manually:
>
> dpkg -x freeswitch-mod-com-g729_1.8.5~7~14d59c97e0~jessie_amd64.deb /
> dpkg -x freeswitch-license-server_0.0.1~42~731842be7d~jessie_amd64.deb /
>
>
>
> I don't know if the installer will remain, from the jira
> https://freeswitch.org/jira/browse/FS-11641 i got the impression that now
> the solution is to use the debian packages.
>
>
>
> On 13/02/2019 18:10, Yuriy Nasida wrote:
>
> I have installed freeswitch-license-server  using
> fs-201902072050-installer  and got issue below.
>
> FS doesn't listen port 8021 and I can't connect to it and run any command
> using fs_cli.
>
> All works fine in case I kill process freeswitch-license-server.
>
>
>
> Can anybody comment please all these very weird things with commercial
> g729 at FS ?
>
> Thanks.
>
>
>
> On Wed, 13 Feb 2019 at 13:36, Yuriy Nasida <ynasida at gmail.com> wrote:
>
> Well. Somebody updated http://files.freeswitch.org/g729/ and now it
> has fs-latest-installer. Thanks a lot whoever you are!!!
>
> But... Now it has not install instructions  :)
>
>
>
> On Wed, 13 Feb 2019 at 12:25, Yuriy Nasida <ynasida at gmail.com> wrote:
>
> Um... Nobody uses g729 licences anymore ? :)
>
>
>
> On Tue, 12 Feb 2019 at 20:07, Yuriy Nasida <ynasida at gmail.com> wrote:
>
> Hi guys,
>
>
>
> Does anybody know why site http://files.freeswitch.org/g729/ contain only
> install instrunction now ?  Where fs-latest-installer-v1.6 script or
> something like this ?
>
>
>
> Please advice.
>
>
>
> _________________________________________________________________________
>
> Professional FreeSWITCH Services
>
> sales at freeswitch.com
>
> https://freeswitch.com
>
>
>
> Official FreeSWITCH Sites
>
> https://freeswitch.com/oss
>
> https://freeswitch.org/confluence
>
> https://cluecon.com
>
>
>
> FreeSWITCH-users mailing list
>
> FreeSWITCH-users at lists.freeswitch.org
>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>
> https://freeswitch.com
>
> --
>
> Saludos / Regards / Cumprimentos
>
> António Silva
>
>
>
> _________________________________________________________________________
>
> Professional FreeSWITCH Services
>
> sales at freeswitch.com
>
> https://freeswitch.com
>
>
>
> Official FreeSWITCH Sites
>
> https://freeswitch.com/oss
>
> https://freeswitch.org/confluence
>
> https://cluecon.com
>
>
>
> FreeSWITCH-users mailing list
>
> FreeSWITCH-users at lists.freeswitch.org
>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>
> https://freeswitch.com
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From joel at textplus.com  Fri Feb 22 22:13:37 2019
From: joel at textplus.com (Joel Serrano)
Date: Fri, 22 Feb 2019 14:13:37 -0800
Subject: [Freeswitch-users] where g729 fs-latest-installer script now ?
In-Reply-To: <CADihe3KOsGs36Bayh8oZKUivJRcnrdZSWsxDqGmC9ifcLK04bg@mail.gmail.com>
References: <CADihe3K9Cvhjtpdox-QXO-KUSHew-aey16zirECudPUQ6kNYpA@mail.gmail.com>
 <CADihe3+UAjHwHJyz2+Co1+oDnvu=79PhBNfVZM5jh+Vy3OOzNA@mail.gmail.com>
 <CADihe3KvzFs5X62R1EXjhY7k7+MEGjM63u7woyL-BXZaA-BZBw@mail.gmail.com>
 <CADihe3LQwqFQ9vLqU34X_taK0HEf2etp=3CO1S9YkQFCy6wSag@mail.gmail.com>
 <4af1376d-29ba-ea47-c952-c4883b7dd654@wirelessmundi.com>
 <CADihe3K-yyXFF8=w_7CLS3thpV9mbHnZWxbwR3TUOVy_k_nwsA@mail.gmail.com>
 <4d7501d4c87d$978a56c0$c69f0440$@freeswitch.org>
 <918f5e6d-bafb-3e1d-bd2d-26648ca948a7@bohboh.info>
 <007e01d4c88c$c717cf30$55476d90$@freeswitch.org>
 <0f4adbf7-4df1-de06-3e3b-e8c7f653712e@bohboh.info>
 <CADihe3KOsGs36Bayh8oZKUivJRcnrdZSWsxDqGmC9ifcLK04bg@mail.gmail.com>
Message-ID: <CAMtXxQk9X0Juu4Y11r2M5YGHu4SuBoFf8-N7oF4zBBzePr5fcg@mail.gmail.com>

When using license server you have to load mod_com_g729 not mod_g729.

mod_bcg729 is a completely different module totally independent from the FS
officially supported g729 modules and the license server.

On Fri, Feb 22, 2019 at 12:34 Yuriy Nasida <ynasida at gmail.com> wrote:

> Well. I tried Antonio's method and have installed mod_com_g729 and
> freeswitch_license_server using deb files.
> But... I have got same result.  FS doesn't listen 8021 port in case I have
> enabled mod_com_g729 in modules.conf.xml
>
> Also I noted that systemd starts two freeswitch processes/
>
> # ps ax | grep free
> 22367 ?        Ss     0:04 /usr/sbin/freeswitch-license-server
> 22776 ?        Ss     0:00 /usr/bin/freeswitch -u freeswitch -g freeswitch
> -ncwait -nonat
> 22777 ?        S<sl   0:01 /usr/bin/freeswitch -u freeswitch -g freeswitch
> -ncwait -nonat
>
> # netstat -nlp | grep free
> tcp        0      0 0.0.0.0:2855            0.0.0.0:*
>  LISTEN      22808/freeswitch
> tcp        0      0 0.0.0.0:2856            0.0.0.0:*
>  LISTEN      22808/freeswitch
> tcp        0      0 my_ip:5060      0.0.0.0:*               LISTEN
> 22808/freeswitch
> tcp        0      0 my_ip:5060      0.0.0.0:*               LISTEN
> 22808/freeswitch
> udp        0      0 my_ip:5060      0.0.0.0:*
>  22808/freeswitch
> udp        0      0 my_ip:5060      0.0.0.0:*
>  22808/freeswitch
>
> FS doesn't listen 8021 like you see.
>
> All things working fine I case I will disable mod_com_g729 in
> modules.conf.xml
>
> Please advice
>
> On Tue, 19 Feb 2019 at 23:20, Social Boh <social at bohboh.info> wrote:
>
>> So you don't update MPL 1.1 with MPL. 2.0 because the 1.1 is not
>> compatible with GPL while 2.0 is compatible with GPL.
>> Regards
>>
>> ---
>> I'm SoCIaL, MayBe
>>
>>
>> El 19/02/2019 a las 14:53, Ken Rice escribió:
>>
>> I was pretty clear on the my statements about taking funding away from
>> the FreeSWITCH Project and the devs by doing such thing
>>
>>
>>
>> Also the bcg module is GPLv2 license violation waiting to be noticed… you
>> cant combine MPL and GPL code and have a resulting MPL codebase. Its not
>> allowed by the GPL. So you you are also ripping off the GPL coder.
>>
>>
>>
>> Now you are polluting a MPL code base with the GPL violating the
>> Copyright holder of the MPL’s software rights to determine how his code is
>> licensed.
>>
>>
>>
>> This happens all the time. People think oh this code is free, and yes
>> most of the code is free, but the devs still have mortgages, children to
>> feed, computers and related hardware to keep updated so that they can
>> continue to support that hardware that everyone is using currently. Website
>> hosting for something like FreeSWITCH isnt free. It requires over 1/2 a
>> rack of colo’d equipment to keep the website, mailing list, build servers,
>> git repos, wiki etc etc etc going.
>>
>>
>>
>> $10 for a channel is a pretty cheap way to support that seeing as that
>> most of the time you don’t even need G729
>>
>>
>>
>> *From:* Social Boh <social at bohboh.info> <social at bohboh.info>
>> *Sent:* Tuesday, February 19, 2019 1:03 PM
>> *To:* Ken Rice <krice at freeswitch.org> <krice at freeswitch.org>
>> *Subject:* Re: [Freeswitch-users] where g729 fs-latest-installer script
>> now ?
>>
>>
>>
>> Can you explain me why I can't use the mod_bcg729 with FreeSWITCH?
>>
>> Thank you
>>
>> ---
>>
>> I'm SoCIaL, MayBe
>>
>> El 19/02/2019 a las 13:04, Ken Rice escribió:
>>
>> My statement wasn’t to you, but to the people that continually want to
>> pirate someone elses G729 code and refuse to support the projects
>> developers.
>>
>>
>>
>> Its one of those things that grinds my gears.
>>
>>
>>
>> K
>>
>>
>>
>> *From:* FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org>
>> <freeswitch-users-bounces at lists.freeswitch.org> *On Behalf Of *Yuriy
>> Nasida
>> *Sent:* Tuesday, February 19, 2019 6:47 AM
>> *To:* António Silva <asilva at wirelessmundi.com> <asilva at wirelessmundi.com>
>> *Cc:* freeswitch-users at lists.freeswitch.org
>> *Subject:* Re: [Freeswitch-users] where g729 fs-latest-installer script
>> now ?
>>
>>
>>
>> Ken, not sure what you talking about. I already bought a lot of g729
>> licenses for my servers and I use them a lot of years. But now I can't use
>> them normally at Deb 9 + fs1.8 + new g729 install script because of issues
>> I described.
>>
>>
>>
>> Thanks a lot Antonio! I will try this way.
>>
>>
>>
>> On Tue, 19 Feb 2019 at 11:41, António Silva <asilva at wirelessmundi.com>
>> wrote:
>>
>> try using the .deb and installed then manuall.:
>>
>> Get the packages from here:
>> http://files.freeswitch.org/repo/deb/freeswitch-1.8/pool/main/f/
>>
>> Installed them manually:
>>
>> dpkg -x freeswitch-mod-com-g729_1.8.5~7~14d59c97e0~jessie_amd64.deb /
>> dpkg -x freeswitch-license-server_0.0.1~42~731842be7d~jessie_amd64.deb /
>>
>>
>>
>> I don't know if the installer will remain, from the jira
>> https://freeswitch.org/jira/browse/FS-11641 i got the impression that
>> now the solution is to use the debian packages.
>>
>>
>>
>> On 13/02/2019 18:10, Yuriy Nasida wrote:
>>
>> I have installed freeswitch-license-server  using
>> fs-201902072050-installer  and got issue below.
>>
>> FS doesn't listen port 8021 and I can't connect to it and run any command
>> using fs_cli.
>>
>> All works fine in case I kill process freeswitch-license-server.
>>
>>
>>
>> Can anybody comment please all these very weird things with commercial
>> g729 at FS ?
>>
>> Thanks.
>>
>>
>>
>> On Wed, 13 Feb 2019 at 13:36, Yuriy Nasida <ynasida at gmail.com> wrote:
>>
>> Well. Somebody updated http://files.freeswitch.org/g729/ and now it
>> has fs-latest-installer. Thanks a lot whoever you are!!!
>>
>> But... Now it has not install instructions  :)
>>
>>
>>
>> On Wed, 13 Feb 2019 at 12:25, Yuriy Nasida <ynasida at gmail.com> wrote:
>>
>> Um... Nobody uses g729 licences anymore ? :)
>>
>>
>>
>> On Tue, 12 Feb 2019 at 20:07, Yuriy Nasida <ynasida at gmail.com> wrote:
>>
>> Hi guys,
>>
>>
>>
>> Does anybody know why site http://files.freeswitch.org/g729/ contain
>> only install instrunction now ?  Where fs-latest-installer-v1.6 script or
>> something like this ?
>>
>>
>>
>> Please advice.
>>
>>
>>
>> _________________________________________________________________________
>>
>> Professional FreeSWITCH Services
>>
>> sales at freeswitch.com
>>
>> https://freeswitch.com
>>
>>
>>
>> Official FreeSWITCH Sites
>>
>> https://freeswitch.com/oss
>>
>> https://freeswitch.org/confluence
>>
>> https://cluecon.com
>>
>>
>>
>> FreeSWITCH-users mailing list
>>
>> FreeSWITCH-users at lists.freeswitch.org
>>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>
>> https://freeswitch.com
>>
>> --
>>
>> Saludos / Regards / Cumprimentos
>>
>> António Silva
>>
>>
>>
>> _________________________________________________________________________
>>
>> Professional FreeSWITCH Services
>>
>> sales at freeswitch.com
>>
>> https://freeswitch.com
>>
>>
>>
>> Official FreeSWITCH Sites
>>
>> https://freeswitch.com/oss
>>
>> https://freeswitch.org/confluence
>>
>> https://cluecon.com
>>
>>
>>
>> FreeSWITCH-users mailing list
>>
>> FreeSWITCH-users at lists.freeswitch.org
>>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>
>> https://freeswitch.com
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From krice at freeswitch.org  Sat Feb 23 02:24:01 2019
From: krice at freeswitch.org (Ken Rice)
Date: Fri, 22 Feb 2019 20:24:01 -0600
Subject: [Freeswitch-users] where g729 fs-latest-installer script now ?
In-Reply-To: <CADihe3KOsGs36Bayh8oZKUivJRcnrdZSWsxDqGmC9ifcLK04bg@mail.gmail.com>
References: <CADihe3K9Cvhjtpdox-QXO-KUSHew-aey16zirECudPUQ6kNYpA@mail.gmail.com>
 <CADihe3+UAjHwHJyz2+Co1+oDnvu=79PhBNfVZM5jh+Vy3OOzNA@mail.gmail.com>
 <CADihe3KvzFs5X62R1EXjhY7k7+MEGjM63u7woyL-BXZaA-BZBw@mail.gmail.com>
 <CADihe3LQwqFQ9vLqU34X_taK0HEf2etp=3CO1S9YkQFCy6wSag@mail.gmail.com>
 <4af1376d-29ba-ea47-c952-c4883b7dd654@wirelessmundi.com>
 <CADihe3K-yyXFF8=w_7CLS3thpV9mbHnZWxbwR3TUOVy_k_nwsA@mail.gmail.com>
 <4d7501d4c87d$978a56c0$c69f0440$@freeswitch.org>
 <918f5e6d-bafb-3e1d-bd2d-26648ca948a7@bohboh.info>
 <007e01d4c88c$c717cf30$55476d90$@freeswitch.org>
 <0f4adbf7-4df1-de06-3e3b-e8c7f653712e@bohboh.info>
 <CADihe3KOsGs36Bayh8oZKUivJRcnrdZSWsxDqGmC9ifcLK04bg@mail.gmail.com>
Message-ID: <F02BA699-1E07-4DD1-A930-3061B880A7AE@freeswitch.org>

you need to look and see why mod_event_socket isnt loading and starting. if its because to added a line for mod_com_g729 the  check your config to make sure its correct



Sent from my iPhone

> On Feb 22, 2019, at 10:59, Yuriy Nasida <ynasida at gmail.com> wrote:
> 
> Well. I tried Antonio's method and have installed mod_com_g729 and freeswitch_license_server using deb files.
> But... I have got same result.  FS doesn't listen 8021 port in case I have enabled mod_com_g729 in modules.conf.xml
> 
> Also I noted that systemd starts two freeswitch processes/
> 
> # ps ax | grep free
> 22367 ?        Ss     0:04 /usr/sbin/freeswitch-license-server
> 22776 ?        Ss     0:00 /usr/bin/freeswitch -u freeswitch -g freeswitch -ncwait -nonat
> 22777 ?        S<sl   0:01 /usr/bin/freeswitch -u freeswitch -g freeswitch -ncwait -nonat
> 
> # netstat -nlp | grep free
> tcp        0      0 0.0.0.0:2855            0.0.0.0:*               LISTEN      22808/freeswitch    
> tcp        0      0 0.0.0.0:2856            0.0.0.0:*               LISTEN      22808/freeswitch    
> tcp        0      0 my_ip:5060      0.0.0.0:*               LISTEN      22808/freeswitch    
> tcp        0      0 my_ip:5060      0.0.0.0:*               LISTEN      22808/freeswitch    
> udp        0      0 my_ip:5060      0.0.0.0:*                           22808/freeswitch    
> udp        0      0 my_ip:5060      0.0.0.0:*                           22808/freeswitch    
> 
> FS doesn't listen 8021 like you see.
> 
> All things working fine I case I will disable mod_com_g729 in modules.conf.xml
> 
> Please advice 
> 
>> On Tue, 19 Feb 2019 at 23:20, Social Boh <social at bohboh.info> wrote:
>> So you don't update MPL 1.1 with MPL. 2.0 because the 1.1 is not compatible with GPL while 2.0 is compatible with GPL.
>> 
>> Regards
>> ---
>> I'm SoCIaL, MayBe
>> 
>>> El 19/02/2019 a las 14:53, Ken Rice escribió:
>>> I was pretty clear on the my statements about taking funding away from the FreeSWITCH Project and the devs by doing such thing
>>> 
>>>  
>>> 
>>> Also the bcg module is GPLv2 license violation waiting to be noticed… you cant combine MPL and GPL code and have a resulting MPL codebase. Its not allowed by the GPL. So you you are also ripping off the GPL coder.
>>> 
>>>  
>>> 
>>> Now you are polluting a MPL code base with the GPL violating the Copyright holder of the MPL’s software rights to determine how his code is licensed.
>>> 
>>>  
>>> 
>>> This happens all the time. People think oh this code is free, and yes most of the code is free, but the devs still have mortgages, children to feed, computers and related hardware to keep             updated so that they can continue to support that hardware that everyone is using currently. Website hosting for something like FreeSWITCH isnt free. It requires over 1/2 a rack of colo’d equipment to keep the website, mailing list, build servers, git repos, wiki etc etc etc going.
>>> 
>>>  
>>> 
>>> $10 for a channel is a pretty cheap way to support that seeing as that most of the time you don’t even need G729
>>> 
>>>  
>>> 
>>> From: Social Boh <social at bohboh.info> 
>>> Sent: Tuesday, February 19, 2019 1:03 PM
>>> To: Ken Rice <krice at freeswitch.org>
>>> Subject: Re: [Freeswitch-users] where g729 fs-latest-installer script now ?
>>> 
>>>  
>>> 
>>> Can you explain me why I can't use the mod_bcg729 with FreeSWITCH?
>>> 
>>> Thank you
>>> 
>>> ---
>>> I'm SoCIaL, MayBe
>>> El 19/02/2019 a las 13:04, Ken Rice escribió:
>>> 
>>> My statement wasn’t to you, but to the people that continually want to pirate someone elses G729 code and refuse to support the projects developers.
>>> 
>>>  
>>> 
>>> Its one of those things that grinds my gears.
>>> 
>>>  
>>> 
>>> K
>>> 
>>>  
>>> 
>>> From: FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> On Behalf Of Yuriy Nasida
>>> Sent: Tuesday, February 19, 2019 6:47 AM
>>> To: António Silva <asilva at wirelessmundi.com>
>>> Cc: freeswitch-users at lists.freeswitch.org
>>> Subject: Re: [Freeswitch-users] where g729 fs-latest-installer script now ?
>>> 
>>>  
>>> 
>>> Ken, not sure what you talking about. I already bought a lot of g729 licenses for my servers and I use them a lot of years. But now I can't use them normally at Deb 9 + fs1.8 + new g729 install script because of issues I described.
>>> 
>>>  
>>> 
>>> Thanks a lot Antonio! I will try this way.
>>> 
>>>  
>>> 
>>> On Tue, 19 Feb 2019 at 11:41, António Silva <asilva at wirelessmundi.com> wrote:
>>> 
>>> try using the .deb and installed then manuall.:
>>> 
>>> Get the packages from here: http://files.freeswitch.org/repo/deb/freeswitch-1.8/pool/main/f/
>>> 
>>> Installed them manually:
>>> 
>>> dpkg -x freeswitch-mod-com-g729_1.8.5~7~14d59c97e0~jessie_amd64.deb / 
>>> dpkg -x freeswitch-license-server_0.0.1~42~731842be7d~jessie_amd64.deb /
>>> 
>>>  
>>> 
>>> I don't know if the installer will remain, from the jira https://freeswitch.org/jira/browse/FS-11641 i got the impression that now the solution is to use the debian packages.
>>> 
>>>  
>>> 
>>> On 13/02/2019 18:10, Yuriy Nasida wrote:
>>> 
>>> I have installed freeswitch-license-server  using fs-201902072050-installer  and got issue below. 
>>> 
>>> FS doesn't listen port 8021 and I can't connect to it and run any command using fs_cli.
>>> 
>>> All works fine in case I kill process freeswitch-license-server.
>>> 
>>>  
>>> 
>>> Can anybody comment please all these very weird things with commercial g729 at FS ?
>>> 
>>> Thanks.
>>> 
>>>  
>>> 
>>> On Wed, 13 Feb 2019 at 13:36, Yuriy Nasida <ynasida at gmail.com> wrote:
>>> 
>>> Well. Somebody updated http://files.freeswitch.org/g729/ and now it has fs-latest-installer. Thanks a lot whoever you are!!!
>>> 
>>> But... Now it has not install instructions  :)
>>> 
>>>  
>>> 
>>> On Wed, 13 Feb 2019 at 12:25, Yuriy Nasida <ynasida at gmail.com> wrote:
>>> 
>>> Um... Nobody uses g729 licences anymore ? :)
>>> 
>>>  
>>> 
>>> On Tue, 12 Feb 2019 at 20:07, Yuriy Nasida <ynasida at gmail.com> wrote:
>>> 
>>> Hi guys,
>>> 
>>>  
>>> 
>>> Does anybody know why site http://files.freeswitch.org/g729/ contain only install instrunction now ?  Where fs-latest-installer-v1.6 script or something like this ?
>>> 
>>>  
>>> 
>>> Please advice.
>>> 
>>>  
>>> 
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Services
>>> sales at freeswitch.com
>>> https://freeswitch.com
>>>  
>>> Official FreeSWITCH Sites
>>> https://freeswitch.com/oss
>>> https://freeswitch.org/confluence
>>> https://cluecon.com
>>>  
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> https://freeswitch.com
>>> -- 
>>> Saludos / Regards / Cumprimentos
>>> António Silva
>>> 
>>> 
>>> 
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Services
>>> sales at freeswitch.com
>>> https://freeswitch.com
>>>  
>>> Official FreeSWITCH Sites
>>> https://freeswitch.com/oss
>>> https://freeswitch.org/confluence
>>> https://cluecon.com
>>>  
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> https://freeswitch.com
>> _________________________________________________________________________
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>> 
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
> 
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From rick at magicmail.mooo.com  Mon Feb 25 19:08:58 2019
From: rick at magicmail.mooo.com (Rick Jarvis)
Date: Mon, 25 Feb 2019 19:08:58 +0000
Subject: [Freeswitch-users] Variable no longer showing in channels
Message-ID: <etPan.5c743d4a.120b4109.ba48@magicmail.mooo.com>

I got this working (with the help of this list) some time ago, but for some reason it seems to have stopped working.

I’m setting a variable ‘callsubject’ in the dialplan via httapi, for which I added a column in the channels table:

  ALTER TABLE channels ADD COLUMN callsubject VARCHAR(256)


 
      <action application="httapi" data="{url=https://server/${destination_number}}”/> <!-- gets variable callsubject as ’test’, amongst other stuff -->
      <action application="log" data="CRIT callsubject: ${callsubject}”/> <!-- correctly shows ’test’ -->


And yet, whilst the column shows up in ’show channels’, the leg in question has an empty value.

What am I missing, maybe?


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From mike at freeswitch.org  Mon Feb 25 21:07:34 2019
From: mike at freeswitch.org (Mike Jerris)
Date: Mon, 25 Feb 2019 14:07:34 -0700
Subject: [Freeswitch-users] Variable no longer showing in channels
In-Reply-To: <etPan.5c743d4a.120b4109.ba48@magicmail.mooo.com>
References: <etPan.5c743d4a.120b4109.ba48@magicmail.mooo.com>
Message-ID: <C7B6BCD9-4E43-47AA-AA44-5D6A8727B927@freeswitch.org>

The dial plan is parsed and extensions created, and then they are run.  The ${callsubject} in your log line is evaluated in routing state, before httapi is run in execute state

> On Feb 25, 2019, at 12:08 PM, Rick Jarvis <rick at magicmail.mooo.com> wrote:
> 
> I got this working (with the help of this list) some time ago, but for some reason it seems to have stopped working.
> 
> I’m setting a variable ‘callsubject’ in the dialplan via httapi, for which I added a column in the channels table:
> 
>   ALTER TABLE channels ADD COLUMN callsubject VARCHAR(256)
> 
> 
>  
>       <action application="httapi" data="{url=https://server/${destination_number}}”/ <https://server/$%7Bdestination_number%7D%7D%E2%80%9D/>> <!-- gets variable callsubject as ’test’, amongst other stuff -->
>       <action application="log" data="CRIT callsubject: ${callsubject}”/> <!-- correctly shows ’test’ -->
> 
> 
> And yet, whilst the column shows up in ’show channels’, the leg in question has an empty value.
> 
> What am I missing, maybe?
> 

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From rick at magicmail.mooo.com  Mon Feb 25 23:04:16 2019
From: rick at magicmail.mooo.com (Rick Jarvis)
Date: Mon, 25 Feb 2019 23:04:16 +0000
Subject: [Freeswitch-users] Variable no longer showing in channels
In-Reply-To: <C7B6BCD9-4E43-47AA-AA44-5D6A8727B927@freeswitch.org>
References: <etPan.5c743d4a.120b4109.ba48@magicmail.mooo.com>
 <C7B6BCD9-4E43-47AA-AA44-5D6A8727B927@freeswitch.org>
Message-ID: <etPan.5c747470.31f7e584.ba48@magicmail.mooo.com>

Sorry Mike, can you expand on that? It was all working fine before, so I’m not sure either what “has changed” or what “needs to change”, if you see what I mean!


From: Mike Jerris <mike at freeswitch.org>
Reply: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Date: 25 February 2019 at 21:08:18
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject:  Re: [Freeswitch-users] Variable no longer showing in channels  

The dial plan is parsed and extensions created, and then they are run.  The ${callsubject} in your log line is evaluated in routing state, before httapi is run in execute state

On Feb 25, 2019, at 12:08 PM, Rick Jarvis <rick at magicmail.mooo.com> wrote:

I got this working (with the help of this list) some time ago, but for some reason it seems to have stopped working.

I’m setting a variable ‘callsubject’ in the dialplan via httapi, for which I added a column in the channels table:

ALTER TABLE channels ADD COLUMN callsubject
VARCHAR(256)


 
      <action application="httapi" data="{url=https://server/${destination_number}}”/> <!-- gets variable callsubject as ’test’, amongst other stuff -->
      <action application="log" data="CRIT callsubject: ${callsubject}”/> <!-- correctly shows ’test’ -->


And yet, whilst the column shows up in ’show channels’, the leg in question has an empty value.

What am I missing, maybe?


_________________________________________________________________________
Professional FreeSWITCH Services
sales at freeswitch.com
https://freeswitch.com

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

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From rajil.s at gmail.com  Tue Feb 26 02:12:01 2019
From: rajil.s at gmail.com (Rajil Saraswat)
Date: Mon, 25 Feb 2019 20:12:01 -0600
Subject: [Freeswitch-users] Zoiper goes crazy with registrations RPORT
	bug?
In-Reply-To: <be1f4cc8-37f0-5af2-abf8-6f04ae3d1314@gmail.com>
References: <be1f4cc8-37f0-5af2-abf8-6f04ae3d1314@gmail.com>
Message-ID: <c2faed6b-01e2-1170-6bf2-6fe70862316a@gmail.com>

On 2/21/19 8:05 PM, Rajil Saraswat wrote:
>   Hello,
>
> I have a Zoiper android client with Freeswitch server version 
> 1.8.4-5-749a6e108b~64bit. If i select RPORT option in Zoiper along 
> with TLS/TCP transport protocol, i get hundreds for registration 
> requests per minute. Zoiper customer support thinks that Freeswitch is 
> at fault here. The message i received from them is below. Is there any 
> fix for this?
>
> ------------
>
> The issue in your case is that the topmost Via header does not have 
> the mandatory received parameter.
>
> According to RFC 3581 (An Extension to the Session Initiation Protocol 
> (SIP) for Symmetric Response Routing) Section 4. Server Behavior:
>
> "When a server compliant to this specification (which can be a proxy
> or UAS) receives a request, it examines the topmost Via header field
> value. If this Via header field value contains an "rport" parameter
> with no value, it MUST set the value of the parameter to the source
> port of the request. This is analogous to the way in which a server
> will insert the "received" parameter into the topmost Via header
> field value. * In fact, the server MUST insert a "received" parameter*
> containing the source IP address that the request came from, even if
> it is identical to the value of the "sent-by" component. Note that
> this processing takes place independent of the transport protocol."
>
> This is a server side issue and it is related most likely to the 
> version of FreeSWITCH that you are using.
>
> ----------
>
> Thanks
>
Anybody else using Zoiper and facing this issue or is it just me?

Thanks



From ynasida at gmail.com  Mon Feb 25 18:48:04 2019
From: ynasida at gmail.com (Yuriy Nasida)
Date: Mon, 25 Feb 2019 21:48:04 +0300
Subject: [Freeswitch-users] where g729 fs-latest-installer script now ?
In-Reply-To: <F02BA699-1E07-4DD1-A930-3061B880A7AE@freeswitch.org>
References: <CADihe3K9Cvhjtpdox-QXO-KUSHew-aey16zirECudPUQ6kNYpA@mail.gmail.com>
 <CADihe3+UAjHwHJyz2+Co1+oDnvu=79PhBNfVZM5jh+Vy3OOzNA@mail.gmail.com>
 <CADihe3KvzFs5X62R1EXjhY7k7+MEGjM63u7woyL-BXZaA-BZBw@mail.gmail.com>
 <CADihe3LQwqFQ9vLqU34X_taK0HEf2etp=3CO1S9YkQFCy6wSag@mail.gmail.com>
 <4af1376d-29ba-ea47-c952-c4883b7dd654@wirelessmundi.com>
 <CADihe3K-yyXFF8=w_7CLS3thpV9mbHnZWxbwR3TUOVy_k_nwsA@mail.gmail.com>
 <4d7501d4c87d$978a56c0$c69f0440$@freeswitch.org>
 <918f5e6d-bafb-3e1d-bd2d-26648ca948a7@bohboh.info>
 <007e01d4c88c$c717cf30$55476d90$@freeswitch.org>
 <0f4adbf7-4df1-de06-3e3b-e8c7f653712e@bohboh.info>
 <CADihe3KOsGs36Bayh8oZKUivJRcnrdZSWsxDqGmC9ifcLK04bg@mail.gmail.com>
 <F02BA699-1E07-4DD1-A930-3061B880A7AE@freeswitch.org>
Message-ID: <CADihe3Keh5vMibfgz9WrCWthTMMUS2v5=ztD5gVj1wepE2rxQw@mail.gmail.com>

Ken, mod_event_socket is started and there is not errors related. But... FS
doesn't listen 8021 when mod_com_g729 is added to list of modules. I  would
be happy to check anything in config but currently haven't ideas what to
check.
Please advice.

On Sat, 23 Feb 2019 at 05:53, Ken Rice <krice at freeswitch.org> wrote:

> you need to look and see why mod_event_socket isnt loading and starting.
> if its because to added a line for mod_com_g729 the  check your config to
> make sure its correct
>
>
>
> Sent from my iPhone
>
> On Feb 22, 2019, at 10:59, Yuriy Nasida <ynasida at gmail.com> wrote:
>
> Well. I tried Antonio's method and have installed mod_com_g729 and
> freeswitch_license_server using deb files.
> But... I have got same result.  FS doesn't listen 8021 port in case I have
> enabled mod_com_g729 in modules.conf.xml
>
> Also I noted that systemd starts two freeswitch processes/
>
> # ps ax | grep free
> 22367 ?        Ss     0:04 /usr/sbin/freeswitch-license-server
> 22776 ?        Ss     0:00 /usr/bin/freeswitch -u freeswitch -g freeswitch
> -ncwait -nonat
> 22777 ?        S<sl   0:01 /usr/bin/freeswitch -u freeswitch -g freeswitch
> -ncwait -nonat
>
> # netstat -nlp | grep free
> tcp        0      0 0.0.0.0:2855            0.0.0.0:*
>  LISTEN      22808/freeswitch
> tcp        0      0 0.0.0.0:2856            0.0.0.0:*
>  LISTEN      22808/freeswitch
> tcp        0      0 my_ip:5060      0.0.0.0:*               LISTEN
> 22808/freeswitch
> tcp        0      0 my_ip:5060      0.0.0.0:*               LISTEN
> 22808/freeswitch
> udp        0      0 my_ip:5060      0.0.0.0:*
>  22808/freeswitch
> udp        0      0 my_ip:5060      0.0.0.0:*
>  22808/freeswitch
>
> FS doesn't listen 8021 like you see.
>
> All things working fine I case I will disable mod_com_g729 in
> modules.conf.xml
>
> Please advice
>
> On Tue, 19 Feb 2019 at 23:20, Social Boh <social at bohboh.info> wrote:
>
>> So you don't update MPL 1.1 with MPL. 2.0 because the 1.1 is not
>> compatible with GPL while 2.0 is compatible with GPL.
>> Regards
>>
>> ---
>> I'm SoCIaL, MayBe
>>
>>
>> El 19/02/2019 a las 14:53, Ken Rice escribió:
>>
>> I was pretty clear on the my statements about taking funding away from
>> the FreeSWITCH Project and the devs by doing such thing
>>
>>
>>
>> Also the bcg module is GPLv2 license violation waiting to be noticed… you
>> cant combine MPL and GPL code and have a resulting MPL codebase. Its not
>> allowed by the GPL. So you you are also ripping off the GPL coder.
>>
>>
>>
>> Now you are polluting a MPL code base with the GPL violating the
>> Copyright holder of the MPL’s software rights to determine how his code is
>> licensed.
>>
>>
>>
>> This happens all the time. People think oh this code is free, and yes
>> most of the code is free, but the devs still have mortgages, children to
>> feed, computers and related hardware to keep updated so that they can
>> continue to support that hardware that everyone is using currently. Website
>> hosting for something like FreeSWITCH isnt free. It requires over 1/2 a
>> rack of colo’d equipment to keep the website, mailing list, build servers,
>> git repos, wiki etc etc etc going.
>>
>>
>>
>> $10 for a channel is a pretty cheap way to support that seeing as that
>> most of the time you don’t even need G729
>>
>>
>>
>> *From:* Social Boh <social at bohboh.info> <social at bohboh.info>
>> *Sent:* Tuesday, February 19, 2019 1:03 PM
>> *To:* Ken Rice <krice at freeswitch.org> <krice at freeswitch.org>
>> *Subject:* Re: [Freeswitch-users] where g729 fs-latest-installer script
>> now ?
>>
>>
>>
>> Can you explain me why I can't use the mod_bcg729 with FreeSWITCH?
>>
>> Thank you
>>
>> ---
>>
>> I'm SoCIaL, MayBe
>>
>> El 19/02/2019 a las 13:04, Ken Rice escribió:
>>
>> My statement wasn’t to you, but to the people that continually want to
>> pirate someone elses G729 code and refuse to support the projects
>> developers.
>>
>>
>>
>> Its one of those things that grinds my gears.
>>
>>
>>
>> K
>>
>>
>>
>> *From:* FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org>
>> <freeswitch-users-bounces at lists.freeswitch.org> *On Behalf Of *Yuriy
>> Nasida
>> *Sent:* Tuesday, February 19, 2019 6:47 AM
>> *To:* António Silva <asilva at wirelessmundi.com> <asilva at wirelessmundi.com>
>> *Cc:* freeswitch-users at lists.freeswitch.org
>> *Subject:* Re: [Freeswitch-users] where g729 fs-latest-installer script
>> now ?
>>
>>
>>
>> Ken, not sure what you talking about. I already bought a lot of g729
>> licenses for my servers and I use them a lot of years. But now I can't use
>> them normally at Deb 9 + fs1.8 + new g729 install script because of issues
>> I described.
>>
>>
>>
>> Thanks a lot Antonio! I will try this way.
>>
>>
>>
>> On Tue, 19 Feb 2019 at 11:41, António Silva <asilva at wirelessmundi.com>
>> wrote:
>>
>> try using the .deb and installed then manuall.:
>>
>> Get the packages from here:
>> http://files.freeswitch.org/repo/deb/freeswitch-1.8/pool/main/f/
>>
>> Installed them manually:
>>
>> dpkg -x freeswitch-mod-com-g729_1.8.5~7~14d59c97e0~jessie_amd64.deb /
>> dpkg -x freeswitch-license-server_0.0.1~42~731842be7d~jessie_amd64.deb /
>>
>>
>>
>> I don't know if the installer will remain, from the jira
>> https://freeswitch.org/jira/browse/FS-11641 i got the impression that
>> now the solution is to use the debian packages.
>>
>>
>>
>> On 13/02/2019 18:10, Yuriy Nasida wrote:
>>
>> I have installed freeswitch-license-server  using
>> fs-201902072050-installer  and got issue below.
>>
>> FS doesn't listen port 8021 and I can't connect to it and run any command
>> using fs_cli.
>>
>> All works fine in case I kill process freeswitch-license-server.
>>
>>
>>
>> Can anybody comment please all these very weird things with commercial
>> g729 at FS ?
>>
>> Thanks.
>>
>>
>>
>> On Wed, 13 Feb 2019 at 13:36, Yuriy Nasida <ynasida at gmail.com> wrote:
>>
>> Well. Somebody updated http://files.freeswitch.org/g729/ and now it
>> has fs-latest-installer. Thanks a lot whoever you are!!!
>>
>> But... Now it has not install instructions  :)
>>
>>
>>
>> On Wed, 13 Feb 2019 at 12:25, Yuriy Nasida <ynasida at gmail.com> wrote:
>>
>> Um... Nobody uses g729 licences anymore ? :)
>>
>>
>>
>> On Tue, 12 Feb 2019 at 20:07, Yuriy Nasida <ynasida at gmail.com> wrote:
>>
>> Hi guys,
>>
>>
>>
>> Does anybody know why site http://files.freeswitch.org/g729/ contain
>> only install instrunction now ?  Where fs-latest-installer-v1.6 script or
>> something like this ?
>>
>>
>>
>> Please advice.
>>
>>
>>
>> _________________________________________________________________________
>>
>> Professional FreeSWITCH Services
>>
>> sales at freeswitch.com
>>
>> https://freeswitch.com
>>
>>
>>
>> Official FreeSWITCH Sites
>>
>> https://freeswitch.com/oss
>>
>> https://freeswitch.org/confluence
>>
>> https://cluecon.com
>>
>>
>>
>> FreeSWITCH-users mailing list
>>
>> FreeSWITCH-users at lists.freeswitch.org
>>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>
>> https://freeswitch.com
>>
>> --
>>
>> Saludos / Regards / Cumprimentos
>>
>> António Silva
>>
>>
>>
>> _________________________________________________________________________
>>
>> Professional FreeSWITCH Services
>>
>> sales at freeswitch.com
>>
>> https://freeswitch.com
>>
>>
>>
>> Official FreeSWITCH Sites
>>
>> https://freeswitch.com/oss
>>
>> https://freeswitch.org/confluence
>>
>> https://cluecon.com
>>
>>
>>
>> FreeSWITCH-users mailing list
>>
>> FreeSWITCH-users at lists.freeswitch.org
>>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>
>> https://freeswitch.com
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From bipin at xbipin.com  Tue Feb 26 08:54:42 2019
From: bipin at xbipin.com (Bipin Patel)
Date: Tue, 26 Feb 2019 12:54:42 +0400
Subject: [Freeswitch-users] Zoiper goes crazy with registrations RPORT bug?
In-Reply-To: <c2faed6b-01e2-1170-6bf2-6fe70862316a@gmail.com>
References: <be1f4cc8-37f0-5af2-abf8-6f04ae3d1314@gmail.com>
 <c2faed6b-01e2-1170-6bf2-6fe70862316a@gmail.com>
Message-ID: <16929036450.279b.b07ebdf329620b8089087c7205b03f01@xbipin.com>

Hi,

I use zoiper with freeswitch and tls+srtp and I don't have such an issue 
with rport other than the tls profile which keeps hanging with is another 
bug. Could you kindly provide the profile xml to which the clients are 
registering



On February 26, 2019 6:19:16 AM Rajil Saraswat <rajil.s at gmail.com> wrote:

> On 2/21/19 8:05 PM, Rajil Saraswat wrote:
>>   Hello,
>>
>> I have a Zoiper android client with Freeswitch server version
>> 1.8.4-5-749a6e108b~64bit. If i select RPORT option in Zoiper along
>> with TLS/TCP transport protocol, i get hundreds for registration
>> requests per minute. Zoiper customer support thinks that Freeswitch is
>> at fault here. The message i received from them is below. Is there any
>> fix for this?
>>
>> ------------
>>
>> The issue in your case is that the topmost Via header does not have
>> the mandatory received parameter.
>>
>> According to RFC 3581 (An Extension to the Session Initiation Protocol
>> (SIP) for Symmetric Response Routing) Section 4. Server Behavior:
>>
>> "When a server compliant to this specification (which can be a proxy
>> or UAS) receives a request, it examines the topmost Via header field
>> value. If this Via header field value contains an "rport" parameter
>> with no value, it MUST set the value of the parameter to the source
>> port of the request. This is analogous to the way in which a server
>> will insert the "received" parameter into the topmost Via header
>> field value. * In fact, the server MUST insert a "received" parameter*
>> containing the source IP address that the request came from, even if
>> it is identical to the value of the "sent-by" component. Note that
>> this processing takes place independent of the transport protocol."
>>
>> This is a server side issue and it is related most likely to the
>> version of FreeSWITCH that you are using.
>>
>> ----------
>>
>> Thanks
>>
> Anybody else using Zoiper and facing this issue or is it just me?
>
> Thanks
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com






From mansig130 at gmail.com  Tue Feb 26 08:14:58 2019
From: mansig130 at gmail.com (Mansi Gupta)
Date: Tue, 26 Feb 2019 01:14:58 -0700 (MST)
Subject: [Freeswitch-users] Codec negotiation with g729.
Message-ID: <1551168898295-0.post@n2.nabble.com>

Hello!
I am trying to make a call using codec g729,but repeatedly i was getting the
error of codec negotiation.However,i have resolved it. Although it offers a
list of codec with different payloads for the same codec but it does not
match with other audio codec.I have to match the audio codec g729 with the
other one.

Here are few relevant lines of the log:

Local SDP:
v=0
o=FreeSWITCH 1551144396 1551144397 IN IP4 192.168.1.123
s=FreeSWITCH
c=IN IP4 192.168.1.123
t=0 0
m=audio 22412 RTP/AVP 0 18 102 9 8 101 104
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:102 opus/48000/2
2019-02-26 02:40:10.077544 [DEBUG] switch_core_media.c:4449 Audio Codec
Compare [PCMU:0:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]
2019-02-26 02:40:10.077544 [DEBUG] switch_core_media.c:4504 Audio Codec
Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match
2019-02-26 02:40:10.077544 [DEBUG] switch_core_media.c:4449 Audio Codec
Compare [PCMU:0:8000:20:64000:1]/[G729:18:8000:20:8000:1]
2019-02-26 02:40:10.077544 [DEBUG] switch_core_media.c:4449 Audio Codec
Compare [PCMU:0:8000:20:64000:1]/[opus:116:48000:20:0:1]
2019-02-26 02:40:10.077544 [DEBUG] switch_core_media.c:4449 Audio Codec
Compare [PCMU:0:8000:20:64000:1]/[G722:9:8000:20:64000:1]
2019-02-26 02:40:10.077544 [DEBUG] switch_core_media.c:4449 Audio Codec
Compare [PCMU:0:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
2019-02-26 02:40:10.077544 [DEBUG] switch_core_media.c:4449 Audio Codec
Compare [GSM:3:8000:20:13200:1]/[PCMU:0:8000:20:64000:1]
2019-02-26 02:40:10.077544 [DEBUG] switch_core_media.c:4449 Audio Codec
Compare [GSM:3:8000:20:13200:1]/[G729:18:8000:20:8000:1]
2019-02-26 02:40:10.077544 [DEBUG] switch_core_media.c:4449 Audio Codec
Compare [GSM:3:8000:20:13200:1]/[opus:116:48000:20:0:1]
2019-02-26 02:40:10.077544 [DEBUG] switch_core_media.c:4449 Audio Codec
Compare [GSM:3:8000:20:13200:1]/[G722:9:8000:20:64000:1]
2019-02-26 02:40:10.077544 [DEBUG] switch_core_media.c:4449 Audio Codec
Compare [GSM:3:8000:20:13200:1]/[PCMA:8:8000:20:64000:1]
2019-02-26 02:40:10.077544 [DEBUG] switch_core_media.c:4449 Audio Codec
Compare [speex:110:8000:20:0:1]/[PCMU:0:8000:20:64000:1]
2019-02-26 02:40:10.077544 [DEBUG] switch_core_media.c:4449 Audio Codec
Compare [speex:110:8000:20:0:1]/[G729:18:8000:20:8000:1]
2019-02-26 02:40:10.077544 [DEBUG] switch_core_media.c:4449 Audio Codec
Compare [speex:110:8000:20:0:1]/[opus:116:48000:20:0:1]
2019-02-26 02:40:10.077544 [DEBUG] switch_core_media.c:4449 Audio Codec
Compare [speex:110:8000:20:0:1]/[G722:9:8000:20:64000:1]
2019-02-26 02:40:10.077544 [DEBUG] switch_core_media.c:4449 Audio Codec
Compare [speex:110:8000:20:0:1]/[PCMA:8:8000:20:64000:1]
2019-02-26 02:40:10.077544 [DEBUG] switch_core_media.c:4449 Audio Codec
Compare [PCMA:8:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]
2019-02-26 02:40:10.077544 [DEBUG] switch_core_media.c:4449 Audio Codec
Compare [PCMA:8:8000:20:64000:1]/[G729:18:8000:20:8000:1]
2019-02-26 02:40:10.077544 [DEBUG] switch_core_media.c:4449 Audio Codec
Compare [PCMA:8:8000:20:64000:1]/[opus:116:48000:20:0:1]
2019-02-26 02:40:10.077544 [DEBUG] switch_core_media.c:4449 Audio Codec
Compare [PCMA:8:8000:20:64000:1]/[G722:9:8000:20:64000:1]
2019-02-26 02:40:10.077544 [DEBUG] switch_core_media.c:4449 Audio Codec
Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
2019-02-26 02:40:10.077544 [DEBUG] switch_core_media.c:4504 Audio Codec
Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match
2019-02-26 02:40:10.077544 [DEBUG] switch_core_media.c:4449 Audio Codec
Compare [iLBC:98:8000:30:0:1]/[PCMU:0:8000:20:64000:1]
2019-02-26 02:40:10.077544 [DEBUG] switch_core_media.c:4449 Audio Codec
Compare [iLBC:98:8000:30:0:1]/[G729:18:8000:20:8000:1]
2019-02-26 02:40:10.077544 [DEBUG] switch_core_media.c:4449 Audio Codec
Compare [iLBC:98:8000:30:0:1]/[opus:116:48000:20:0:1]
2019-02-26 02:40:10.077544 [DEBUG] switch_core_media.c:4449 Audio Codec
Compare [iLBC:98:8000:30:0:1]/[G722:9:8000:20:64000:1]
2019-02-26 02:40:10.077544 [DEBUG] switch_core_media.c:4449 Audio Codec
Compare [iLBC:98:8000:30:0:1]/[PCMA:8:8000:20:64000:1]


Can anyone please point,how to resolve it?
Thanks.



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


From asilva at wirelessmundi.com  Tue Feb 26 08:55:38 2019
From: asilva at wirelessmundi.com (=?UTF-8?Q?Ant=c3=b3nio_Silva?=)
Date: Tue, 26 Feb 2019 09:55:38 +0100
Subject: [Freeswitch-users] where g729 fs-latest-installer script now ?
In-Reply-To: <CADihe3Keh5vMibfgz9WrCWthTMMUS2v5=ztD5gVj1wepE2rxQw@mail.gmail.com>
References: <CADihe3K9Cvhjtpdox-QXO-KUSHew-aey16zirECudPUQ6kNYpA@mail.gmail.com>
 <CADihe3+UAjHwHJyz2+Co1+oDnvu=79PhBNfVZM5jh+Vy3OOzNA@mail.gmail.com>
 <CADihe3KvzFs5X62R1EXjhY7k7+MEGjM63u7woyL-BXZaA-BZBw@mail.gmail.com>
 <CADihe3LQwqFQ9vLqU34X_taK0HEf2etp=3CO1S9YkQFCy6wSag@mail.gmail.com>
 <4af1376d-29ba-ea47-c952-c4883b7dd654@wirelessmundi.com>
 <CADihe3K-yyXFF8=w_7CLS3thpV9mbHnZWxbwR3TUOVy_k_nwsA@mail.gmail.com>
 <4d7501d4c87d$978a56c0$c69f0440$@freeswitch.org>
 <918f5e6d-bafb-3e1d-bd2d-26648ca948a7@bohboh.info>
 <007e01d4c88c$c717cf30$55476d90$@freeswitch.org>
 <0f4adbf7-4df1-de06-3e3b-e8c7f653712e@bohboh.info>
 <CADihe3KOsGs36Bayh8oZKUivJRcnrdZSWsxDqGmC9ifcLK04bg@mail.gmail.com>
 <F02BA699-1E07-4DD1-A930-3061B880A7AE@freeswitch.org>
 <CADihe3Keh5vMibfgz9WrCWthTMMUS2v5=ztD5gVj1wepE2rxQw@mail.gmail.com>
Message-ID: <63fc76c7-023e-4942-d8c0-994556e65fb7@wirelessmundi.com>

Yuriy,

i think the best is to open a jira and put there all the information you 
can,  configuration, logs...

check:

https://freeswitch.org/confluence/display/FREESWITCH/Reporting+Bugs+to+JIRA


On 25/02/2019 19:48, Yuriy Nasida wrote:
> Ken, mod_event_socket is started and there is not errors related. 
> But... FS doesn't listen 8021 when mod_com_g729 is added to list of 
> modules. I  would be happy to check anything in config but currently 
> haven't ideas what to check.
> Please advice.
>
> On Sat, 23 Feb 2019 at 05:53, Ken Rice <krice at freeswitch.org 
> <mailto:krice at freeswitch.org>> wrote:
>
>     you need to look and see why mod_event_socket isnt loading and
>     starting. if its because to added a line for mod_com_g729 the
>      check your config to make sure its correct
>
>
>
>     Sent from my iPhone
>
>     On Feb 22, 2019, at 10:59, Yuriy Nasida <ynasida at gmail.com
>     <mailto:ynasida at gmail.com>> wrote:
>
>>     Well. I tried Antonio's method and have installed mod_com_g729
>>     and freeswitch_license_server using deb files.
>>     But... I have got same result.  FS doesn't listen 8021 port in
>>     case I have enabled mod_com_g729 in modules.conf.xml
>>
>>     Also I noted that systemd starts two freeswitch processes/
>>
>>     # ps ax | grep free
>>     22367 ?        Ss     0:04 /usr/sbin/freeswitch-license-server
>>     22776 ?        Ss     0:00 /usr/bin/freeswitch -u freeswitch -g
>>     freeswitch -ncwait -nonat
>>     22777 ?        S<sl   0:01 /usr/bin/freeswitch -u freeswitch -g
>>     freeswitch -ncwait -nonat
>>
>>     # netstat -nlp | grep free
>>     tcp        0      0 0.0.0.0:2855 <http://0.0.0.0:2855>          
>>     0.0.0.0:*               LISTEN     22808/freeswitch
>>     tcp        0      0 0.0.0.0:2856 <http://0.0.0.0:2856>          
>>     0.0.0.0:*               LISTEN     22808/freeswitch
>>     tcp        0      0 my_ip:5060 0.0.0.0:*               LISTEN
>>     22808/freeswitch
>>     tcp        0      0 my_ip:5060 0.0.0.0:*               LISTEN
>>     22808/freeswitch
>>     udp        0      0 my_ip:5060 0.0.0.0:*  22808/freeswitch
>>     udp        0      0 my_ip:5060 0.0.0.0:*  22808/freeswitch
>>
>>     FS doesn't listen 8021 like you see.
>>
>>     All things working fine I case I will disable mod_com_g729 in
>>     modules.conf.xml
>>
>>     Please advice
>>
>>     On Tue, 19 Feb 2019 at 23:20, Social Boh <social at bohboh.info
>>     <mailto:social at bohboh.info>> wrote:
>>
>>         So you don't update MPL 1.1 with MPL. 2.0 because the 1.1 is
>>         not compatible with GPL while 2.0 is compatible with GPL.
>>
>>         Regards
>>
>>         ---
>>         I'm SoCIaL, MayBe
>>
>>         El 19/02/2019 a las 14:53, Ken Rice escribió:
>>>
>>>         I was pretty clear on the my statements about taking funding
>>>         away from the FreeSWITCH Project and the devs by doing such
>>>         thing
>>>
>>>         Also the bcg module is GPLv2 license violation waiting to be
>>>         noticed… you cant combine MPL and GPL code and have a
>>>         resulting MPL codebase. Its not allowed by the GPL. So you
>>>         you are also ripping off the GPL coder.
>>>
>>>         Now you are polluting a MPL code base with the GPL violating
>>>         the Copyright holder of the MPL’s software rights to
>>>         determine how his code is licensed.
>>>
>>>         This happens all the time. People think oh this code is
>>>         free, and yes most of the code is free, but the devs still
>>>         have mortgages, children to feed, computers and related
>>>         hardware to keep updated so that they can continue to
>>>         support that hardware that everyone is using currently.
>>>         Website hosting for something like FreeSWITCH isnt free. It
>>>         requires over 1/2 a rack of colo’d equipment to keep the
>>>         website, mailing list, build servers, git repos, wiki etc
>>>         etc etc going.
>>>
>>>         $10 for a channel is a pretty cheap way to support that
>>>         seeing as that most of the time you don’t even need G729
>>>
>>>         *From:*Social Boh <social at bohboh.info>
>>>         <mailto:social at bohboh.info>
>>>         *Sent:* Tuesday, February 19, 2019 1:03 PM
>>>         *To:* Ken Rice <krice at freeswitch.org>
>>>         <mailto:krice at freeswitch.org>
>>>         *Subject:* Re: [Freeswitch-users] where g729
>>>         fs-latest-installer script now ?
>>>
>>>         Can you explain me why I can't use the mod_bcg729 with
>>>         FreeSWITCH?
>>>
>>>         Thank you
>>>
>>>         ---
>>>         I'm SoCIaL, MayBe
>>>
>>>         El 19/02/2019 a las 13:04, Ken Rice escribió:
>>>
>>>             My statement wasn’t to you, but to the people that
>>>             continually want to pirate someone elses G729 code and
>>>             refuse to support the projects developers.
>>>
>>>             Its one of those things that grinds my gears.
>>>
>>>             K
>>>
>>>             *From:* FreeSWITCH-users
>>>             <freeswitch-users-bounces at lists.freeswitch.org>
>>>             <mailto:freeswitch-users-bounces at lists.freeswitch.org>
>>>             *On Behalf Of *Yuriy Nasida
>>>             *Sent:* Tuesday, February 19, 2019 6:47 AM
>>>             *To:* António Silva <asilva at wirelessmundi.com>
>>>             <mailto:asilva at wirelessmundi.com>
>>>             *Cc:* freeswitch-users at lists.freeswitch.org
>>>             <mailto:freeswitch-users at lists.freeswitch.org>
>>>             *Subject:* Re: [Freeswitch-users] where g729
>>>             fs-latest-installer script now ?
>>>
>>>             Ken, not sure what you talking about. I already bought a
>>>             lot of g729 licenses for my servers and I use them a lot
>>>             of years. But now I can't use them normally at Deb 9 +
>>>             fs1.8 + new g729 install script because of issues I
>>>             described.
>>>
>>>             Thanks a lot Antonio! I will try this way.
>>>
>>>             On Tue, 19 Feb 2019 at 11:41, António Silva
>>>             <asilva at wirelessmundi.com
>>>             <mailto:asilva at wirelessmundi.com>> wrote:
>>>
>>>                 try using the .deb and installed then manuall.:
>>>
>>>                 Get the packages from here:
>>>                 http://files.freeswitch.org/repo/deb/freeswitch-1.8/pool/main/f/
>>>
>>>                 Installed them manually:
>>>
>>>                 dpkg -x
>>>                 freeswitch-mod-com-g729_1.8.5~7~14d59c97e0~jessie_amd64.deb
>>>                 /
>>>                 dpkg -x
>>>                 freeswitch-license-server_0.0.1~42~731842be7d~jessie_amd64.deb
>>>                 /
>>>
>>>                 I don't know if the installer will remain, from the
>>>                 jira https://freeswitch.org/jira/browse/FS-11641 i
>>>                 got the impression that now the solution is to use
>>>                 the debian packages.
>>>
>>>                 On 13/02/2019 18:10, Yuriy Nasida wrote:
>>>
>>>                     I have installed freeswitch-license-server using
>>>                     fs-201902072050-installer  and got issue below.
>>>
>>>                     FS doesn't listen port 8021 and I can't connect
>>>                     to it and run any command using fs_cli.
>>>
>>>                     All works fine in case I kill process
>>>                     freeswitch-license-server.
>>>
>>>                     Can anybody comment please all these very weird
>>>                     things with commercial g729 at FS ?
>>>
>>>                     Thanks.
>>>
>>>                     On Wed, 13 Feb 2019 at 13:36, Yuriy Nasida
>>>                     <ynasida at gmail.com <mailto:ynasida at gmail.com>>
>>>                     wrote:
>>>
>>>                         Well. Somebody updated
>>>                         http://files.freeswitch.org/g729/ and now it
>>>                         has fs-latest-installer. Thanks a
>>>                         lot whoever you are!!!
>>>
>>>                         But... Now it has not install instructions  :)
>>>
>>>                         On Wed, 13 Feb 2019 at 12:25, Yuriy Nasida
>>>                         <ynasida at gmail.com
>>>                         <mailto:ynasida at gmail.com>> wrote:
>>>
>>>                             Um... Nobody uses g729 licences anymore ? :)
>>>
>>>                             On Tue, 12 Feb 2019 at 20:07, Yuriy
>>>                             Nasida <ynasida at gmail.com
>>>                             <mailto:ynasida at gmail.com>> wrote:
>>>
>>>                                 Hi guys,
>>>
>>>                                 Does anybody know why site
>>>                                 http://files.freeswitch.org/g729/
>>>                                 contain only install instrunction
>>>                                 now ? Where fs-latest-installer-v1.6
>>>                                 script or something like this ?
>>>
>>>                                 Please advice.
>>>
>>>                     _________________________________________________________________________
>>>
>>>                     Professional FreeSWITCH Services
>>>
>>>                     sales at freeswitch.com  <mailto:sales at freeswitch.com>
>>>
>>>                     https://freeswitch.com
>>>
>>>                       
>>>
>>>                     Official FreeSWITCH Sites
>>>
>>>                     https://freeswitch.com/oss
>>>
>>>                     https://freeswitch.org/confluence
>>>
>>>                     https://cluecon.com
>>>
>>>                       
>>>
>>>                     FreeSWITCH-users mailing list
>>>
>>>                     FreeSWITCH-users at lists.freeswitch.org  <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>
>>>                     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>
>>>                     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>
>>>                     https://freeswitch.com
>>>
>>>                 -- 
>>>
>>>                 Saludos / Regards / Cumprimentos
>>>
>>>                 António Silva
>>>
>>>
>>>
>>>             _________________________________________________________________________
>>>
>>>             Professional FreeSWITCH Services
>>>
>>>             sales at freeswitch.com  <mailto:sales at freeswitch.com>
>>>
>>>             https://freeswitch.com
>>>
>>>               
>>>
>>>             Official FreeSWITCH Sites
>>>
>>>             https://freeswitch.com/oss
>>>
>>>             https://freeswitch.org/confluence
>>>
>>>             https://cluecon.com
>>>
>>>               
>>>
>>>             FreeSWITCH-users mailing list
>>>
>>>             FreeSWITCH-users at lists.freeswitch.org  <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>
>>>             http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>
>>>             UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>
>>>             https://freeswitch.com
>>>
>>         _________________________________________________________________________
>>         Professional FreeSWITCH Services
>>         sales at freeswitch.com <mailto:sales at freeswitch.com>
>>         https://freeswitch.com
>>
>>         Official FreeSWITCH Sites
>>         https://freeswitch.com/oss
>>         https://freeswitch.org/confluence
>>         https://cluecon.com
>>
>>         FreeSWITCH-users mailing list
>>         FreeSWITCH-users at lists.freeswitch.org
>>         <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>         https://freeswitch.com
>>
>>     _________________________________________________________________________
>>     Professional FreeSWITCH Services
>>     sales at freeswitch.com <mailto:sales at freeswitch.com>
>>     https://freeswitch.com
>>
>>     Official FreeSWITCH Sites
>>     https://freeswitch.com/oss
>>     https://freeswitch.org/confluence
>>     https://cluecon.com
>>
>>     FreeSWITCH-users mailing list
>>     FreeSWITCH-users at lists.freeswitch.org
>>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>     https://freeswitch.com
>     _________________________________________________________________________
>     Professional FreeSWITCH Services
>     sales at freeswitch.com <mailto:sales at freeswitch.com>
>     https://freeswitch.com
>
>     Official FreeSWITCH Sites
>     https://freeswitch.com/oss
>     https://freeswitch.org/confluence
>     https://cluecon.com
>
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     https://freeswitch.com
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com

-- 
Saludos / Regards / Cumprimentos
António Silva

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From s.safarov at gmail.com  Tue Feb 26 10:56:31 2019
From: s.safarov at gmail.com (Sergey Safarov)
Date: Tue, 26 Feb 2019 13:56:31 +0300
Subject: [Freeswitch-users] Codec negotiation with g729.
In-Reply-To: <1551168898295-0.post@n2.nabble.com>
References: <1551168898295-0.post@n2.nabble.com>
Message-ID: <CAHtxdDffbbS1K6x3hDC_uO6hrF-28GrBxDsV_djm0Cn3qT7+Zw@mail.gmail.com>

Try enable additional logs. This may help

sofia loglevel all 9
sofia global siptrace on



вт, 26 февр. 2019 г. в 12:59, Mansi Gupta <mansig130 at gmail.com>:

> Hello!
> I am trying to make a call using codec g729,but repeatedly i was getting
> the
> error of codec negotiation.However,i have resolved it. Although it offers a
> list of codec with different payloads for the same codec but it does not
> match with other audio codec.I have to match the audio codec g729 with the
> other one.
>
> Here are few relevant lines of the log:
>
> Local SDP:
> v=0
> o=FreeSWITCH 1551144396 1551144397 IN IP4 192.168.1.123
> s=FreeSWITCH
> c=IN IP4 192.168.1.123
> t=0 0
> m=audio 22412 RTP/AVP 0 18 102 9 8 101 104
> a=rtpmap:0 PCMU/8000
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:102 opus/48000/2
> 2019-02-26 02:40:10.077544 [DEBUG] switch_core_media.c:4449 Audio Codec
> Compare [PCMU:0:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]
> 2019-02-26 02:40:10.077544 [DEBUG] switch_core_media.c:4504 Audio Codec
> Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match
> 2019-02-26 02:40:10.077544 [DEBUG] switch_core_media.c:4449 Audio Codec
> Compare [PCMU:0:8000:20:64000:1]/[G729:18:8000:20:8000:1]
> 2019-02-26 02:40:10.077544 [DEBUG] switch_core_media.c:4449 Audio Codec
> Compare [PCMU:0:8000:20:64000:1]/[opus:116:48000:20:0:1]
> 2019-02-26 02:40:10.077544 [DEBUG] switch_core_media.c:4449 Audio Codec
> Compare [PCMU:0:8000:20:64000:1]/[G722:9:8000:20:64000:1]
> 2019-02-26 02:40:10.077544 [DEBUG] switch_core_media.c:4449 Audio Codec
> Compare [PCMU:0:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
> 2019-02-26 02:40:10.077544 [DEBUG] switch_core_media.c:4449 Audio Codec
> Compare [GSM:3:8000:20:13200:1]/[PCMU:0:8000:20:64000:1]
> 2019-02-26 02:40:10.077544 [DEBUG] switch_core_media.c:4449 Audio Codec
> Compare [GSM:3:8000:20:13200:1]/[G729:18:8000:20:8000:1]
> 2019-02-26 02:40:10.077544 [DEBUG] switch_core_media.c:4449 Audio Codec
> Compare [GSM:3:8000:20:13200:1]/[opus:116:48000:20:0:1]
> 2019-02-26 02:40:10.077544 [DEBUG] switch_core_media.c:4449 Audio Codec
> Compare [GSM:3:8000:20:13200:1]/[G722:9:8000:20:64000:1]
> 2019-02-26 02:40:10.077544 [DEBUG] switch_core_media.c:4449 Audio Codec
> Compare [GSM:3:8000:20:13200:1]/[PCMA:8:8000:20:64000:1]
> 2019-02-26 02:40:10.077544 [DEBUG] switch_core_media.c:4449 Audio Codec
> Compare [speex:110:8000:20:0:1]/[PCMU:0:8000:20:64000:1]
> 2019-02-26 02:40:10.077544 [DEBUG] switch_core_media.c:4449 Audio Codec
> Compare [speex:110:8000:20:0:1]/[G729:18:8000:20:8000:1]
> 2019-02-26 02:40:10.077544 [DEBUG] switch_core_media.c:4449 Audio Codec
> Compare [speex:110:8000:20:0:1]/[opus:116:48000:20:0:1]
> 2019-02-26 02:40:10.077544 [DEBUG] switch_core_media.c:4449 Audio Codec
> Compare [speex:110:8000:20:0:1]/[G722:9:8000:20:64000:1]
> 2019-02-26 02:40:10.077544 [DEBUG] switch_core_media.c:4449 Audio Codec
> Compare [speex:110:8000:20:0:1]/[PCMA:8:8000:20:64000:1]
> 2019-02-26 02:40:10.077544 [DEBUG] switch_core_media.c:4449 Audio Codec
> Compare [PCMA:8:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]
> 2019-02-26 02:40:10.077544 [DEBUG] switch_core_media.c:4449 Audio Codec
> Compare [PCMA:8:8000:20:64000:1]/[G729:18:8000:20:8000:1]
> 2019-02-26 02:40:10.077544 [DEBUG] switch_core_media.c:4449 Audio Codec
> Compare [PCMA:8:8000:20:64000:1]/[opus:116:48000:20:0:1]
> 2019-02-26 02:40:10.077544 [DEBUG] switch_core_media.c:4449 Audio Codec
> Compare [PCMA:8:8000:20:64000:1]/[G722:9:8000:20:64000:1]
> 2019-02-26 02:40:10.077544 [DEBUG] switch_core_media.c:4449 Audio Codec
> Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
> 2019-02-26 02:40:10.077544 [DEBUG] switch_core_media.c:4504 Audio Codec
> Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match
> 2019-02-26 02:40:10.077544 [DEBUG] switch_core_media.c:4449 Audio Codec
> Compare [iLBC:98:8000:30:0:1]/[PCMU:0:8000:20:64000:1]
> 2019-02-26 02:40:10.077544 [DEBUG] switch_core_media.c:4449 Audio Codec
> Compare [iLBC:98:8000:30:0:1]/[G729:18:8000:20:8000:1]
> 2019-02-26 02:40:10.077544 [DEBUG] switch_core_media.c:4449 Audio Codec
> Compare [iLBC:98:8000:30:0:1]/[opus:116:48000:20:0:1]
> 2019-02-26 02:40:10.077544 [DEBUG] switch_core_media.c:4449 Audio Codec
> Compare [iLBC:98:8000:30:0:1]/[G722:9:8000:20:64000:1]
> 2019-02-26 02:40:10.077544 [DEBUG] switch_core_media.c:4449 Audio Codec
> Compare [iLBC:98:8000:30:0:1]/[PCMA:8:8000:20:64000:1]
>
>
> Can anyone please point,how to resolve it?
> Thanks.
>
>
>
> --
> Sent from: http://freeswitch-users.2379917.n2.nabble.com/
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From koketsom at tluka.co.za  Tue Feb 26 19:29:19 2019
From: koketsom at tluka.co.za (Koketso Mabuse)
Date: Tue, 26 Feb 2019 21:29:19 +0200
Subject: [Freeswitch-users] Incoming Calls Fails
In-Reply-To: <CAHtxdDffbbS1K6x3hDC_uO6hrF-28GrBxDsV_djm0Cn3qT7+Zw@mail.gmail.com>
References: <1551168898295-0.post@n2.nabble.com>
 <CAHtxdDffbbS1K6x3hDC_uO6hrF-28GrBxDsV_djm0Cn3qT7+Zw@mail.gmail.com>
Message-ID: <962a1c48-5687-77db-87bf-e5f7d75f2bf4@tluka.co.za>

Good Evening

Its been a rough 2 weeks trying to figure out what the issue could be.

Current Status:

  * Gateway set, can call out... (Outbound Working)
  * Extensions registering fine.
  * Incoming call fails, nothing gets through... Checked codecs, all
    setting seems to be according to all tutorials...

There lies the issue...Any thoughts anyone.

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From igor.potjevlesch at gmail.com  Wed Feb 27 09:34:16 2019
From: igor.potjevlesch at gmail.com (igor.potjevlesch at gmail.com)
Date: Wed, 27 Feb 2019 10:34:16 +0100
Subject: [Freeswitch-users] Call hold issue
Message-ID: <003201d4ce7f$9b9504b0$d2bf0e10$@gmail.com>

Hello,

 

I have a weird behaviour with a call hold scenario. When the customer on leg
A is placing the call on-hold (by sending a RE INVITE with sendonly
attribute in the SDP), Freeswitch is switching to hold: "Callstate Change
ACTIVE -> HELD". But only on the leg A. 

The holding is not forwarded  to the leg B. So, it causes the called party
not to hear the Music On Hold generated by the calling party.

 

Any idea of what could result in this behaviour?

 

Regards,

 

Igor.

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From mirkobrankovic at gmail.com  Wed Feb 27 10:33:55 2019
From: mirkobrankovic at gmail.com (Mirko Brankovic)
Date: Wed, 27 Feb 2019 11:33:55 +0100
Subject: [Freeswitch-users] Call hold issue
In-Reply-To: <003201d4ce7f$9b9504b0$d2bf0e10$@gmail.com>
References: <003201d4ce7f$9b9504b0$d2bf0e10$@gmail.com>
Message-ID: <CAND18T01EmqY-87i1w9zOv2tb3ioz-2o3rKOJRiRbEFtYm1y6w@mail.gmail.com>

I think this is a standard behaviour,
so sendonly party will receive MOH or silence if there is no moh and other
bridged side will still hear him

On Wed, Feb 27, 2019 at 10:34 AM <igor.potjevlesch at gmail.com> wrote:

> Hello,
>
>
>
> I have a weird behaviour with a call hold scenario. When the customer on
> leg A is placing the call on-hold (by sending a RE INVITE with sendonly
> attribute in the SDP), Freeswitch is switching to hold: "Callstate Change
> ACTIVE -> HELD". But only on the leg A.
>
> The holding is not forwarded  to the leg B. So, it causes the called party
> not to hear the Music On Hold generated by the calling party.
>
>
>
> Any idea of what could result in this behaviour?
>
>
>
> Regards,
>
>
>
> Igor.
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com



-- 
Regards,
Mirko
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From stefano.bertuola at gmail.com  Wed Feb 27 16:01:22 2019
From: stefano.bertuola at gmail.com (Stefano Bertuola)
Date: Wed, 27 Feb 2019 17:01:22 +0100
Subject: [Freeswitch-users] iPhone (AMR) to PSTN (G.711) silent call issue -
	dropped
Message-ID: <CAHeTMT6xbCnfYH4wyRs9p77B+O+sCWUbYh=H8Si1=TvuSM-yGQ@mail.gmail.com>

Hi experts.

I am using FS for transcoding a call between an iPhone (AMR codec used) to
a Cisco 880 Voice simulating PSTN (a legacy phone is connected here).

A call is initiated by iPhone and terminated to the fixed phone; if the B
side is silent (VAD enabled), the Cisco 880 stops sending RTP towards FS
and, then, FS to iPhone.

After 10sec not receiving RTPs, iPhone drops the call with "Reason:
SIP;cause=200;text="RTP Timeout" ".

I tried several options to make the FS not stopping sending RTPs towards
iPhone or any other possibility to avoid this issue, but I was not able to
get a workaround.

Any idea?

Br. Stefano
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From vldtoma at gmail.com  Wed Feb 27 16:03:57 2019
From: vldtoma at gmail.com (Vlad Toma)
Date: Wed, 27 Feb 2019 18:03:57 +0200
Subject: [Freeswitch-users] iPhone (AMR) to PSTN (G.711) silent call
 issue - dropped
In-Reply-To: <CAHeTMT6xbCnfYH4wyRs9p77B+O+sCWUbYh=H8Si1=TvuSM-yGQ@mail.gmail.com>
References: <CAHeTMT6xbCnfYH4wyRs9p77B+O+sCWUbYh=H8Si1=TvuSM-yGQ@mail.gmail.com>
Message-ID: <CAFLU3XhvBbb9LOYUpgDmi89JKXbVTemWf7gTOgD+B2FjPTcqJg@mail.gmail.com>

Did you try to disable VAD?

On Wed, Feb 27, 2019, 6:02 PM Stefano Bertuola <stefano.bertuola at gmail.com>
wrote:

> Hi experts.
>
> I am using FS for transcoding a call between an iPhone (AMR codec used) to
> a Cisco 880 Voice simulating PSTN (a legacy phone is connected here).
>
> A call is initiated by iPhone and terminated to the fixed phone; if the B
> side is silent (VAD enabled), the Cisco 880 stops sending RTP towards FS
> and, then, FS to iPhone.
>
> After 10sec not receiving RTPs, iPhone drops the call with "Reason:
> SIP;cause=200;text="RTP Timeout" ".
>
> I tried several options to make the FS not stopping sending RTPs towards
> iPhone or any other possibility to avoid this issue, but I was not able to
> get a workaround.
>
> Any idea?
>
> Br. Stefano
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From stefano.bertuola at gmail.com  Wed Feb 27 16:54:31 2019
From: stefano.bertuola at gmail.com (Stefano Bertuola)
Date: Wed, 27 Feb 2019 17:54:31 +0100
Subject: [Freeswitch-users] iPhone (AMR) to PSTN (G.711) silent call
 issue - dropped
In-Reply-To: <CAFLU3XhvBbb9LOYUpgDmi89JKXbVTemWf7gTOgD+B2FjPTcqJg@mail.gmail.com>
References: <CAHeTMT6xbCnfYH4wyRs9p77B+O+sCWUbYh=H8Si1=TvuSM-yGQ@mail.gmail.com>
 <CAFLU3XhvBbb9LOYUpgDmi89JKXbVTemWf7gTOgD+B2FjPTcqJg@mail.gmail.com>
Message-ID: <CAHeTMT7nkk1+gRA95Aaab-t1oRFHmTLf-7LuZtucrQ2+-=4sbQ@mail.gmail.com>

About VAD the documentation is not really clear.

I tried to place in dialplan following lines, but I am not really sure if
they are correct:

<action application="set" data="bridge_generate_comfort_noise=true"/>
<action application="set" data="rtp_disable_vad_in=true"/>
<action application="set" data="rtp_disable_vad_out=true"/>


>
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From vldtoma at gmail.com  Wed Feb 27 17:18:49 2019
From: vldtoma at gmail.com (Vlad Toma)
Date: Wed, 27 Feb 2019 19:18:49 +0200
Subject: [Freeswitch-users] iPhone (AMR) to PSTN (G.711) silent call
 issue - dropped
In-Reply-To: <CAHeTMT7nkk1+gRA95Aaab-t1oRFHmTLf-7LuZtucrQ2+-=4sbQ@mail.gmail.com>
References: <CAHeTMT6xbCnfYH4wyRs9p77B+O+sCWUbYh=H8Si1=TvuSM-yGQ@mail.gmail.com>
 <CAFLU3XhvBbb9LOYUpgDmi89JKXbVTemWf7gTOgD+B2FjPTcqJg@mail.gmail.com>
 <CAHeTMT7nkk1+gRA95Aaab-t1oRFHmTLf-7LuZtucrQ2+-=4sbQ@mail.gmail.com>
Message-ID: <CAFLU3XiQUOJ2B-yXp5AkKcO54uKM=g9Q9GvOywyVOWafQ4nwzQ@mail.gmail.com>

Hope this helps :
VAD in FreeSWITCH

VAD can be set in endpoint profiles and can have 4 values:

   - in - turn on VAD for incoming media,
   - out - turn on VAD for outgoing media,
   - both - turn on VAD for both incoming and outgoing media,
   - none - VAD is completely turned off.




On Wed, Feb 27, 2019, 6:55 PM Stefano Bertuola <stefano.bertuola at gmail.com>
wrote:

> About VAD the documentation is not really clear.
>
> I tried to place in dialplan following lines, but I am not really sure if
> they are correct:
>
> <action application="set" data="bridge_generate_comfort_noise=true"/>
> <action application="set" data="rtp_disable_vad_in=true"/>
> <action application="set" data="rtp_disable_vad_out=true"/>
>
>
>> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From dragos at freeswitch.org  Wed Feb 27 18:54:54 2019
From: dragos at freeswitch.org (Dragos Oancea)
Date: Wed, 27 Feb 2019 18:54:54 +0000
Subject: [Freeswitch-users] iPhone (AMR) to PSTN (G.711) silent call
 issue - dropped
In-Reply-To: <CAFLU3XiQUOJ2B-yXp5AkKcO54uKM=g9Q9GvOywyVOWafQ4nwzQ@mail.gmail.com>
References: <CAHeTMT6xbCnfYH4wyRs9p77B+O+sCWUbYh=H8Si1=TvuSM-yGQ@mail.gmail.com>
 <CAFLU3XhvBbb9LOYUpgDmi89JKXbVTemWf7gTOgD+B2FjPTcqJg@mail.gmail.com>
 <CAHeTMT7nkk1+gRA95Aaab-t1oRFHmTLf-7LuZtucrQ2+-=4sbQ@mail.gmail.com>
 <CAFLU3XiQUOJ2B-yXp5AkKcO54uKM=g9Q9GvOywyVOWafQ4nwzQ@mail.gmail.com>
Message-ID: <CAGvJV4pFMbj9SRio4YX3VjzSP2r-UDdE+KM+dkP0afSawCFeUA@mail.gmail.com>

Hi Stefano,

You will need probably this:
https://freeswitch.org/confluence/display/FREESWITCH/bridge_generate_comfort_noise

http://lists.freeswitch.org/pipermail/freeswitch-users/2017-December/128510.html

Cheers,
Dragos

On Wed, Feb 27, 2019 at 5:56 PM Vlad Toma <vldtoma at gmail.com> wrote:

> Hope this helps :
> VAD in FreeSWITCH
>
> VAD can be set in endpoint profiles and can have 4 values:
>
>    - in - turn on VAD for incoming media,
>    - out - turn on VAD for outgoing media,
>    - both - turn on VAD for both incoming and outgoing media,
>    - none - VAD is completely turned off.
>
>
>
>
> On Wed, Feb 27, 2019, 6:55 PM Stefano Bertuola <stefano.bertuola at gmail.com>
> wrote:
>
>> About VAD the documentation is not really clear.
>>
>> I tried to place in dialplan following lines, but I am not really sure if
>> they are correct:
>>
>> <action application="set" data="bridge_generate_comfort_noise=true"/>
>> <action application="set" data="rtp_disable_vad_in=true"/>
>> <action application="set" data="rtp_disable_vad_out=true"/>
>>
>>
>>> _________________________________________________________________________
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From therebel22 at gmail.com  Wed Feb 27 12:40:50 2019
From: therebel22 at gmail.com (Marc S)
Date: Wed, 27 Feb 2019 13:40:50 +0100
Subject: [Freeswitch-users] Modify SIP timers (T1X64) to improve failover
Message-ID: <CAMs117WAfHex6OAiNYZGf+fkiVFttC7uMqNssx9fJOsxUfNrcw@mail.gmail.com>

Hello,

I'm testing gateways failover with several gateway for bridge AND ping
setting. It works but INVITE timeout and SIP Options timeout is quite long
(32 seconds) because of timer-T1X64 setting (it's what i hav readen)

I would like to modify timer-T1X64, for example : 3000. Invite failover
would and SIP option timout would now be : 3 seconds. Tests are OK, Invite
failover and gateway marked down is very quick.

My question : is there any impact on behavior of other transactions in SIP
protocol ? Is there no risk to reduce timer-T1X64 from 32000 to 3000 ?

Thanks,
Best regards,

Marc
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From gmaruzz at gmail.com  Wed Feb 27 22:34:14 2019
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Wed, 27 Feb 2019 23:34:14 +0100
Subject: [Freeswitch-users] iPhone (AMR) to PSTN (G.711) silent call
 issue - dropped
In-Reply-To: <CAGvJV4pFMbj9SRio4YX3VjzSP2r-UDdE+KM+dkP0afSawCFeUA@mail.gmail.com>
References: <CAHeTMT6xbCnfYH4wyRs9p77B+O+sCWUbYh=H8Si1=TvuSM-yGQ@mail.gmail.com>
 <CAFLU3XhvBbb9LOYUpgDmi89JKXbVTemWf7gTOgD+B2FjPTcqJg@mail.gmail.com>
 <CAHeTMT7nkk1+gRA95Aaab-t1oRFHmTLf-7LuZtucrQ2+-=4sbQ@mail.gmail.com>
 <CAFLU3XiQUOJ2B-yXp5AkKcO54uKM=g9Q9GvOywyVOWafQ4nwzQ@mail.gmail.com>
 <CAGvJV4pFMbj9SRio4YX3VjzSP2r-UDdE+KM+dkP0afSawCFeUA@mail.gmail.com>
Message-ID: <CALXCt0o5gCUu_DspA3H4vY8pnUnvzkg1p29kxMShAV6NsjE=jw@mail.gmail.com>

and, if you are recording:
https://freeswitch.org/confluence/display/FREESWITCH/record_waste_resources



On Wed, Feb 27, 2019 at 9:01 PM Dragos Oancea <dragos at freeswitch.org> wrote:

> Hi Stefano,
>
> You will need probably this:
>
> https://freeswitch.org/confluence/display/FREESWITCH/bridge_generate_comfort_noise
>
>
> http://lists.freeswitch.org/pipermail/freeswitch-users/2017-December/128510.html
>
> Cheers,
> Dragos
>
> On Wed, Feb 27, 2019 at 5:56 PM Vlad Toma <vldtoma at gmail.com> wrote:
>
>> Hope this helps :
>> VAD in FreeSWITCH
>>
>> VAD can be set in endpoint profiles and can have 4 values:
>>
>>    - in - turn on VAD for incoming media,
>>    - out - turn on VAD for outgoing media,
>>    - both - turn on VAD for both incoming and outgoing media,
>>    - none - VAD is completely turned off.
>>
>>
>>
>>
>> On Wed, Feb 27, 2019, 6:55 PM Stefano Bertuola <
>> stefano.bertuola at gmail.com> wrote:
>>
>>> About VAD the documentation is not really clear.
>>>
>>> I tried to place in dialplan following lines, but I am not really sure
>>> if they are correct:
>>>
>>> <action application="set" data="bridge_generate_comfort_noise=true"/>
>>> <action application="set" data="rtp_disable_vad_in=true"/>
>>> <action application="set" data="rtp_disable_vad_out=true"/>
>>>
>>>
>>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Services
>>> sales at freeswitch.com
>>> https://freeswitch.com
>>>
>>> Official FreeSWITCH Sites
>>> https://freeswitch.com/oss
>>> https://freeswitch.org/confluence
>>> https://cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> https://freeswitch.com
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com



-- 
Sincerely,

Giovanni Maruzzelli
OpenTelecom.IT
cell: +39 347 266 56 18
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From rajil.s at gmail.com  Thu Feb 28 04:08:42 2019
From: rajil.s at gmail.com (Rajil Saraswat)
Date: Wed, 27 Feb 2019 22:08:42 -0600
Subject: [Freeswitch-users] Zoiper goes crazy with registrations RPORT
	bug?
In-Reply-To: <c2faed6b-01e2-1170-6bf2-6fe70862316a@gmail.com>
References: <be1f4cc8-37f0-5af2-abf8-6f04ae3d1314@gmail.com>
 <c2faed6b-01e2-1170-6bf2-6fe70862316a@gmail.com>
Message-ID: <CAFoaQoBke4FfWxU9Deb5cLSxtcAqV123P=_UZTq7Q+Sp00myFw@mail.gmail.com>

Can somebody from Freeswitch dev team confirm if this is a legitimate bug?

Thanks

On Mon, Feb 25, 2019, 8:12 PM Rajil Saraswat <rajil.s at gmail.com> wrote:

> On 2/21/19 8:05 PM, Rajil Saraswat wrote:
> >   Hello,
> >
> > I have a Zoiper android client with Freeswitch server version
> > 1.8.4-5-749a6e108b~64bit. If i select RPORT option in Zoiper along
> > with TLS/TCP transport protocol, i get hundreds for registration
> > requests per minute. Zoiper customer support thinks that Freeswitch is
> > at fault here. The message i received from them is below. Is there any
> > fix for this?
> >
> > ------------
> >
> > The issue in your case is that the topmost Via header does not have
> > the mandatory received parameter.
> >
> > According to RFC 3581 (An Extension to the Session Initiation Protocol
> > (SIP) for Symmetric Response Routing) Section 4. Server Behavior:
> >
> > "When a server compliant to this specification (which can be a proxy
> > or UAS) receives a request, it examines the topmost Via header field
> > value. If this Via header field value contains an "rport" parameter
> > with no value, it MUST set the value of the parameter to the source
> > port of the request. This is analogous to the way in which a server
> > will insert the "received" parameter into the topmost Via header
> > field value. * In fact, the server MUST insert a "received" parameter*
> > containing the source IP address that the request came from, even if
> > it is identical to the value of the "sent-by" component. Note that
> > this processing takes place independent of the transport protocol."
> >
> > This is a server side issue and it is related most likely to the
> > version of FreeSWITCH that you are using.
> >
> > ----------
> >
> > Thanks
> >
> Anybody else using Zoiper and facing this issue or is it just me?
>
> Thanks
>
>
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From bipin at xbipin.com  Thu Feb 28 04:42:46 2019
From: bipin at xbipin.com (Bipin Patel)
Date: Thu, 28 Feb 2019 08:42:46 +0400
Subject: [Freeswitch-users] Zoiper goes crazy with registrations RPORT bug?
In-Reply-To: <CAFoaQoBke4FfWxU9Deb5cLSxtcAqV123P=_UZTq7Q+Sp00myFw@mail.gmail.com>
References: <be1f4cc8-37f0-5af2-abf8-6f04ae3d1314@gmail.com>
 <c2faed6b-01e2-1170-6bf2-6fe70862316a@gmail.com>
 <CAFoaQoBke4FfWxU9Deb5cLSxtcAqV123P=_UZTq7Q+Sp00myFw@mail.gmail.com>
Message-ID: <16932697188.279b.b07ebdf329620b8089087c7205b03f01@xbipin.com>

This is not a bug and works fine at my side using zoiper on a TLS profile


On February 28, 2019 8:27:41 AM Rajil Saraswat <rajil.s at gmail.com> wrote:
> Can somebody from Freeswitch dev team confirm if this is a legitimate bug?
>
> Thanks
>
> On Mon, Feb 25, 2019, 8:12 PM Rajil Saraswat <rajil.s at gmail.com> wrote:
> On 2/21/19 8:05 PM, Rajil Saraswat wrote:
>>   Hello,
>>
>> I have a Zoiper android client with Freeswitch server version
>> 1.8.4-5-749a6e108b~64bit. If i select RPORT option in Zoiper along
>> with TLS/TCP transport protocol, i get hundreds for registration
>> requests per minute. Zoiper customer support thinks that Freeswitch is
>> at fault here. The message i received from them is below. Is there any
>> fix for this?
>>
>> ------------
>>
>> The issue in your case is that the topmost Via header does not have
>> the mandatory received parameter.
>>
>> According to RFC 3581 (An Extension to the Session Initiation Protocol
>> (SIP) for Symmetric Response Routing) Section 4. Server Behavior:
>>
>> "When a server compliant to this specification (which can be a proxy
>> or UAS) receives a request, it examines the topmost Via header field
>> value. If this Via header field value contains an "rport" parameter
>> with no value, it MUST set the value of the parameter to the source
>> port of the request. This is analogous to the way in which a server
>> will insert the "received" parameter into the topmost Via header
>> field value. * In fact, the server MUST insert a "received" parameter*
>> containing the source IP address that the request came from, even if
>> it is identical to the value of the "sent-by" component. Note that
>> this processing takes place independent of the transport protocol."
>>
>> This is a server side issue and it is related most likely to the
>> version of FreeSWITCH that you are using.
>>
>> ----------
>>
>> Thanks
>>
> Anybody else using Zoiper and facing this issue or is it just me?
>
> Thanks
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com

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From stefano.bertuola at gmail.com  Thu Feb 28 10:20:24 2019
From: stefano.bertuola at gmail.com (Stefano Bertuola)
Date: Thu, 28 Feb 2019 11:20:24 +0100
Subject: [Freeswitch-users] iPhone (AMR) to PSTN (G.711) silent call
 issue - dropped
In-Reply-To: <CALXCt0o5gCUu_DspA3H4vY8pnUnvzkg1p29kxMShAV6NsjE=jw@mail.gmail.com>
References: <CAHeTMT6xbCnfYH4wyRs9p77B+O+sCWUbYh=H8Si1=TvuSM-yGQ@mail.gmail.com>
 <CAFLU3XhvBbb9LOYUpgDmi89JKXbVTemWf7gTOgD+B2FjPTcqJg@mail.gmail.com>
 <CAHeTMT7nkk1+gRA95Aaab-t1oRFHmTLf-7LuZtucrQ2+-=4sbQ@mail.gmail.com>
 <CAFLU3XiQUOJ2B-yXp5AkKcO54uKM=g9Q9GvOywyVOWafQ4nwzQ@mail.gmail.com>
 <CAGvJV4pFMbj9SRio4YX3VjzSP2r-UDdE+KM+dkP0afSawCFeUA@mail.gmail.com>
 <CALXCt0o5gCUu_DspA3H4vY8pnUnvzkg1p29kxMShAV6NsjE=jw@mail.gmail.com>
Message-ID: <CAHeTMT6SEXGypQ8uFFNY6isRS5TZAuuO-Ym-PgWzQ60dqUBriQ@mail.gmail.com>

I tried most of the combinations of below settings, but the result the
same: when the RTP stops to be received from Cisco Gateway, also it is
stopped towards iPhone.

Maybe I am missing the right combinations of parameters?

<action application="set" data="bridge_generate_comfort_noise=true"/>
<!--
<action application="set" data="rtp_disable_vad_in=true"/>
<action application="set" data="rtp_disable_vad_out=true"/>
-->
<action application="set" data="record_waste_resources=true"/>
<action application="set" data="rtp_enable_vad_in=false"/>
<action application="set" data="rtp_enable_vad_out=false"/>
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From igor.potjevlesch at gmail.com  Thu Feb 28 14:13:36 2019
From: igor.potjevlesch at gmail.com (igor.potjevlesch at gmail.com)
Date: Thu, 28 Feb 2019 15:13:36 +0100
Subject: [Freeswitch-users] Call hold issue
In-Reply-To: <CAND18T01EmqY-87i1w9zOv2tb3ioz-2o3rKOJRiRbEFtYm1y6w@mail.gmail.com>
References: <003201d4ce7f$9b9504b0$d2bf0e10$@gmail.com>
 <CAND18T01EmqY-87i1w9zOv2tb3ioz-2o3rKOJRiRbEFtYm1y6w@mail.gmail.com>
Message-ID: <001d01d4cf6f$cbf55740$63e005c0$@gmail.com>

Not sure to understand your answer. The sendonly party is sending the MOH. Freeswitch switch on recvonly. But leg B has a silence. 

Is it normal?

 

Regards,

 

Igor.

 

De : FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> De la part de Mirko Brankovic
Envoyé : mercredi 27 février 2019 11:34
À : FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Objet : Re: [Freeswitch-users] Call hold issue

 

I think this is a standard behaviour,

so sendonly party will receive MOH or silence if there is no moh and other bridged side will still hear him

 

On Wed, Feb 27, 2019 at 10:34 AM <igor.potjevlesch at gmail.com <mailto:igor.potjevlesch at gmail.com> > wrote:

Hello,

 

I have a weird behaviour with a call hold scenario. When the customer on leg A is placing the call on-hold (by sending a RE INVITE with sendonly attribute in the SDP), Freeswitch is switching to hold: "Callstate Change ACTIVE -> HELD". But only on the leg A. 

The holding is not forwarded  to the leg B. So, it causes the called party not to hear the Music On Hold generated by the calling party.

 

Any idea of what could result in this behaviour?

 

Regards,

 

Igor.

_________________________________________________________________________
Professional FreeSWITCH Services
sales at freeswitch.com <mailto:sales at freeswitch.com> 
https://freeswitch.com

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

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




 

-- 

Regards,

Mirko

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From dragos at freeswitch.org  Thu Feb 28 16:05:51 2019
From: dragos at freeswitch.org (Dragos Oancea)
Date: Thu, 28 Feb 2019 16:05:51 +0000
Subject: [Freeswitch-users] iPhone (AMR) to PSTN (G.711) silent call
 issue - dropped
In-Reply-To: <CAHeTMT6SEXGypQ8uFFNY6isRS5TZAuuO-Ym-PgWzQ60dqUBriQ@mail.gmail.com>
References: <CAHeTMT6xbCnfYH4wyRs9p77B+O+sCWUbYh=H8Si1=TvuSM-yGQ@mail.gmail.com>
 <CAFLU3XhvBbb9LOYUpgDmi89JKXbVTemWf7gTOgD+B2FjPTcqJg@mail.gmail.com>
 <CAHeTMT7nkk1+gRA95Aaab-t1oRFHmTLf-7LuZtucrQ2+-=4sbQ@mail.gmail.com>
 <CAFLU3XiQUOJ2B-yXp5AkKcO54uKM=g9Q9GvOywyVOWafQ4nwzQ@mail.gmail.com>
 <CAGvJV4pFMbj9SRio4YX3VjzSP2r-UDdE+KM+dkP0afSawCFeUA@mail.gmail.com>
 <CALXCt0o5gCUu_DspA3H4vY8pnUnvzkg1p29kxMShAV6NsjE=jw@mail.gmail.com>
 <CAHeTMT6SEXGypQ8uFFNY6isRS5TZAuuO-Ym-PgWzQ60dqUBriQ@mail.gmail.com>
Message-ID: <CAGvJV4p4ypyZKheqT5vCcG72a9paTpgu=xTbanb324kCrAOWjQ@mail.gmail.com>

Stefano, this could be a bug. Please open a jira and attach a pcap trace:
https://freeswitch.org/confluence/display/FREESWITCH/Reporting+Bugs+to+JIRA



On Thu, Feb 28, 2019 at 10:47 AM Stefano Bertuola <
stefano.bertuola at gmail.com> wrote:

> I tried most of the combinations of below settings, but the result the
> same: when the RTP stops to be received from Cisco Gateway, also it is
> stopped towards iPhone.
>
> Maybe I am missing the right combinations of parameters?
>
> <action application="set" data="bridge_generate_comfort_noise=true"/>
> <!--
> <action application="set" data="rtp_disable_vad_in=true"/>
> <action application="set" data="rtp_disable_vad_out=true"/>
> -->
> <action application="set" data="record_waste_resources=true"/>
> <action application="set" data="rtp_enable_vad_in=false"/>
> <action application="set" data="rtp_enable_vad_out=false"/>
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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