From ksrigo at gmail.com  Sun Apr  1 07:48:12 2018
From: ksrigo at gmail.com (Srigo Kanapathipillai)
Date: Sun, 01 Apr 2018 07:48:12 +0000
Subject: [Freeswitch-users] how to access the XML_REQUEST in mod_python
In-Reply-To: <CACgE3xunkzv9OA1jECQ8XDQuPKuXfsEnjU_utgwkzAnMLEL+KQ@mail.gmail.com>
References: <CACgE3xunkzv9OA1jECQ8XDQuPKuXfsEnjU_utgwkzAnMLEL+KQ@mail.gmail.com>
Message-ID: <CAAiAxuzo11HuxSP=chux7mTRycROgCSjDGC0h9vKXqVT1Sg9FQ@mail.gmail.com>

Hi

If you want to have dynamic directory, i suggest you to use mod_xml_curl
which will do an http request to your web server to get ur directory info.
Your webserver serving your fs config can be a python process (fcgiwrap,
uwsgi, gunicorn, etc with a nginx or apache for production environnement).

Srigo

On Sat, 31 Mar 2018, 06:23 Do Nguyen Ha, <donguyenha at gmail.com> wrote:

> hi list
>
> i want to use mod_python to dynamic load user directory and read the wiki
> but no documents for access XML_REQUEST
>
> how to get the XML_REQUEST in python script
>
> thank you
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 donguyenha at gmail.com  Sun Apr  1 15:08:36 2018
From: donguyenha at gmail.com (Do Nguyen Ha)
Date: Sun, 1 Apr 2018 22:08:36 +0700
Subject: [Freeswitch-users] how to access the XML_REQUEST in mod_python
In-Reply-To: <CAAiAxuzo11HuxSP=chux7mTRycROgCSjDGC0h9vKXqVT1Sg9FQ@mail.gmail.com>
References: <CACgE3xunkzv9OA1jECQ8XDQuPKuXfsEnjU_utgwkzAnMLEL+KQ@mail.gmail.com>
 <CAAiAxuzo11HuxSP=chux7mTRycROgCSjDGC0h9vKXqVT1Sg9FQ@mail.gmail.com>
Message-ID: <CACgE3xtChRMHMnnry7tCUyf6qqFeppXYCTCB=NMMX7M3YxJBVA@mail.gmail.com>

hi Srigo

thank you for your suggestion, i did read the mod_xml_curl documents but i
refer the mod_lua/mod_python :)

any idea for the mod_python

On Sun, Apr 1, 2018 at 2:48 PM, Srigo Kanapathipillai <ksrigo at gmail.com>
wrote:

> Hi
>
> If you want to have dynamic directory, i suggest you to use mod_xml_curl
> which will do an http request to your web server to get ur directory info.
> Your webserver serving your fs config can be a python process (fcgiwrap,
> uwsgi, gunicorn, etc with a nginx or apache for production environnement).
>
> Srigo
>
> On Sat, 31 Mar 2018, 06:23 Do Nguyen Ha, <donguyenha at gmail.com> wrote:
>
>> hi list
>>
>> i want to use mod_python to dynamic load user directory and read the wiki
>> but no documents for access XML_REQUEST
>>
>> how to get the XML_REQUEST in python script
>>
>> thank you
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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 tculjaga at gmail.com  Sun Apr  1 18:07:36 2018
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Sun, 1 Apr 2018 20:07:36 +0200
Subject: [Freeswitch-users] postgres db contains non existent channels
In-Reply-To: <6C59B394-8BCE-4316-AE52-D2883ECB23D0@dissauer.net>
References: <6C59B394-8BCE-4316-AE52-D2883ECB23D0@dissauer.net>
Message-ID: <CAJoKKdWt2_PXnhV81ccDiEe_vjiyV8+cn=TzTMSxO9Cm=fjSHQ@mail.gmail.com>

Hi,

 can you capture what fs is trying to save ?
do you have a sip message capture in wireshark ?



On 29 March 2018 at 20:14, Roman Dissauer <roman at dissauer.net> wrote:

> Hi,
>
> we have about 800 connected users, 150 simultaneous calls and some user
> agents send sip messages with non UTF-8 characters.
> When FreeSWITCH tries to alter the channel information in the postgres db
> it throughs an error:
>
> 2018-03-29 15:47:46.794346 [ERR] switch_pgsql.c:656 Error executing query:
> ERROR:  invalid byte sequence for encoding "UTF8": 0xf6 0x63 0x68 0x6c
>
> The Problem now is that we see some calls in the database which do not
> exist anymore (uuid_exists returns false). This are other calls, not the
> one which cause the error above!
> It seems that everytime the error happens, Freeswitch cannot delete other
> channels from db when the bye is arriving +/- 100ms near the error.
>
> We tried to set client_encoding in the postgres connection string to some
> options e.g. LATIN1 but that didn’t solve our problem. Whichever
> client_encoding we used, we always had some user agents with some unknown
> characters.
>
> How should we deal with that?
>
> Thanks,
> Roman
>
> *RDI SOLUTIONS e.U.*
> Hollenthon 105
> 2812 Hollenthon, Österreich
> T: +43 1 3530349 <+43%201%203530349> - 10
> F: +43 1 3530349 <+43%201%203530349> - 99
> roman.dissauer at rdi.at
> *www.rdi.at <http://www.rdi.at>*
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 tudor.gabriell at gmail.com  Sun Apr  1 18:20:06 2018
From: tudor.gabriell at gmail.com (Adridan Tudor)
Date: Sun, 1 Apr 2018 21:20:06 +0300
Subject: [Freeswitch-users] Conference auto-outcall to a callcenter number
	with verto agents
Message-ID: <CAFLFN6TZrASi=N9GP--_g0xvJ-c0mYbf3D4AZk-vgFwa_VZw6g@mail.gmail.com>

Hello,

The outcall works and agents join the conference but the verto communicator
agents don't see the other participants. mod_verto is not sending the
conference-liveArray-join verto.event to the agents. If I auto-outcall
directly the agents with (user/xxx@${domain}) everything works.

My setup is like this:

<action application="conference_set_auto_outcall"
data="{conference_member_flags=moderator}sofia/internal/799@${domain}"/>
<action application="conference"
data="conf-${caller_id_number}@video-mcu-stereo"/>

video-mcu-stereo profile has:
<param name="conference-flags"
value="video-floor-only|rfc-4579|minimize-video-encoding|livearray-json-status|json-events|livearray-sync"/>

<extension name="my_callcenter">
  <condition field="destination_number" expression="799">
    <action application="set" data="hangup_after_bridge=true"/>
    <action application="set" data="continue_on_fail=true"/>
    <action application="set" data="originate_continue_on_timeout=true"/>
    <action application="set" data="call_timeout=15"/>
    <action application="callcenter" data="my_callcenter at x.x.x.x"/>
  </condition>
</extension>

in directory/default.xml I have:
<param name="jsonrpc-allowed-methods" value="verto"/>
<param name="jsonrpc-allowed-event-channels"
value="demo,conference,presence,conference-liveArray,conference-mod"/>

Any help is highly appreciated!

Thanks,
Tudor
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From shane.mitchell at fonedynamics.com.au  Tue Apr  3 07:26:16 2018
From: shane.mitchell at fonedynamics.com.au (Shane Mitchell)
Date: Tue, 3 Apr 2018 07:26:16 +0000
Subject: [Freeswitch-users] Bridge and+or carrier failover
Message-ID: <SYXPR01MB14233967A0341FDE0DFE3C37CAA50@SYXPR01MB1423.ausprd01.prod.outlook.com>

Hi,

I'm trying to implement carrier failover with the bridge application, specifically when working with multiple simultaneous destinations.

For a single destination, I can use OR (|) and fail_on_single_reject to appropriately failover.  For example (simplified):

{fail_on_single_reject=user_busy}sofia/gateway/GW1/ENDPOINT1|sofia/gateway/GW2/ENDPOINT1

However with multiple simultaneous destinations, I'm having issues with approach.  I've tried the following (simplified):

{fail_on_single_reject=user_busy}sofia/gateway/GW1/ENDPOINT1|sofia/gateway/GW2/ENDPOINT1,sofia/gateway/GW3/ENDPOINT2|sofia/gateway/GW4/ENDPOINT2

This terminates all new channels when any one of them fails.

What I'm trying to do is to stop bridge attempting a specific OR chain when one fails, but to continue executing the rest of the ANDs.

Is there a way to achieve bridging to multiple destinations simultaneously, with each destination having its own list of failover gateways, and having failover for a destination only occur on certain cause codes (i.e. we don't want to fail over on user_busy/etc, but we do on other types of failures)?

Thanks & regards,
Shane Mitchell.

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From alex at freeswitch.com  Tue Apr  3 08:44:04 2018
From: alex at freeswitch.com (Alexey Sibyakin)
Date: Tue, 3 Apr 2018 17:44:04 +0900
Subject: [Freeswitch-users] Bridge and+or carrier failover
In-Reply-To: <SYXPR01MB14233967A0341FDE0DFE3C37CAA50@SYXPR01MB1423.ausprd01.prod.outlook.com>
References: <SYXPR01MB14233967A0341FDE0DFE3C37CAA50@SYXPR01MB1423.ausprd01.prod.outlook.com>
Message-ID: <CAGJLwFiqFGYbBb0Z83t8wqT1qOEa9h1B5DX=8s=EWPvFnmSbuA@mail.gmail.com>

Hi,

Do you have ignore_early_media=true in your dialplan? Probably you should.

https://freeswitch.org/confluence/display/FREESWITCH/mod_dptools:+bridge

Alex

On Tue, Apr 3, 2018 at 4:26 PM, Shane Mitchell <
shane.mitchell at fonedynamics.com.au> wrote:

> Hi,
>
>
>
> I’m trying to implement carrier failover with the bridge application,
> specifically when working with multiple simultaneous destinations.
>
>
>
> For a single destination, I can use OR (|) and fail_on_single_reject to
> appropriately failover.  For example (simplified):
>
>
>
> {fail_on_single_reject=user_busy}sofia/gateway/GW1/
> ENDPOINT1|sofia/gateway/GW2/ENDPOINT1
>
>
>
> However with multiple simultaneous destinations, I’m having issues with
> approach.  I’ve tried the following (simplified):
>
>
>
> {fail_on_single_reject=user_busy}sofia/gateway/GW1/
> ENDPOINT1|sofia/gateway/GW2/ENDPOINT1,sofia/gateway/GW3/
> ENDPOINT2|sofia/gateway/GW4/ENDPOINT2
>
>
>
> This terminates all new channels when any one of them fails.
>
>
>
> What I’m trying to do is to stop bridge attempting a specific OR chain
> when one fails, but to continue executing the rest of the ANDs.
>
>
>
> Is there a way to achieve bridging to multiple destinations
> simultaneously, with each destination having its own list of failover
> gateways, and having failover for a destination only occur on certain cause
> codes (i.e. we don’t want to fail over on user_busy/etc, but we do on other
> types of failures)?
>
>
>
> Thanks & regards,
>
> Shane Mitchell.
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 Engineer
FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045
<https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g>
Email: alex at freeswitch.com
Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/>
Need commercial support? Contact sales at freeswitch.com for details.
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From tudor.gabriell at gmail.com  Tue Apr  3 08:51:22 2018
From: tudor.gabriell at gmail.com (Adridan Tudor)
Date: Tue, 3 Apr 2018 11:51:22 +0300
Subject: [Freeswitch-users] Conference auto-outcall to a callcenter
 number with verto agents
In-Reply-To: <CAFLFN6TZrASi=N9GP--_g0xvJ-c0mYbf3D4AZk-vgFwa_VZw6g@mail.gmail.com>
References: <CAFLFN6TZrASi=N9GP--_g0xvJ-c0mYbf3D4AZk-vgFwa_VZw6g@mail.gmail.com>
Message-ID: <1004a6b4-b22a-bd0f-4250-63e02150ee66@gmail.com>

Hello again,

Seems like neither:
- bgapi originate 
{origination_uuid=${create_uuid}}sofia/internal/799 at domain 
&conference($conf-303)
nor
- conference conf-303 dial {originate_timeout=30}sofia/internal/799 at domain
work. The conference is working but the verto communicator agents are 
not invited to join the livearray.

For test purposes I tried this commands and the 
conference_set_auto_outcall using sofia/internal/{user}@domain. Where 
user is a verto_contact. Same result.

Is there any workaround to make this work? I there a way to send 
conference-liveArray-join from FS to verto communicator?

Thanks!

On 01-Apr-18 9:20 PM, tudor.gabriell at gmail.com (Adridan Tudor) wrote:
> Hello,
>
> The outcall works and agents join the conference but the verto communicator
> agents don't see the other participants. mod_verto is not sending the
> conference-liveArray-join verto.event to the agents. If I auto-outcall
> directly the agents with (user/xxx@${domain}) everything works.
>
> My setup is like this:
>
> <action application="conference_set_auto_outcall"
> data="{conference_member_flags=moderator}sofia/internal/799@${domain}"/>
> <action application="conference"
> data="conf-${caller_id_number}@video-mcu-stereo"/>
>
> video-mcu-stereo profile has:
> <param name="conference-flags"
> value="video-floor-only|rfc-4579|minimize-video-encoding|livearray-json-status|json-events|livearray-sync"/>
>
> <extension name="my_callcenter">
>    <condition field="destination_number" expression="799">
>      <action application="set" data="hangup_after_bridge=true"/>
>      <action application="set" data="continue_on_fail=true"/>
>      <action application="set" data="originate_continue_on_timeout=true"/>
>      <action application="set" data="call_timeout=15"/>
>      <action application="callcenter" data="my_callcenter at x.x.x.x"/>
>    </condition>
> </extension>
>
> in directory/default.xml I have:
> <param name="jsonrpc-allowed-methods" value="verto"/>
> <param name="jsonrpc-allowed-event-channels"
> value="demo,conference,presence,conference-liveArray,conference-mod"/>
>
> Any help is highly appreciated!
>
> Thanks,
> Tudor
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From achinthau at gmail.com  Tue Apr  3 12:58:43 2018
From: achinthau at gmail.com (Achintha)
Date: Tue, 3 Apr 2018 18:28:43 +0530
Subject: [Freeswitch-users] Freeswitch Crashed- Segmentation Fault
In-Reply-To: <29BC9A0C-00CE-43D7-9D0D-07610BD347E6@jerris.com>
References: <CA+JwzxsBgrQBzUNR=E4=Xk6WoXy37O-YTqt+V_0UQLZ4i0aekg@mail.gmail.com>
 <29BC9A0C-00CE-43D7-9D0D-07610BD347E6@jerris.com>
Message-ID: <CA+Jwzxv1qdpC7d6qPkeAC2mcXMDh=kuOdyFJsAL9jiVb7xYwSg@mail.gmail.com>

Hi Mr. Michale,

our freeswitch version is  “1.6.20-37-987c9b9~64bit (-37-987c9b9 64bit)”
Operating system is "Debian 8.1 -jessie"
it installed by "apt-get install freeswitch-meta-all" command but debug
symbols not installed.
can you please advice me to install debug symbols on it.

Thanking You.


On Thu, Mar 29, 2018 at 8:56 PM, Michael Jerris <mike at jerris.com> wrote:

> You have no debug symbols installed so it is impossible to tell what is
> going on.  Please install those and reproduce the bug report with debug
> symbols on lira.
>
> On Mar 29, 2018, at 3:16 AM, Achintha <achinthau at gmail.com> wrote:
>
>
> Hi all,
>
> Our freeswitch server handles 3000 – 4000 calls per day. We have custom
> module to rout calls.
>
> Anyway freeswitch was crashed two times I have attached core dump on “
> https://pastebin.freeswitch.org/view/4463025f ”
>
> Our freeswitch version is  “1.6.20-37-987c9b9~64bit (-37-987c9b9 64bit)”
>
> Please advise to solve this matter asap.
>
>
> --
> Best Regards..
> Achintha Udukumbura
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Best Regards..
Achintha Udukumbura
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From shaun.stokes at itec-support.co.uk  Tue Apr  3 14:04:38 2018
From: shaun.stokes at itec-support.co.uk (Shaun Stokes)
Date: Tue, 3 Apr 2018 14:04:38 +0000
Subject: [Freeswitch-users] Freeswitch Crashed- Segmentation Fault
In-Reply-To: <CA+Jwzxv1qdpC7d6qPkeAC2mcXMDh=kuOdyFJsAL9jiVb7xYwSg@mail.gmail.com>
References: <CA+JwzxsBgrQBzUNR=E4=Xk6WoXy37O-YTqt+V_0UQLZ4i0aekg@mail.gmail.com>
 <29BC9A0C-00CE-43D7-9D0D-07610BD347E6@jerris.com>,
 <CA+Jwzxv1qdpC7d6qPkeAC2mcXMDh=kuOdyFJsAL9jiVb7xYwSg@mail.gmail.com>
Message-ID: <1522764277889.93536@itec-support.co.uk>

We've had problems getting debug symbols to work in the past when using the packaged install.


Debug symbols are working when installing from source, providing you include this before ./configure:

export CFLAGS="-g -ggdb"
export MOD_CFLAGS="-g -ggdb"?



________________________________
From: FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> on behalf of Achintha <achinthau at gmail.com>
Sent: 03 April 2018 13:58
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Freeswitch Crashed- Segmentation Fault

Hi Mr. Michale,

our freeswitch version is  "1.6.20-37-987c9b9~64bit (-37-987c9b9 64bit)"  Operating system is "Debian 8.1 -jessie"
it installed by "apt-get install freeswitch-meta-all" command but debug symbols not installed.
can you please advice me to install debug symbols on it.

Thanking You.


On Thu, Mar 29, 2018 at 8:56 PM, Michael Jerris <mike at jerris.com<mailto:mike at jerris.com>> wrote:
You have no debug symbols installed so it is impossible to tell what is going on.  Please install those and reproduce the bug report with debug symbols on lira.

On Mar 29, 2018, at 3:16 AM, Achintha <achinthau at gmail.com<mailto:achinthau at gmail.com>> wrote:


Hi all,
Our freeswitch server handles 3000 - 4000 calls per day. We have custom module to rout calls.
Anyway freeswitch was crashed two times I have attached core dump on " https://pastebin.freeswitch.org/view/4463025f "
Our freeswitch version is  "1.6.20-37-987c9b9~64bit (-37-987c9b9 64bit)"
Please advise to solve this matter asap.

--
Best Regards..
Achintha Udukumbura
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com

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

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


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

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

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



--
Best Regards..
Achintha Udukumbura

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

______________________________________________________________________
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From brian at freeswitch.com  Tue Apr  3 14:14:54 2018
From: brian at freeswitch.com (Brian West)
Date: Tue, 3 Apr 2018 09:14:54 -0500
Subject: [Freeswitch-users] Freeswitch Crashed- Segmentation Fault
In-Reply-To: <1522764277889.93536@itec-support.co.uk>
References: <CA+JwzxsBgrQBzUNR=E4=Xk6WoXy37O-YTqt+V_0UQLZ4i0aekg@mail.gmail.com>
 <29BC9A0C-00CE-43D7-9D0D-07610BD347E6@jerris.com>
 <CA+Jwzxv1qdpC7d6qPkeAC2mcXMDh=kuOdyFJsAL9jiVb7xYwSg@mail.gmail.com>
 <1522764277889.93536@itec-support.co.uk>
Message-ID: <CAEJMVkBsdDnif+OHiU1Qf+sjZ4mO8bYdBuGkNiE-jdGWC_RrmA@mail.gmail.com>

you just have to install the -dbg packages when installing from debian
packaging, There is no need to recompile from src.

If issues persists you can email sales at freeswitch.com and we can see what
our professional services can do for you.

/b


On Tue, Apr 3, 2018 at 9:04 AM, Shaun Stokes <
shaun.stokes at itec-support.co.uk> wrote:

> We've had problems getting debug symbols to work in the past when using
> the packaged install.
>
>
> Debug symbols are working when installing from source, providing you
> include this before ./configure:
>
> export CFLAGS="-g -ggdb"
> export MOD_CFLAGS="-g -ggdb"​
>
>
>
> ------------------------------
> *From:* FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org>
> on behalf of Achintha <achinthau at gmail.com>
> *Sent:* 03 April 2018 13:58
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] Freeswitch Crashed- Segmentation Fault
>
> Hi Mr. Michale,
>
> our freeswitch version is  “1.6.20-37-987c9b9~64bit (-37-987c9b9 64bit)”
> Operating system is "Debian 8.1 -jessie"
> it installed by "apt-get install freeswitch-meta-all" command but debug
> symbols not installed.
> can you please advice me to install debug symbols on it.
>
> Thanking You.
>
>
> On Thu, Mar 29, 2018 at 8:56 PM, Michael Jerris <mike at jerris.com> wrote:
>
>> You have no debug symbols installed so it is impossible to tell what is
>> going on.  Please install those and reproduce the bug report with debug
>> symbols on lira.
>>
>> On Mar 29, 2018, at 3:16 AM, Achintha <achinthau at gmail.com> wrote:
>>
>>
>> Hi all,
>>
>> Our freeswitch server handles 3000 – 4000 calls per day. We have custom
>> module to rout calls.
>>
>> Anyway freeswitch was crashed two times I have attached core dump on “
>> https://pastebin.freeswitch.org/view/4463025f ”
>>
>> Our freeswitch version is  “1.6.20-37-987c9b9~64bit (-37-987c9b9 64bit)”
>>
>> Please advise to solve this matter asap.
>>
>>
>> --
>> Best Regards..
>> Achintha Udukumbura
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Best Regards..
> Achintha Udukumbura
>
> ______________________________________________________________________
> This message has been checked for all known viruses by MessageLabs Virus
> Scanning Service.
> ______________________________________________________________________
>
> ______________________________________________________________________
> This message has been checked for all known viruses by MessageLabs Virus
> Scanning Service.
> ______________________________________________________________________
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

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: color-facebook-96.png] <https://www.facebook.com/freeswitch/>[image:
color-twitter-96.png]
<https://twitter.com/freeswitch?ref_src=twsrc%5Egoogle%7Ctwcamp%5Eserp%7Ctwgr%5Eauthor>
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From ap at gen-ip.fr  Tue Apr  3 09:50:59 2018
From: ap at gen-ip.fr (Alexis Prodhomme)
Date: Tue, 3 Apr 2018 11:50:59 +0200
Subject: [Freeswitch-users] Bridge and+or carrier failover
In-Reply-To: <SYXPR01MB14233967A0341FDE0DFE3C37CAA50@SYXPR01MB1423.ausprd01.prod.outlook.com>
References: <SYXPR01MB14233967A0341FDE0DFE3C37CAA50@SYXPR01MB1423.ausprd01.prod.outlook.com>
Message-ID: <5d64229d-f567-2328-cbe9-02d2c6c9b54f@gen-ip.fr>

Hi,

Did you tried to put fail_on_single_reject in [ ] instead of { } ?

You can find more informations in 
https://freeswitch.org/confluence/display/FREESWITCH/Channel+Variables#ChannelVariables-ChannelVariablesinDialStrings

Alexis Prodhomme

Le 03/04/2018 à 09:26, Shane Mitchell a écrit :
>
> Hi,
>
> I’m trying to implement carrier failover with the bridge application, 
> specifically when working with multiple simultaneous destinations.
>
> For a single destination, I can use OR (|) and fail_on_single_reject 
> to appropriately failover.  For example (simplified):
>
> {fail_on_single_reject=user_busy}sofia/gateway/GW1/ENDPOINT1|sofia/gateway/GW2/ENDPOINT1
>
> However with multiple simultaneous destinations, I’m having issues 
> with approach.  I’ve tried the following (simplified):
>
> {fail_on_single_reject=user_busy}sofia/gateway/GW1/ENDPOINT1|sofia/gateway/GW2/ENDPOINT1,sofia/gateway/GW3/ENDPOINT2|sofia/gateway/GW4/ENDPOINT2
>
> This terminates all new channels when any one of them fails.
>
> What I’m trying to do is to stop bridge attempting a specific OR chain 
> when one fails, but to continue executing the rest of the ANDs.
>
> Is there a way to achieve bridging to multiple destinations 
> simultaneously, with each destination having its own list of failover 
> gateways, and having failover for a destination only occur on certain 
> cause codes (i.e. we don’t want to fail over on user_busy/etc, but we 
> do on other types of failures)?
>
> Thanks & regards,
>
> Shane Mitchell.
>

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From vldtoma at gmail.com  Tue Apr  3 14:39:03 2018
From: vldtoma at gmail.com (Vlad Toma)
Date: Tue, 3 Apr 2018 17:39:03 +0300
Subject: [Freeswitch-users] Question about limit max_calls
Message-ID: <5ac39191.480f1c0a.98619.7465@mx.google.com>

Hello,
I want to restrict the number of incoming calls to my internal extensions to 1 per extension, because I don’t want to receive another call to an extension that is already in a call.
I tried setting the max_calls variable to 1 and adding this to the dialplan 

<extension name="limit_exceeded">
  <condition field="destination_number" expression="^limit_exceeded$">
    <action application="playback" data="/sounds/overthelimit.wav"/>
    <action application="hangup"/>
  </condition>
</extension>
 
<extension name="limit" continue="true">
  <condition>
    <!-- You can set ${max_calls} per user in directory.conf and remove the line below -->
    <action application="set" data="max_calls=1" inline="true"/>
    <action application="limit" data="db $${domain} ${sip_auth_username} ${max_calls}"/>
  </condition>
</extension>

Also I tried adding this to my dialplan where the condition matches my internal extensions 
<action application="limit_execute" data="hash outbound carrier1 1 bridge {ignore_early_media=true}user/${dialed_extension}@${domain_name}" />
 
It works if I dial normally but if I do a conference conf dial {ignore_early_media=true,conference_member_flags=moderator}user/XXX@${domain_name} it always calls even if that user is already in a call.  
How can I fix this? Thank you.


Kind regards,

Vlad Toma
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From michael at mailworks.org  Wed Apr  4 07:16:45 2018
From: michael at mailworks.org (Michael Avers)
Date: Wed, 04 Apr 2018 00:16:45 -0700
Subject: [Freeswitch-users] mod_verto error when using Postgres DSN - no
	such table: json_store
Message-ID: <1522826205.2587950.1325901536.5A66C4B2@webmail.messagingengine.com>

Hello,

I'm using Postgres in the core and everywhere else. Relevant modules
such as callcenter, fifo, etc. all have the ODBC defined and working
with the tables just fine.
The only issue is Verto - I have json_store table in Postgres but Verto
doesn't appear to be accessing it at all whenever mod_verto is loaded or
reloaded. Getting this error:
2018-04-04 07:05:31.695917 [CONSOLE] switch_loadable_module.c:2034
mod_verto unloaded.*2018-04-04 07:05:31.695917 [ERR] switch_core_sqldb.c:587 NATIVE SQL ERR
[no such table: json_store]** **select name from json_store where name=''*
2018-04-04 07:05:31.695917 [INFO] mod_verto.c:4749 Secure key and
cert specified
It says no such table because I did not have the sqlite json.db file,
but after I put it back in it's able to read from it. So clearly it
looks like it's not even trying to access json_store in the Postgres DB.
I even added a <param name="odbc-dsn".. to verto.conf.xml even though
it's not documented anywhere, still no luck.
Any ideas? Is mod_verto not able to access the table when using
Postgres?
FYI this is on latest master and also 1.6.20 production release

Thanks
Mike
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From vbvbrj at gmail.com  Wed Apr  4 08:10:42 2018
From: vbvbrj at gmail.com (Mimiko)
Date: Wed, 4 Apr 2018 11:10:42 +0300
Subject: [Freeswitch-users] Using lcr in lua session and
	fail_on_single_reject problem
In-Reply-To: <3af44928-2411-d6ec-33cf-11e064b39b17@gmail.com>
References: <3af44928-2411-d6ec-33cf-11e064b39b17@gmail.com>
Message-ID: <e1147e6e-9022-4e4e-cf37-357d93f32c7c@gmail.com>

Any idea?

> Hello.
> 
> I want to originate a call using lcr from lua script:
> freeswitch.Session("{ignore_early_media=true,originate_timeout=15,fail_on_single_reject=^^:NO_ANSWER:CALL_REJECTED:NORMAL_CLEARING:USER_BUSY}lcr/default/"..number) 
> 
> 
> Using this command, originate_timeout is not taken into account and all routes are tried regardless of response reason, for example NO_ANSWER.
> If I remove fail_on_single_reject, then originate_timeout works but same problem: all routes are tried, except when the call was answered.
> 
> In diaplan
> <action application="lcr" data="${destination_number} default"/>
> <action application="bridge" data="{fail_on_single_reject=^^:NO_ANSWER:CALL_REJECTED:NORMAL_CLEARING:USER_BUSY}${lcr_auto_route}"/>
> 
> this works fine. Maybe its due to using bridge, not lcr directly. But in lua ${lcr_auto_route} variable is not available. Any suggestion?
> 
> FreeSWITCH Version 1.5.6b+git~20130928T022323Z~6b9382290d (git 6b93822 2013-09-28 02:23:23Z)
> Yes, it is old, but can't update at this time.



From royj at yandex.ru  Wed Apr  4 14:17:10 2018
From: royj at yandex.ru (royj at yandex.ru)
Date: Wed, 04 Apr 2018 17:17:10 +0300
Subject: [Freeswitch-users] Question about limit max_calls
In-Reply-To: <5ac39191.480f1c0a.98619.7465@mx.google.com>
References: <5ac39191.480f1c0a.98619.7465@mx.google.com>
Message-ID: <1524231522851430@web26o.yandex.ru>

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From enp at itx.ru  Wed Apr  4 09:53:11 2018
From: enp at itx.ru (Eugene Prokopiev)
Date: Wed, 4 Apr 2018 12:53:11 +0300
Subject: [Freeswitch-users] Global object in startup lua script
Message-ID: <CANM4Rui4LV1-pXj3kfRsP0XhTG87fh2XtGs89DMaygnjOYWrgQ@mail.gmail.com>

Hi,

Is it possible to create global object in startup lua script and use
it in other lua scripts in dialplan?

-- 
WBR,
Eugene Prokopiev


From alexandr.popov at iqoption.com  Wed Apr  4 14:59:47 2018
From: alexandr.popov at iqoption.com (Alexandr Popov)
Date: Wed, 4 Apr 2018 17:59:47 +0300
Subject: [Freeswitch-users] Global object in startup lua script
In-Reply-To: <CANM4Rui4LV1-pXj3kfRsP0XhTG87fh2XtGs89DMaygnjOYWrgQ@mail.gmail.com>
References: <CANM4Rui4LV1-pXj3kfRsP0XhTG87fh2XtGs89DMaygnjOYWrgQ@mail.gmail.com>
Message-ID: <CAKeZRhsQ0ePZZ0=-3qfUCupg8Yi6mQSM-RCXBiwL+-QKAhjB0w@mail.gmail.com>

you can store any data in channel variables

2018-04-04 12:53 GMT+03:00 Eugene Prokopiev <enp at itx.ru>:

> Hi,
>
> Is it possible to create global object in startup lua script and use
> it in other lua scripts in dialplan?
>
> --
> WBR,
> Eugene Prokopiev
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 cmrienzo at gmail.com  Wed Apr  4 15:25:54 2018
From: cmrienzo at gmail.com (Christopher Rienzo)
Date: Wed, 4 Apr 2018 11:25:54 -0400
Subject: [Freeswitch-users] Global object in startup lua script
In-Reply-To: <CANM4Rui4LV1-pXj3kfRsP0XhTG87fh2XtGs89DMaygnjOYWrgQ@mail.gmail.com>
References: <CANM4Rui4LV1-pXj3kfRsP0XhTG87fh2XtGs89DMaygnjOYWrgQ@mail.gmail.com>
Message-ID: <CAD-qRkgw7Y9yof+H0P9xQH0AZSpQA4niJegHcW7DAT+Bk08MDA@mail.gmail.com>

You can save channel variables and read them back in subsequent scripts.

On Wed, Apr 4, 2018 at 5:53 AM, Eugene Prokopiev <enp at itx.ru> wrote:

> Hi,
>
> Is it possible to create global object in startup lua script and use
> it in other lua scripts in dialplan?
>
> --
> WBR,
> Eugene Prokopiev
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 vldtoma at gmail.com  Wed Apr  4 15:50:25 2018
From: vldtoma at gmail.com (Vlad Toma)
Date: Wed, 04 Apr 2018 15:50:25 +0000
Subject: [Freeswitch-users] Question about limit max_calls
In-Reply-To: <1524231522851430@web26o.yandex.ru>
References: <5ac39191.480f1c0a.98619.7465@mx.google.com>
 <1524231522851430@web26o.yandex.ru>
Message-ID: <CAFLU3XjaCmeBF5ZeZ-PXZRDPT0ZS-aPYHNPQ0KUg5z59Tog37A@mail.gmail.com>

First of all thank you for the reply, but I am using verto communicator and
it doesn't have call waiting. I have to limit the number of calls from the
server, the max_calls option works for bridging but if I originate the call
from a conference using conference dial user/XXX at domain the max_calls
doesn't work anymore.

Kind regards,
Vlad

On Wed, Apr 4, 2018, 6:15 PM <royj at yandex.ru> wrote:

> Another approach is to disable "call waiting" feature on the phone.
>
> 03.04.2018, 19:09, "Vlad Toma" <vldtoma at gmail.com>:
>
> Hello,
>
> I want to restrict the number of incoming calls to my internal extensions
> to 1 per extension, because I don’t want to receive another call to an
> extension that is already in a call.
>
> I tried setting the max_calls variable to 1 and adding this to the dialplan
>
>
>
> <extension name="limit_exceeded">
>
>   <condition field="destination_number" expression="^limit_exceeded$">
>
>     <action application="playback" data="/sounds/overthelimit.wav"/>
>
>     <action application="hangup"/>
>
>   </condition>
>
> </extension>
>
>
>
> <extension name="limit" continue="true">
>
>   <condition>
>
>     <!-- You can set ${max_calls} per user in directory.conf and remove
> the line below -->
>
>     <action application="set" data="max_calls=1" inline="true"/>
>
>     <action application="limit" data="db $${domain} ${sip_auth_username}
> ${max_calls}"/>
>
>   </condition>
>
> </extension>
>
>
>
> Also I tried adding this to my dialplan where the condition matches my
> internal extensions
>
> <action application="limit_execute" data="hash outbound carrier1 1 bridge
> {ignore_early_media=true}user/${dialed_extension}@${domain_name}" />
>
>
>
> It works if I dial normally but if I do a conference conf dial
> {ignore_early_media=true,conference_member_flags=moderator}user/XXX@${domain_name}
> it always calls even if that user is already in a call.
>
> How can I fix this? Thank you.
>
>
>
>
>
> Kind regards,
>
>
>
> Vlad Toma
> ,
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 ksrigo at gmail.com  Wed Apr  4 19:22:25 2018
From: ksrigo at gmail.com (Srigo Kanapathipillai)
Date: Wed, 04 Apr 2018 19:22:25 +0000
Subject: [Freeswitch-users] how to access the XML_REQUEST in mod_python
In-Reply-To: <CACgE3xtChRMHMnnry7tCUyf6qqFeppXYCTCB=NMMX7M3YxJBVA@mail.gmail.com>
References: <CACgE3xunkzv9OA1jECQ8XDQuPKuXfsEnjU_utgwkzAnMLEL+KQ@mail.gmail.com>
 <CAAiAxuzo11HuxSP=chux7mTRycROgCSjDGC0h9vKXqVT1Sg9FQ@mail.gmail.com>
 <CACgE3xtChRMHMnnry7tCUyf6qqFeppXYCTCB=NMMX7M3YxJBVA@mail.gmail.com>
Message-ID: <CAAiAxuzL+_yZM=qzrsOqxyrUAxPGKGVwgSxSL1s5i=g=gqdmPg@mail.gmail.com>

Hi,

I never worked with mod python for fetching user directory dynamically.

Some quick search on google, i found these links:
http://lists.freeswitch.org/pipermail/freeswitch-users/2010-February/053696.html

https://wiki.freeswitch.org/wiki/Mod_python#Can_it_serve_configuration_.28like_Lua.29.3F

Srigo


On Sun, 1 Apr 2018, 18:40 Do Nguyen Ha, <donguyenha at gmail.com> wrote:

> hi Srigo
>
> thank you for your suggestion, i did read the mod_xml_curl documents but
> i refer the mod_lua/mod_python :)
>
> any idea for the mod_python
>
> On Sun, Apr 1, 2018 at 2:48 PM, Srigo Kanapathipillai <ksrigo at gmail.com>
> wrote:
>
>> Hi
>>
>> If you want to have dynamic directory, i suggest you to use mod_xml_curl
>> which will do an http request to your web server to get ur directory info.
>> Your webserver serving your fs config can be a python process (fcgiwrap,
>> uwsgi, gunicorn, etc with a nginx or apache for production environnement).
>>
>> Srigo
>>
>> On Sat, 31 Mar 2018, 06:23 Do Nguyen Ha, <donguyenha at gmail.com> wrote:
>>
>>> hi list
>>>
>>> i want to use mod_python to dynamic load user directory and read the
>>> wiki but no documents for access XML_REQUEST
>>>
>>> how to get the XML_REQUEST in python script
>>>
>>> thank you
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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 ksrigo at gmail.com  Wed Apr  4 19:29:07 2018
From: ksrigo at gmail.com (Srigo Kanapathipillai)
Date: Wed, 04 Apr 2018 19:29:07 +0000
Subject: [Freeswitch-users] Extract variables from 300 redirect
In-Reply-To: <B5AE4540-CEFF-4A68-98CF-1AE4F2F39BFB@tm.net.uk>
References: <B5AE4540-CEFF-4A68-98CF-1AE4F2F39BFB@tm.net.uk>
Message-ID: <CAAiAxuzpproroaHZp9CUgqs07wpazfySBwTK_UzL7pi1iN699A@mail.gmail.com>

Hi

I was wondering if you could add your variable to redirect uri as a
parameter then you can catch it in your fs dialplan and with some regex you
should be able to clean everything.

Srigo

On Tue, 20 Mar 2018, 23:50 Joseph Waite, <joelists at tm.net.uk> wrote:

> Hi Guys
>
> I am wondering if its possible to extract a custom variable from a 300
> Redirect response from a billing system?
>
> Effectively I am trying to add a custom variable to the 300 message to
> give a maximum call length to then use in the dial plan for a Scheduled
> Hangup.
>
> Any pointers of how to do this or the magic words to search
> confluence/google with?
>
> Regards
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From nzaytsevc at gmail.com  Thu Apr  5 07:03:55 2018
From: nzaytsevc at gmail.com (Nikolay Zaytsev)
Date: Thu, 5 Apr 2018 10:03:55 +0300
Subject: [Freeswitch-users] Freeswitch crash after reloadxml
Message-ID: <CADeoHuoDup79OX5=jXXYAx6rOpOLpVHVtPJ5SjLBPBc+JAhRMg@mail.gmail.com>

Hi everybody,

My freeswitch machine sometimes crashes when I do 'reloadxml' and there is
nothing neither in the freeswitch logs nor in system logs.

I am using the following configuration:

   - Centos 7 on Amazon EC2 m4.large instance
   - Freeswitch 1.6.2 with a maximum load of 250 concurrent calls
   - Dialplan is written in python and working via ESL

Does anybody have ideas?

-Best regards,
Nikolay Zaytsev
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From alex at freeswitch.com  Thu Apr  5 07:54:30 2018
From: alex at freeswitch.com (Alexey Sibyakin)
Date: Thu, 5 Apr 2018 16:54:30 +0900
Subject: [Freeswitch-users] Freeswitch crash after reloadxml
In-Reply-To: <CADeoHuoDup79OX5=jXXYAx6rOpOLpVHVtPJ5SjLBPBc+JAhRMg@mail.gmail.com>
References: <CADeoHuoDup79OX5=jXXYAx6rOpOLpVHVtPJ5SjLBPBc+JAhRMg@mail.gmail.com>
Message-ID: <CAGJLwFhQXUS3ceQ=vu5S1-hgqavRevrQbFgM1+20KUD1Bnw3Uw@mail.gmail.com>

Hi,

You need to upgrade. A lot of problems were fixed since 2015

Alex

On Thu, Apr 5, 2018 at 4:03 PM, Nikolay Zaytsev <nzaytsevc at gmail.com> wrote:

> Hi everybody,
>
> My freeswitch machine sometimes crashes when I do 'reloadxml' and there is
> nothing neither in the freeswitch logs nor in system logs.
>
> I am using the following configuration:
>
>    - Centos 7 on Amazon EC2 m4.large instance
>    - Freeswitch 1.6.2 with a maximum load of 250 concurrent calls
>    - Dialplan is written in python and working via ESL
>
> Does anybody have ideas?
>
> -Best regards,
> Nikolay Zaytsev
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 Engineer
FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045
<https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g>
Email: alex at freeswitch.com
Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/>
Need commercial support? Contact sales at freeswitch.com for details.
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From enp at itx.ru  Wed Apr  4 17:57:09 2018
From: enp at itx.ru (Eugene Prokopiev)
Date: Wed, 4 Apr 2018 20:57:09 +0300
Subject: [Freeswitch-users] Global object in startup lua script
In-Reply-To: <CAKeZRhsQ0ePZZ0=-3qfUCupg8Yi6mQSM-RCXBiwL+-QKAhjB0w@mail.gmail.com>
References: <CANM4Rui4LV1-pXj3kfRsP0XhTG87fh2XtGs89DMaygnjOYWrgQ@mail.gmail.com>
 <CAKeZRhsQ0ePZZ0=-3qfUCupg8Yi6mQSM-RCXBiwL+-QKAhjB0w@mail.gmail.com>
Message-ID: <CANM4RujGMYoPvEaa67erEHdRCh+-CuTg0LqvxosOMwQ4_Fb2HQ@mail.gmail.com>

Channel variables can be used while call executing, right? So,
impossible to share them between calls. I guess global_setvar in
startup script and global_getvar in dialplan can helps, but anyway I
can store only some value, not reference to some lua object with
methods, right?

2018-04-04 17:59 GMT+03:00 Alexandr Popov <alexandr.popov at iqoption.com>:
> you can store any data in channel variables
>
> 2018-04-04 12:53 GMT+03:00 Eugene Prokopiev <enp at itx.ru>:
>>
>> Hi,
>>
>> Is it possible to create global object in startup lua script and use
>> it in other lua scripts in dialplan?
>>
>> --
>> WBR,
>> Eugene Prokopiev
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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



-- 
WBR,
Eugene Prokopiev


From mike at jerris.com  Thu Apr  5 13:45:23 2018
From: mike at jerris.com (Michael Jerris)
Date: Thu, 05 Apr 2018 13:45:23 +0000
Subject: [Freeswitch-users] Global object in startup lua script
In-Reply-To: <CANM4RujGMYoPvEaa67erEHdRCh+-CuTg0LqvxosOMwQ4_Fb2HQ@mail.gmail.com>
References: <CANM4Rui4LV1-pXj3kfRsP0XhTG87fh2XtGs89DMaygnjOYWrgQ@mail.gmail.com>
 <CAKeZRhsQ0ePZZ0=-3qfUCupg8Yi6mQSM-RCXBiwL+-QKAhjB0w@mail.gmail.com>
 <CANM4RujGMYoPvEaa67erEHdRCh+-CuTg0LqvxosOMwQ4_Fb2HQ@mail.gmail.com>
Message-ID: <CAEwTsAgKVokG63vE+CZVLiLapRbhCN9ajFKDyZqHARMTmwKsrA@mail.gmail.com>

On Thu, Apr 5, 2018 at 9:39 AM Eugene Prokopiev <enp at itx.ru> wrote:

> Channel variables can be used while call executing, right? So,
> impossible to share them between calls. I guess global_setvar in
> startup script and global_getvar in dialplan can helps, but anyway I
> can store only some value, not reference to some lua object with
> methods, right?


Correct

>
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From cmrienzo at gmail.com  Thu Apr  5 14:11:25 2018
From: cmrienzo at gmail.com (cmrienzo at gmail.com)
Date: Thu, 5 Apr 2018 10:11:25 -0400
Subject: [Freeswitch-users] Global object in startup lua script
In-Reply-To: <CAEwTsAgKVokG63vE+CZVLiLapRbhCN9ajFKDyZqHARMTmwKsrA@mail.gmail.com>
References: <CANM4Rui4LV1-pXj3kfRsP0XhTG87fh2XtGs89DMaygnjOYWrgQ@mail.gmail.com>
 <CAKeZRhsQ0ePZZ0=-3qfUCupg8Yi6mQSM-RCXBiwL+-QKAhjB0w@mail.gmail.com>
 <CANM4RujGMYoPvEaa67erEHdRCh+-CuTg0LqvxosOMwQ4_Fb2HQ@mail.gmail.com>
 <CAEwTsAgKVokG63vE+CZVLiLapRbhCN9ajFKDyZqHARMTmwKsrA@mail.gmail.com>
Message-ID: <589C1A03-F1F2-4C69-B5C2-86097673E623@gmail.com>

You could serialize the data as json prior to writing to the channel variable.   Dkjson works well in lua for that. 

Chris


> On Apr 5, 2018, at 09:45, Michael Jerris <mike at jerris.com> wrote:
> 
> 
>> On Thu, Apr 5, 2018 at 9:39 AM Eugene Prokopiev <enp at itx.ru> wrote:
>> Channel variables can be used while call executing, right? So,
>> impossible to share them between calls. I guess global_setvar in
>> startup script and global_getvar in dialplan can helps, but anyway I
>> can store only some value, not reference to some lua object with
>> methods, right?
> 
> Correct 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 francesco at delagarda.com  Thu Apr  5 14:39:48 2018
From: francesco at delagarda.com (Francesco Facco de Lagarda)
Date: Thu, 5 Apr 2018 16:39:48 +0200
Subject: [Freeswitch-users] cdr via javascript
Message-ID: <048d01d3cceb$f472ee30$dd58ca90$@delagarda.com>

Is there any way I can interrogate cdr data of a call that has finished via
javascript?
I see no samples of cdr data with js.

 

Thanks

F

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From mail at paulzillmann.de  Thu Apr  5 16:00:25 2018
From: mail at paulzillmann.de (Paul Zillmann)
Date: Thu, 5 Apr 2018 18:00:25 +0200
Subject: [Freeswitch-users] cdr via javascript
In-Reply-To: <048d01d3cceb$f472ee30$dd58ca90$@delagarda.com>
References: <048d01d3cceb$f472ee30$dd58ca90$@delagarda.com>
Message-ID: <3b9e319a-0b7d-9966-17fa-13ab933dda60@paulzillmann.de>

nodeJS or Browser?


Am 05.04.2018 um 16:39 schrieb Francesco Facco de Lagarda:
>
> Is there any way I can interrogate cdr data of a call that has 
> finished via javascript?
> I see no samples of cdr data with js.
>
> Thanks
>
> F
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From francesco at delagarda.com  Thu Apr  5 16:45:12 2018
From: francesco at delagarda.com (Francesco Facco de Lagarda)
Date: Thu, 5 Apr 2018 18:45:12 +0200
Subject: [Freeswitch-users] cdr via javascript
In-Reply-To: <3b9e319a-0b7d-9966-17fa-13ab933dda60@paulzillmann.de>
References: <048d01d3cceb$f472ee30$dd58ca90$@delagarda.com>
 <3b9e319a-0b7d-9966-17fa-13ab933dda60@paulzillmann.de>
Message-ID: <056701d3ccfd$782db550$68891ff0$@delagarda.com>

Thanks Paul for you interest! Much appreciated.

Almost solved!

Js script called dialplan.

I’m using this code:

 

if (session.ready()) {

 

session.answer();

 

var sessOut = new Session("sofia/gateway/realtoneFXO/" +  dialedNum +
"@192.168.0.216:5060");

                

if (sessOut.ready()) {

                                bridge(session, sessOut);

}

 

sessOut.hangup();

session.hangup();

 

var xml = new XML ("<xml>" + sessOut.generateXmlCdr() + "</xml>");

var xmlCDR                        = xml.getChild('cdr');

var xmCallFlow                 = xmlCDR.getChild('callflow');

var xmTimes                      = xmCallFlow.getChild('times');

var created_time                                            =
parseInt(xmTimes.getChild('created_time').data);

var profile_created_time                             =
parseInt(xmTimes.getChild('profile_created_time').data);

var progress_time                                          =
parseInt(xmTimes.getChild('progress_time').data);

var progress_media_time                           =
parseInt(xmTimes.getChild('progress_media_time').data);      

var answered_time                                       =
parseInt(xmTimes.getChild('answered_time').data);   

var bridged_time                                            =
parseInt(xmTimes.getChild('bridged_time').data);       

var last_hold_time                                         =
parseInt(xmTimes.getChild('last_hold_time').data);    

var hold_accum_time                                   =
parseInt(xmTimes.getChild('hold_accum_time').data);              

var hangup_time                                            =
parseInt(xmTimes.getChild('hangup_time').data);        

var resurrect_time                                         =
parseInt(xmTimes.getChild('resurrect_time').data);     

var transfer_time                                           =
parseInt(xmTimes.getChild('transfer_time').data);       

                

var callLength = (hangup_time - answered_time)/1000000;

}

 

which works fine BUT, as "sofia/gateway/realtoneFXO/" is on analog line
(FXO), it does not actually communicate the on-answer properly, so the call
length is the total length including dialing, waiting for ring, ringing 

and not just the speaking time


 

as you see from these values, the answered_time and bridge_time are
identical!

 

<created_time>1522946076054074</created_time>

<profile_created_time>1522946076054074</profile_created_time>

<progress_time>0</progress_time>

<progress_media_time>0</progress_media_time>

<answered_time>1522946079154074</answered_time>

<bridged_time>1522946079154074</bridged_time>

<last_hold_time>0</last_hold_time>

<hold_accum_time>0</hold_accum_time>

<hangup_time>1522946095634072</hangup_time>

<resurrect_time>0</resurrect_time>

<transfer_time>0</transfer_time>

 

 

 

From: FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> On
Behalf Of Paul Zillmann
Sent: giovedì 5 aprile 2018 18:00
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] cdr via javascript

 

nodeJS or Browser?

 

Am 05.04.2018 um 16:39 schrieb Francesco Facco de Lagarda:

Is there any way I can interrogate cdr data of a call that has finished via
javascript?
I see no samples of cdr data with js.

 

Thanks

F






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

 

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From mail at paulzillmann.de  Thu Apr  5 17:48:30 2018
From: mail at paulzillmann.de (Paul Zillmann)
Date: Thu, 5 Apr 2018 19:48:30 +0200
Subject: [Freeswitch-users] cdr via javascript
In-Reply-To: <056701d3ccfd$782db550$68891ff0$@delagarda.com>
References: <048d01d3cceb$f472ee30$dd58ca90$@delagarda.com>
 <3b9e319a-0b7d-9966-17fa-13ab933dda60@paulzillmann.de>
 <056701d3ccfd$782db550$68891ff0$@delagarda.com>
Message-ID: <ab31210a-e125-962f-889c-9fcc6dc96bcb@paulzillmann.de>

Hey Francesco,

my guess is that you don't have to answer the a-Leg. Usually the bridge 
application answers the a-Leg when the b-Leg answers.
So your FXO card doesn't push an answer-event?

Paul

Am 05.04.2018 um 18:45 schrieb Francesco Facco de Lagarda:
>
> Thanks Paul for you interest! Much appreciated.
>
> Almost solved!
>
> Js script called dialplan.
>
> I’m using this code:
>
> if (session.ready()) {
>
> session.answer();
>
> var sessOut = new Session("sofia/gateway/realtoneFXO/" +  dialedNum + 
> "@192.168.0.216:5060");
>
> if (sessOut.ready()) {
>
> bridge(session, sessOut);
>
> }
>
> sessOut.hangup();
>
> session.hangup();
>
> var xml = new XML ("<xml>" + sessOut.generateXmlCdr() + "</xml>");
>
> var xmlCDR                        = xml.getChild('cdr');
>
> var xmCallFlow                 = xmlCDR.getChild('callflow');
>
> var xmTimes                 = xmCallFlow.getChild('times');
>
> var created_time                                            = 
> parseInt(xmTimes.getChild('created_time').data);
>
> var profile_created_time                             = 
> parseInt(xmTimes.getChild('profile_created_time').data);
>
> var progress_time                                          = 
> parseInt(xmTimes.getChild('progress_time').data);
>
> var progress_media_time                           = 
> parseInt(xmTimes.getChild('progress_media_time').data);
>
> var answered_time                                       = 
> parseInt(xmTimes.getChild('answered_time').data);
>
> var bridged_time                                            = 
> parseInt(xmTimes.getChild('bridged_time').data);
>
> var last_hold_time                                         = 
> parseInt(xmTimes.getChild('last_hold_time').data);
>
> var hold_accum_time                                   = 
> parseInt(xmTimes.getChild('hold_accum_time').data);
>
> var hangup_time                                            = 
> parseInt(xmTimes.getChild('hangup_time').data);
>
> var resurrect_time                                         = 
> parseInt(xmTimes.getChild('resurrect_time').data);
>
> var transfer_time                                           = 
> parseInt(xmTimes.getChild('transfer_time').data);
>
> var callLength = (hangup_time - answered_time)/1000000;
>
> }
>
> which works fine BUT, as "sofia/gateway/realtoneFXO/" is on analog 
> line (FXO), it does not actually communicate the on-answer properly, 
> so the call length is the total length including dialing, waiting for 
> ring, ringing … and not just the speaking time…
>
> as you see from these values, the answered_time and bridge_time are 
> identical!
>
> <created_time>1522946076054074</created_time>
>
> <profile_created_time>1522946076054074</profile_created_time>
>
> <progress_time>0</progress_time>
>
> <progress_media_time>0</progress_media_time>
>
> <answered_time>1522946079154074</answered_time>
>
> <bridged_time>1522946079154074</bridged_time>
>
> <last_hold_time>0</last_hold_time>
>
> <hold_accum_time>0</hold_accum_time>
>
> <hangup_time>1522946095634072</hangup_time>
>
> <resurrect_time>0</resurrect_time>
>
> <transfer_time>0</transfer_time>
>
> *From:*FreeSWITCH-users 
> <freeswitch-users-bounces at lists.freeswitch.org> *On Behalf Of *Paul 
> Zillmann
> *Sent:* giovedì 5 aprile 2018 18:00
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] cdr via javascript
>
> nodeJS or Browser?
>
> Am 05.04.2018 um 16:39 schrieb Francesco Facco de Lagarda:
>
>     Is there any way I can interrogate cdr data of a call that has
>     finished via javascript?
>     I see no samples of cdr data with js.
>
>     Thanks
>
>     F
>
>
>
>
>     _________________________________________________________________________
>
>     Professional FreeSWITCH Consulting Services:
>
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>
>     http://www.freeswitchsolutions.com
>
>     Official FreeSWITCH Sites
>
>     http://www.freeswitch.org
>
>     http://confluence.freeswitch.org
>
>     http://www.cluecon.com
>
>     FreeSWITCH-users mailing list
>
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>
>     http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From bob.mccarthy at westtel.com  Sat Apr  7 16:04:35 2018
From: bob.mccarthy at westtel.com (Bob McCarthy)
Date: Sat, 7 Apr 2018 16:04:35 +0000
Subject: [Freeswitch-users] Adding a Parked Call into a conference
In-Reply-To: <EE02A9B2-BA9A-4A82-A221-63755B16C0E3@jerris.com>
References: <DM5PR10MB12571F4511693A3A1DC8FA7E87A30@DM5PR10MB1257.namprd10.prod.outlook.com>
 <EE02A9B2-BA9A-4A82-A221-63755B16C0E3@jerris.com>
Message-ID: <DM5PR10MB125720907B30B204A9DBEED087B90@DM5PR10MB1257.namprd10.prod.outlook.com>

Ended up using the uuid_transfer in a lua script to get the call out of the lot.

Thanks
Bob

From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris
Sent: Thursday, March 29, 2018 9:24 AM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Adding a Parked Call into a conference

are you transferring out of the parking lot?  sounds like maybe you are executing extension while its parked.

On Mar 27, 2018, at 8:34 PM, Bob McCarthy <bob.mccarthy at westtel.com<mailto:bob.mccarthy at westtel.com>> wrote:

Anyone have a best practice for taking a channel out of a parking lot and adding it to a conference?
I have successfully transferred into the parking lot but the results are less than desired.  The BLF remains active until the conference ends
and pressing it give you hold music.  Looking for an API / lua solution.

Thanks in advance,
Bob McCarthy

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From bob.mccarthy at westtel.com  Sat Apr  7 16:10:02 2018
From: bob.mccarthy at westtel.com (Bob McCarthy)
Date: Sat, 7 Apr 2018 16:10:02 +0000
Subject: [Freeswitch-users] sip_presence
Message-ID: <DM5PR10MB1257706E7E3F69D03C25A9AA87B90@DM5PR10MB1257.namprd10.prod.outlook.com>

Is there any way to access the sip_presence db from the api ?  Have documented a memory leak in presence FS-10808 and I would like to troubleshoot its root cause.

Bob McCarthy
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From andrew.keil at visytel.com  Mon Apr  9 07:48:48 2018
From: andrew.keil at visytel.com (Andrew Keil)
Date: Mon, 9 Apr 2018 07:48:48 +0000
Subject: [Freeswitch-users] Controlling an inbound call remotely over ESL -
	dialplan question
Message-ID: <MEXPR01MB12235911AF59C55EB3F14701F3BF0@MEXPR01MB1223.ausprd01.prod.outlook.com>

To FreeSWITCH users,


I would like to control an inbound call to FreeSWITCH via "inbound ESL to FreeSWITCH" and I need to know how I can write a simple diaplan which basically does nothing (just sits there and does not even answer the caller and does not hangup on the caller).  Then I should be able to control the entire call remotely over ESL (even call answer).  Maybe I do not even need a dialplan?


Please note that I am wishing to use "inbound ESL to FreeSWITCH" to control the call, not outbound ESL, and I fully understand how to send API commands over ESL.


Andrew





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From vma at vallimamod.org  Mon Apr  9 09:04:44 2018
From: vma at vallimamod.org (Vallimamod Abdullah)
Date: Mon, 9 Apr 2018 11:04:44 +0200
Subject: [Freeswitch-users] Controlling an inbound call remotely over
 ESL - dialplan question
In-Reply-To: <MEXPR01MB12235911AF59C55EB3F14701F3BF0@MEXPR01MB1223.ausprd01.prod.outlook.com>
References: <MEXPR01MB12235911AF59C55EB3F14701F3BF0@MEXPR01MB1223.ausprd01.prod.outlook.com>
Message-ID: <03B7D817-9041-4668-A38C-8010E3F061B9@vallimamod.org>

Hi,

You can use the park application to place an incoming call in a waiting state. Then  you can listen and react to channel_park event through esl.
You can also look at mod_fifo if you want some moh and announcement.

Best Regards,
-- 
Vallimamod Abdullah
SIP Solutions
github.com/vma/esl
linkedin.com/in/vallimamod
.

> On 9 Apr 2018, at 09:48, Andrew Keil <andrew.keil at visytel.com> wrote:
> 
> To FreeSWITCH users,
> 
> I would like to control an inbound call to FreeSWITCH via "inbound ESL to FreeSWITCH" and I need to know how I can write a simple diaplan which basically does nothing (just sits there and does not even answer the caller and does not hangup on the caller).  Then I should be able to control the entire call remotely over ESL (even call answer).  Maybe I do not even need a dialplan?
> 
> Please note that I am wishing to use "inbound ESL to FreeSWITCH" to control the call, not outbound ESL, and I fully understand how to send API commands over ESL.
> 
> Andrew
> 
>  
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> http://www.cluecon.com <http://www.cluecon.com/>
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> http://www.freeswitch.org <http://www.freeswitch.org/>
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From donguyenha at gmail.com  Mon Apr  9 10:45:21 2018
From: donguyenha at gmail.com (Do Nguyen Ha)
Date: Mon, 9 Apr 2018 17:45:21 +0700
Subject: [Freeswitch-users] how to access the XML_REQUEST in mod_python
In-Reply-To: <CAAiAxuzL+_yZM=qzrsOqxyrUAxPGKGVwgSxSL1s5i=g=gqdmPg@mail.gmail.com>
References: <CACgE3xunkzv9OA1jECQ8XDQuPKuXfsEnjU_utgwkzAnMLEL+KQ@mail.gmail.com>
 <CAAiAxuzo11HuxSP=chux7mTRycROgCSjDGC0h9vKXqVT1Sg9FQ@mail.gmail.com>
 <CACgE3xtChRMHMnnry7tCUyf6qqFeppXYCTCB=NMMX7M3YxJBVA@mail.gmail.com>
 <CAAiAxuzL+_yZM=qzrsOqxyrUAxPGKGVwgSxSL1s5i=g=gqdmPg@mail.gmail.com>
Message-ID: <CACgE3xtC4Wza1KnALSLnhKNp_FPqzOsoxVpQM3_HvgOhuwDDPw@mail.gmail.com>

thank you for your help

On Thu, Apr 5, 2018 at 2:22 AM, Srigo Kanapathipillai <ksrigo at gmail.com>
wrote:

> Hi,
>
> I never worked with mod python for fetching user directory dynamically.
>
> Some quick search on google, i found these links:
> http://lists.freeswitch.org/pipermail/freeswitch-users/
> 2010-February/053696.html
>
> https://wiki.freeswitch.org/wiki/Mod_python#Can_it_serve_
> configuration_.28like_Lua.29.3F
>
> Srigo
>
>
> On Sun, 1 Apr 2018, 18:40 Do Nguyen Ha, <donguyenha at gmail.com> wrote:
>
>> hi Srigo
>>
>> thank you for your suggestion, i did read the mod_xml_curl documents but
>> i refer the mod_lua/mod_python :)
>>
>> any idea for the mod_python
>>
>> On Sun, Apr 1, 2018 at 2:48 PM, Srigo Kanapathipillai <ksrigo at gmail.com>
>> wrote:
>>
>>> Hi
>>>
>>> If you want to have dynamic directory, i suggest you to use mod_xml_curl
>>> which will do an http request to your web server to get ur directory info.
>>> Your webserver serving your fs config can be a python process (fcgiwrap,
>>> uwsgi, gunicorn, etc with a nginx or apache for production environnement).
>>>
>>> Srigo
>>>
>>> On Sat, 31 Mar 2018, 06:23 Do Nguyen Ha, <donguyenha at gmail.com> wrote:
>>>
>>>> hi list
>>>>
>>>> i want to use mod_python to dynamic load user directory and read the
>>>> wiki but no documents for access XML_REQUEST
>>>>
>>>> how to get the XML_REQUEST in python script
>>>>
>>>> thank you
>>>> ____________________________________________________________
>>>> _____________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/
>>>> options/freeswitch-users
>>>> http://www.freeswitch.org
>>>
>>>
>>> ____________________________________________________________
>>> _____________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From andrew.keil at visytel.com  Mon Apr  9 11:11:34 2018
From: andrew.keil at visytel.com (Andrew Keil)
Date: Mon, 9 Apr 2018 11:11:34 +0000
Subject: [Freeswitch-users] Controlling an inbound call remotely over
 ESL - dialplan question
In-Reply-To: <03B7D817-9041-4668-A38C-8010E3F061B9@vallimamod.org>
References: <MEXPR01MB12235911AF59C55EB3F14701F3BF0@MEXPR01MB1223.ausprd01.prod.outlook.com>
 <03B7D817-9041-4668-A38C-8010E3F061B9@vallimamod.org>
Message-ID: <MEXPR01MB1223C233227448C01EC264B7F3BF0@MEXPR01MB1223.ausprd01.prod.outlook.com>

I will give that a go and see how it goes.

From: FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> On Behalf Of Vallimamod Abdullah
Sent: Monday, 9 April 2018 7:05 PM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Controlling an inbound call remotely over ESL - dialplan question

Hi,

You can use the park application to place an incoming call in a waiting state. Then  you can listen and react to channel_park event through esl.
You can also look at mod_fifo if you want some moh and announcement.

Best Regards,
--
Vallimamod Abdullah
SIP Solutions
github.com/vma/esl<http://github.com/vma/esl>
linkedin.com/in/vallimamod<http://linkedin.com/in/vallimamod>
.


On 9 Apr 2018, at 09:48, Andrew Keil <andrew.keil at visytel.com<mailto:andrew.keil at visytel.com>> wrote:

To FreeSWITCH users,

I would like to control an inbound call to FreeSWITCH via "inbound ESL to FreeSWITCH" and I need to know how I can write a simple diaplan which basically does nothing (just sits there and does not even answer the caller and does not hangup on the caller).  Then I should be able to control the entire call remotely over ESL (even call answer).  Maybe I do not even need a dialplan?

Please note that I am wishing to use "inbound ESL to FreeSWITCH" to control the call, not outbound ESL, and I fully understand how to send API commands over ESL.

Andrew




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

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

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

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From nerad.peter at gmail.com  Mon Apr  9 12:46:11 2018
From: nerad.peter at gmail.com (=?iso-8859-2?Q?Peter_Ner=E1d?=)
Date: Mon, 9 Apr 2018 14:46:11 +0200
Subject: [Freeswitch-users] Older OpenVox G400E card
Message-ID: <!&!AAAAAAAAAAAYAAAAAAAAAI/RfDx9+axBlz43ajVi2RXCgAAAEAAAAJc+9ZtTNwlLildRzXcfuhQBAAAAAA==@gmail.com>

Hi it is possible to use older openvox G400E card with the latest freeswitch
? 

Where to start to get this card to live ? :-D

Thanks for any help

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From ch.chhatra at gmail.com  Sun Apr  8 17:18:42 2018
From: ch.chhatra at gmail.com (Chhorm Chhatra)
Date: Mon, 9 Apr 2018 00:18:42 +0700
Subject: [Freeswitch-users] Setting Headers in mod_xml_curl and mod_curl
Message-ID: <CAJssr-isOWwsNn0jXWyJYta5ATsH+Cdt=h_cQEu5Y-LW0F_RNA@mail.gmail.com>

Thank you for making freeswitch awesome!
Currently, i face an issue regarding setting headers in mod_xml_curl as per
the security requirements to the remote server.
I would like to know how to include custom headers in mod_xml_curl
configuration.
If you do not mind, please kindly let me know if there is any possibility
to do so.
Thank you so much.

Best Regard,
Chhatra Chhorm
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From freeswitch-users at enlai.net  Mon Apr  9 04:33:23 2018
From: freeswitch-users at enlai.net (E Chu)
Date: Mon, 9 Apr 2018 00:33:23 -0400
Subject: [Freeswitch-users] How to get 200 Contact to use FQDN instead of IP
Message-ID: <CAJ_-ZtvGSyo07wMDkXUhK350Kh67JM-EG7Y2pErfZ2SCD6JMnw@mail.gmail.com>

Hi all,

Does anyone know how I can get FS to respond with a FQDN instead of an IP
address in the contact field of the 200?

The domain is set to FQDN in vars.xml:

[hostnames and IPs changed for public list privacy]

<X-PRE-PROCESS cmd=“set” data="domain=fs-dev1.sipserver.com"/>

i.e. but this is what FS is sending in the 200:

Contact: <sip:123123123 at 1.2.3.4:5060;transport=tcp>

I’d like it to be:

Contact: <sip:123123123 at fs-dev1.sipserver.com:5060;transport=tcp>

This is the full 200 response from FS:

SIP/2.0 200 OK
Via: SIP/2.0/TCP 192.168.0.1:52631;rport=42810;branch=
z9hG4bKPjwPkMUAMcou1XPnDrrWJH9GDwdlaPKXCF;alias;received=2.3.4.5
From: "1020" <sip:1020 at fs-dev1.sipserver.com>;tag=h9ydSS-
jO2dQ7xT8yxpVcLu-appl6Xtl
To: <sip:123123123 at fs-dev1.sipserver.com>;tag=vXtN8yZgtHy0B
Call-ID: Y3tWlOhNz3f9bt-uHO-LjFjh2RQwQh9l
CSeq: 13167 INVITE
Contact: <sip:123123123 at 1.2.3.4:5060;transport=tcp> //HOW TO GET THIS TO
USE FQDN
User-Agent: FreeSWITCH-mod_sofia/1.6.16~64bit
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
REFER, NOTIFY, PUBLISH, SUBSCRIBE
Require: timer
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
Session-Expires: 120;refresher=uac
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 248
Remote-Party-ID: “123123123" <sip:123123123 at fs-dev1.sipserver.com
>;party=calling;privacy=off;screen=no

v=0
o=FreeSWITCH 1522636701 1522636702 IN IP4 1.2.3.4
s=FreeSWITCH
c=IN IP4 1.2.3.4
t=0 0
m=audio 18966 RTP/AVP 0 96
a=rtpmap:0 PCMU/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=ptime:20
a=rtcp:18967 IN IP4 1.2.3.4



Any ideas please?

Thanks!
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From mike at jerris.com  Mon Apr  9 14:54:06 2018
From: mike at jerris.com (Michael Jerris)
Date: Mon, 9 Apr 2018 10:54:06 -0400
Subject: [Freeswitch-users] sip_presence
In-Reply-To: <DM5PR10MB1257706E7E3F69D03C25A9AA87B90@DM5PR10MB1257.namprd10.prod.outlook.com>
References: <DM5PR10MB1257706E7E3F69D03C25A9AA87B90@DM5PR10MB1257.namprd10.prod.outlook.com>
Message-ID: <84296795-1975-424F-81DA-81DE342E69E1@jerris.com>

put it in odbc source and its easy to access.

> On Apr 7, 2018, at 12:10 PM, Bob McCarthy <bob.mccarthy at westtel.com> wrote:
> 
> Is there any way to access the sip_presence db from the api ?  Have documented a memory leak in presence FS-10808 and I would like to troubleshoot its root cause.
>  
> Bob McCarthy

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From mike at jerris.com  Mon Apr  9 14:55:34 2018
From: mike at jerris.com (Michael Jerris)
Date: Mon, 9 Apr 2018 10:55:34 -0400
Subject: [Freeswitch-users] Older OpenVox G400E card
In-Reply-To: <!&!AAAAAAAAAAAYAAAAAAAAAI/RfDx9+axBlz43ajVi2RXCgAAAEAAAAJc+9ZtTNwlLildRzXcfuhQBAAAAAA==@gmail.com>
References: <!&!AAAAAAAAAAAYAAAAAAAAAI/RfDx9+axBlz43ajVi2RXCgAAAEAAAAJc+9ZtTNwlLildRzXcfuhQBAAAAAA==@gmail.com>
Message-ID: <813B3012-8D5E-4ED3-8E6E-F2626B572712@jerris.com>

possibly with freetdm.  We don’t really maintain it anymore as demand for hardware interfaces is so low.


> On Apr 9, 2018, at 8:46 AM, Peter Nerád <nerad.peter at gmail.com> wrote:
> 
> Hi it is possible to use older openvox G400E card with the latest freeswitch ? 
> Where to start to get this card to live ? :-D
> Thanks for any help


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From royj at yandex.ru  Mon Apr  9 16:04:24 2018
From: royj at yandex.ru (royj at yandex.ru)
Date: Mon, 09 Apr 2018 19:04:24 +0300
Subject: [Freeswitch-users] Question about limit max_calls
In-Reply-To: <CAFLU3XjaCmeBF5ZeZ-PXZRDPT0ZS-aPYHNPQ0KUg5z59Tog37A@mail.gmail.com>
References: <5ac39191.480f1c0a.98619.7465@mx.google.com>
 <1524231522851430@web26o.yandex.ru>
 <CAFLU3XjaCmeBF5ZeZ-PXZRDPT0ZS-aPYHNPQ0KUg5z59Tog37A@mail.gmail.com>
Message-ID: <654711523289864@web20g.yandex.ru>

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From alexandr.popov at iqoption.com  Mon Apr  9 16:10:43 2018
From: alexandr.popov at iqoption.com (Alexandr Popov)
Date: Mon, 9 Apr 2018 19:10:43 +0300
Subject: [Freeswitch-users] Setting Headers in mod_xml_curl and mod_curl
In-Reply-To: <CAJssr-isOWwsNn0jXWyJYta5ATsH+Cdt=h_cQEu5Y-LW0F_RNA@mail.gmail.com>
References: <CAJssr-isOWwsNn0jXWyJYta5ATsH+Cdt=h_cQEu5Y-LW0F_RNA@mail.gmail.com>
Message-ID: <CAKeZRhvtA6zDSLoNQwEMDm1F3nh1n+21qQSxCRC8K_iiYbH4Og@mail.gmail.com>

use proxy

2018-04-08 20:18 GMT+03:00 Chhorm Chhatra <ch.chhatra at gmail.com>:

> Thank you for making freeswitch awesome!
> Currently, i face an issue regarding setting headers in mod_xml_curl as
> per the security requirements to the remote server.
> I would like to know how to include custom headers in mod_xml_curl
> configuration.
> If you do not mind, please kindly let me know if there is any possibility
> to do so.
> Thank you so much.
>
> Best Regard,
> Chhatra Chhorm
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 mike at jerris.com  Mon Apr  9 16:17:21 2018
From: mike at jerris.com (Michael Jerris)
Date: Mon, 9 Apr 2018 12:17:21 -0400
Subject: [Freeswitch-users] How to get 200 Contact to use FQDN instead
 of IP
In-Reply-To: <CAJ_-ZtvGSyo07wMDkXUhK350Kh67JM-EG7Y2pErfZ2SCD6JMnw@mail.gmail.com>
References: <CAJ_-ZtvGSyo07wMDkXUhK350Kh67JM-EG7Y2pErfZ2SCD6JMnw@mail.gmail.com>
Message-ID: <8EB3F3B8-B1D9-49BF-85D6-30A7236D15BD@jerris.com>

why would you want to do this?

> On Apr 9, 2018, at 12:33 AM, E Chu <freeswitch-users at enlai.net> wrote:
> 
> Hi all,
> 
> Does anyone know how I can get FS to respond with a FQDN instead of an IP address in the contact field of the 200? 
> 
> The domain is set to FQDN in vars.xml:
> 
> [hostnames and IPs changed for public list privacy]
> 
> <X-PRE-PROCESS cmd=“set” data="domain=fs-dev1.sipserver.com <http://fs-dev1.sipserver.com/>"/>
> 
> i.e. but this is what FS is sending in the 200:
> 
> Contact: <sip:123123123 at 1.2.3.4:5060 <http://sip:123123123 at 1.2.3.4:5060/>;transport=tcp>
> 
> I’d like it to be:
> 
> Contact: <sip:123123123 at fs-dev1.sipserver.com:5060;transport=tcp>
> 
> This is the full 200 response from FS:
> 
> SIP/2.0 200 OK
> Via: SIP/2.0/TCP 192.168.0.1:52631;rport=42810;branch=z9hG4bKPjwPkMUAMcou1XPnDrrWJH9GDwdlaPKXCF;alias;received=2.3.4.5
> From: "1020" <sip:1020 at fs-dev1.sipserver.com <mailto:sip%3A1020 at fs-dev1.sipserver.com>>;tag=h9ydSS-jO2dQ7xT8yxpVcLu-appl6Xtl
> To: <sip:123123123 at fs-dev1.sipserver.com <mailto:sip%3A123123123 at fs-dev1.sipserver.com>>;tag=vXtN8yZgtHy0B
> Call-ID: Y3tWlOhNz3f9bt-uHO-LjFjh2RQwQh9l
> CSeq: 13167 INVITE
> Contact: <sip:123123123 at 1.2.3.4:5060 <http://sip:123123123 at 1.2.3.4:5060/>;transport=tcp> //HOW TO GET THIS TO USE FQDN
> User-Agent: FreeSWITCH-mod_sofia/1.6.16~64bit
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
> Require: timer
> 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
> Session-Expires: 120;refresher=uac
> Content-Type: application/sdp
> Content-Disposition: session
> Content-Length: 248
> Remote-Party-ID: “123123123" <sip:123123123 at fs-dev1.sipserver.com <mailto:sip%3A123123123 at fs-dev1.sipserver.com>>;party=calling;privacy=off;screen=no
> 
> v=0
> o=FreeSWITCH 1522636701 1522636702 IN IP4 1.2.3.4
> s=FreeSWITCH
> c=IN IP4 1.2.3.4
> t=0 0
> m=audio 18966 RTP/AVP 0 96
> a=rtpmap:0 PCMU/8000
> a=rtpmap:96 telephone-event/8000
> a=fmtp:96 0-16
> a=ptime:20
> a=rtcp:18967 IN IP4 1.2.3.4
> 

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From mike at jerris.com  Mon Apr  9 16:19:52 2018
From: mike at jerris.com (Michael Jerris)
Date: Mon, 9 Apr 2018 12:19:52 -0400
Subject: [Freeswitch-users] Setting Headers in mod_xml_curl and mod_curl
In-Reply-To: <CAJssr-isOWwsNn0jXWyJYta5ATsH+Cdt=h_cQEu5Y-LW0F_RNA@mail.gmail.com>
References: <CAJssr-isOWwsNn0jXWyJYta5ATsH+Cdt=h_cQEu5Y-LW0F_RNA@mail.gmail.com>
Message-ID: <83C967F2-430B-48B1-A4B3-D10ECF75ED66@jerris.com>

there is no way in the code to do this currently, it would require modifications to the module to add support for this feature.

> On Apr 8, 2018, at 1:18 PM, Chhorm Chhatra <ch.chhatra at gmail.com> wrote:
> 
> Thank you for making freeswitch awesome!
> Currently, i face an issue regarding setting headers in mod_xml_curl as per the security requirements to the remote server.
> I would like to know how to include custom headers in mod_xml_curl configuration.
> If you do not mind, please kindly let me know if there is any possibility to do so.
> Thank you so much.



From paul.muaddib83 at gmail.com  Mon Apr  9 18:24:29 2018
From: paul.muaddib83 at gmail.com (Paul Muaddib)
Date: Mon, 9 Apr 2018 20:24:29 +0200
Subject: [Freeswitch-users] caller redirect
Message-ID: <CAEjNe_G9HdDSWXa2qNrMe+b8G0rdnjqeB5PS7KZs5JOMvhNgiQ@mail.gmail.com>

Hi,

one of our products has a service number which we purchase externally from
a service provider. Now we don't want to change the numbers on our
documents every time we change the service provider. Our company is located
in Germany, the current service provider in the USA. Can I simply forward
the incoming calls to the phone number in the USA with the "deflect"
application without any costs for us?

What about consumer protection? For example, the customer calls from
Germany or Europe, but then has to make an international call with
additional costs.


Regards,

Paul
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From bob.mccarthy at westtel.com  Mon Apr  9 19:58:09 2018
From: bob.mccarthy at westtel.com (Bob McCarthy)
Date: Mon, 9 Apr 2018 19:58:09 +0000
Subject: [Freeswitch-users] sip_presence
In-Reply-To: <84296795-1975-424F-81DA-81DE342E69E1@jerris.com>
References: <DM5PR10MB1257706E7E3F69D03C25A9AA87B90@DM5PR10MB1257.namprd10.prod.outlook.com>
 <84296795-1975-424F-81DA-81DE342E69E1@jerris.com>
Message-ID: <DM5PR10MB1257955865878084F2AEF7C387BF0@DM5PR10MB1257.namprd10.prod.outlook.com>

How about using the pgsql built into the core ?

From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris
Sent: Monday, April 09, 2018 8:54 AM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] sip_presence

put it in odbc source and its easy to access.

On Apr 7, 2018, at 12:10 PM, Bob McCarthy <bob.mccarthy at westtel.com<mailto:bob.mccarthy at westtel.com>> wrote:

Is there any way to access the sip_presence db from the api ?  Have documented a memory leak in presence FS-10808 and I would like to troubleshoot its root cause.

Bob McCarthy

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From mike at jerris.com  Mon Apr  9 20:37:39 2018
From: mike at jerris.com (Michael Jerris)
Date: Mon, 9 Apr 2018 16:37:39 -0400
Subject: [Freeswitch-users] sip_presence
In-Reply-To: <DM5PR10MB1257955865878084F2AEF7C387BF0@DM5PR10MB1257.namprd10.prod.outlook.com>
References: <DM5PR10MB1257706E7E3F69D03C25A9AA87B90@DM5PR10MB1257.namprd10.prod.outlook.com>
 <84296795-1975-424F-81DA-81DE342E69E1@jerris.com>
 <DM5PR10MB1257955865878084F2AEF7C387BF0@DM5PR10MB1257.namprd10.prod.outlook.com>
Message-ID: <D05BCCDD-84DC-4D72-AC79-DC41F916B89C@jerris.com>

should work the same, yes.

> On Apr 9, 2018, at 3:58 PM, Bob McCarthy <bob.mccarthy at westtel.com> wrote:
> 
> How about using the pgsql built into the core ?
>  
> From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris
> Sent: Monday, April 09, 2018 8:54 AM
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Subject: Re: [Freeswitch-users] sip_presence
>  
> put it in odbc source and its easy to access.
>  
> On Apr 7, 2018, at 12:10 PM, Bob McCarthy <bob.mccarthy at westtel.com <mailto:bob.mccarthy at westtel.com>> wrote:
>  
> Is there any way to access the sip_presence db from the api ?  Have documented a memory leak in presence FS-10808 and I would like to troubleshoot its root cause.
>  \

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From bob.mccarthy at westtel.com  Mon Apr  9 21:02:17 2018
From: bob.mccarthy at westtel.com (Bob McCarthy)
Date: Mon, 9 Apr 2018 21:02:17 +0000
Subject: [Freeswitch-users] sip_presence
In-Reply-To: <D05BCCDD-84DC-4D72-AC79-DC41F916B89C@jerris.com>
References: <DM5PR10MB1257706E7E3F69D03C25A9AA87B90@DM5PR10MB1257.namprd10.prod.outlook.com>
 <84296795-1975-424F-81DA-81DE342E69E1@jerris.com>
 <DM5PR10MB1257955865878084F2AEF7C387BF0@DM5PR10MB1257.namprd10.prod.outlook.com>
 <D05BCCDD-84DC-4D72-AC79-DC41F916B89C@jerris.com>
Message-ID: <DM5PR10MB12572B59AC58971BDA4EA7F387BF0@DM5PR10MB1257.namprd10.prod.outlook.com>

Having issues trying to access the db.

>From the wiki PostgreSQL in the core, I have set:
   <param name="core-db-dsn" value="pgsql://hostaddr=127.0.0.1
    dbname=freeswitch user=freeswitch password=&apos;&apos; options=&apos;-c
    client_min_messages=NOTICE&apos; application_name=&apos;freeswitch&apos;" />

but when I try to access the DB


?  psql -d freeswitch -U freeswitch
psql: could not connect to server: No such file or directory
        Is the server running locally and accepting
        connections on Unix domain socket "/var/run/postgresql/.s.PGSQL.5432"?

freeswitch is on debian 9

From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris
Sent: Monday, April 09, 2018 2:38 PM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] sip_presence

should work the same, yes.

On Apr 9, 2018, at 3:58 PM, Bob McCarthy <bob.mccarthy at westtel.com<mailto:bob.mccarthy at westtel.com>> wrote:

How about using the pgsql built into the core ?

From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris
Sent: Monday, April 09, 2018 8:54 AM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
Subject: Re: [Freeswitch-users] sip_presence

put it in odbc source and its easy to access.

On Apr 7, 2018, at 12:10 PM, Bob McCarthy <bob.mccarthy at westtel.com<mailto:bob.mccarthy at westtel.com>> wrote:

Is there any way to access the sip_presence db from the api ?  Have documented a memory leak in presence FS-10808 and I would like to troubleshoot its root cause.
 \

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From mike at jerris.com  Mon Apr  9 21:07:05 2018
From: mike at jerris.com (Michael Jerris)
Date: Mon, 9 Apr 2018 17:07:05 -0400
Subject: [Freeswitch-users] sip_presence
In-Reply-To: <DM5PR10MB12572B59AC58971BDA4EA7F387BF0@DM5PR10MB1257.namprd10.prod.outlook.com>
References: <DM5PR10MB1257706E7E3F69D03C25A9AA87B90@DM5PR10MB1257.namprd10.prod.outlook.com>
 <84296795-1975-424F-81DA-81DE342E69E1@jerris.com>
 <DM5PR10MB1257955865878084F2AEF7C387BF0@DM5PR10MB1257.namprd10.prod.outlook.com>
 <D05BCCDD-84DC-4D72-AC79-DC41F916B89C@jerris.com>
 <DM5PR10MB12572B59AC58971BDA4EA7F387BF0@DM5PR10MB1257.namprd10.prod.outlook.com>
Message-ID: <1ECF536A-2901-4B64-B894-D80DE5F53FD6@jerris.com>

you would have to make sure postgres is setup and configured properly and running so you can access it from psql and from freeswitch itself.

> On Apr 9, 2018, at 5:02 PM, Bob McCarthy <bob.mccarthy at westtel.com> wrote:
> 
> Having issues trying to access the db.
>  
> From the wiki PostgreSQL in the core, I have set:
>    <param name="core-db-dsn" value="pgsql://hostaddr=127.0.0.1 <pgsql://hostaddr=127.0.0.1>
>     dbname=freeswitch user=freeswitch password=&apos;&apos; options=&apos;-c
>     client_min_messages=NOTICE&apos; application_name=&apos;freeswitch&apos;" />
>  
> but when I try to access the DB
>  
> è  psql -d freeswitch -U freeswitch
> psql: could not connect to server: No such file or directory
>         Is the server running locally and accepting
>         connections on Unix domain socket "/var/run/postgresql/.s.PGSQL.5432"?
>  
> freeswitch is on debian 9

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From freeswitch-users at enlai.net  Mon Apr  9 21:35:05 2018
From: freeswitch-users at enlai.net (E Chu)
Date: Mon, 09 Apr 2018 21:35:05 +0000
Subject: [Freeswitch-users] How to get 200 Contact to use FQDN instead
	of IP
In-Reply-To: <mailman.164.1523290666.1211.freeswitch-users@lists.freeswitch.org>
References: <mailman.164.1523290666.1211.freeswitch-users@lists.freeswitch.org>
Message-ID: <CAJ_-ZtufB9J+A0UBucRf_2vAy4sor+2s+YRtNStJVwbNkHm76A@mail.gmail.com>

Hi Michael,

The SIP client is an iOS app behind a NAT64 network interfacing to
Freeswitch which is on an IPV4 network.

In order for the 200 from FS to be properly acknowledged, the client needs
to send the ACK to the host in the Contact field of the 200.

However, if the Contact of the 200 contains the IPV4 address of FS, the
client isn't able to send the ACK to the IPV4 address from the NAT64
network.

So we're hoping that we can get FS to send its FQDN in the Contact field of
the 200.

Any suggestions appreciated.

Thanks!


---------- Forwarded message ----------
> From: Michael Jerris <mike at jerris.com>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Cc:
> Bcc:
> Date: Mon, 9 Apr 2018 12:17:21 -0400
> Subject: Re: [Freeswitch-users] How to get 200 Contact to use FQDN instead
> of IP
> why would you want to do this?
>
> On Apr 9, 2018, at 12:33 AM, E Chu <freeswitch-users at enlai.net> wrote:
>
> Hi all,
>
> Does anyone know how I can get FS to respond with a FQDN instead of an IP
> address in the contact field of the 200?
>
> The domain is set to FQDN in vars.xml:
>
> [hostnames and IPs changed for public list privacy]
>
> <X-PRE-PROCESS cmd=“set” data="domain=fs-dev1.sipserver.com"/>
>
> i.e. but this is what FS is sending in the 200:
>
> Contact: <sip:123123123 at 1.2.3.4:5060;transport=tcp>
>
> I’d like it to be:
>
> Contact: <sip:123123123 at fs-dev1.sipserver.com:5060;transport=tcp>
>
> This is the full 200 response from FS:
>
> SIP/2.0 200 OK
> Via: SIP/2.0/TCP 192.168.0.1:52631
> ;rport=42810;branch=z9hG4bKPjwPkMUAMcou1XPnDrrWJH9GDwdlaPKXCF;alias;received=2.3.4.5
> From: "1020" <sip:1020 at fs-dev1.sipserver.com
> >;tag=h9ydSS-jO2dQ7xT8yxpVcLu-appl6Xtl
> To: <sip:123123123 at fs-dev1.sipserver.com>;tag=vXtN8yZgtHy0B
> Call-ID: Y3tWlOhNz3f9bt-uHO-LjFjh2RQwQh9l
> CSeq: 13167 INVITE
> Contact: <sip:123123123 at 1.2.3.4:5060;transport=tcp> //HOW TO GET THIS TO
> USE FQDN
> User-Agent: FreeSWITCH-mod_sofia/1.6.16~64bit
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
> REFER, NOTIFY, PUBLISH, SUBSCRIBE
> Require: timer
> 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
> Session-Expires: 120;refresher=uac
> Content-Type: application/sdp
> Content-Disposition: session
> Content-Length: 248
> Remote-Party-ID: “123123123" <sip:123123123 at fs-dev1.sipserver.com
> >;party=calling;privacy=off;screen=no
>
> v=0
> o=FreeSWITCH 1522636701 1522636702 IN IP4 1.2.3.4
> s=FreeSWITCH
> c=IN IP4 1.2.3.4
> t=0 0
> m=audio 18966 RTP/AVP 0 96
> a=rtpmap:0 PCMU/8000
> a=rtpmap:96 telephone-event/8000
> a=fmtp:96 0-16
> a=ptime:20
> a=rtcp:18967 IN IP4 1.2.3.4
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 bob.mccarthy at westtel.com  Mon Apr  9 22:16:28 2018
From: bob.mccarthy at westtel.com (Bob McCarthy)
Date: Mon, 9 Apr 2018 22:16:28 +0000
Subject: [Freeswitch-users] sip_presence
In-Reply-To: <1ECF536A-2901-4B64-B894-D80DE5F53FD6@jerris.com>
References: <DM5PR10MB1257706E7E3F69D03C25A9AA87B90@DM5PR10MB1257.namprd10.prod.outlook.com>
 <84296795-1975-424F-81DA-81DE342E69E1@jerris.com>
 <DM5PR10MB1257955865878084F2AEF7C387BF0@DM5PR10MB1257.namprd10.prod.outlook.com>
 <D05BCCDD-84DC-4D72-AC79-DC41F916B89C@jerris.com>
 <DM5PR10MB12572B59AC58971BDA4EA7F387BF0@DM5PR10MB1257.namprd10.prod.outlook.com>
 <1ECF536A-2901-4B64-B894-D80DE5F53FD6@jerris.com>
Message-ID: <DM5PR10MB12573338F25F8DB35642E62087BF0@DM5PR10MB1257.namprd10.prod.outlook.com>

Just using the freeswitch installation to install the core pgsql in accordance with the wiki.

sudo apt-get install libpq-dev
./configure --enable-core-pgsql-support

make – install et al

Made addition to
autoload_configs/switch.conf.xml as previously noted.


I also installed a pgsql client so that I could access the db:
sudo apt-get install postgresql-client

Any tips on how to see if postgres is running from freeswitch ?

ps auxwww | grep postgres shows postgres is not running under that name.

Bob


From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris
Sent: Monday, April 09, 2018 3:07 PM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] sip_presence

you would have to make sure postgres is setup and configured properly and running so you can access it from psql and from freeswitch itself.

On Apr 9, 2018, at 5:02 PM, Bob McCarthy <bob.mccarthy at westtel.com<mailto:bob.mccarthy at westtel.com>> wrote:

Having issues trying to access the db.

From the wiki PostgreSQL in the core, I have set:
   <param name="core-db-dsn" value="pgsql://hostaddr=127.0.0.1
    dbname=freeswitch user=freeswitch password=&apos;&apos; options=&apos;-c
    client_min_messages=NOTICE&apos; application_name=&apos;freeswitch&apos;" />

but when I try to access the DB

==>  psql -d freeswitch -U freeswitch
psql: could not connect to server: No such file or directory
        Is the server running locally and accepting
        connections on Unix domain socket "/var/run/postgresql/.s.PGSQL.5432"?

freeswitch is on debian 9

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From mike at jerris.com  Mon Apr  9 23:05:30 2018
From: mike at jerris.com (Michael Jerris)
Date: Mon, 9 Apr 2018 19:05:30 -0400
Subject: [Freeswitch-users] sip_presence
In-Reply-To: <DM5PR10MB12573338F25F8DB35642E62087BF0@DM5PR10MB1257.namprd10.prod.outlook.com>
References: <DM5PR10MB1257706E7E3F69D03C25A9AA87B90@DM5PR10MB1257.namprd10.prod.outlook.com>
 <84296795-1975-424F-81DA-81DE342E69E1@jerris.com>
 <DM5PR10MB1257955865878084F2AEF7C387BF0@DM5PR10MB1257.namprd10.prod.outlook.com>
 <D05BCCDD-84DC-4D72-AC79-DC41F916B89C@jerris.com>
 <DM5PR10MB12572B59AC58971BDA4EA7F387BF0@DM5PR10MB1257.namprd10.prod.outlook.com>
 <1ECF536A-2901-4B64-B894-D80DE5F53FD6@jerris.com>
 <DM5PR10MB12573338F25F8DB35642E62087BF0@DM5PR10MB1257.namprd10.prod.outlook.com>
Message-ID: <8FE85A19-B61A-47A6-B77D-E4EEF22A6AC5@jerris.com>

you still need to setup postgres server.

> On Apr 9, 2018, at 6:16 PM, Bob McCarthy <bob.mccarthy at westtel.com> wrote:
> 
> Just using the freeswitch installation to install the core pgsql in accordance with the wiki.
>  
> sudo apt-get install libpq-dev
> ./configure --enable-core-pgsql-support
>  
> make – install et al
>  
> Made addition to
> autoload_configs/switch.conf.xml as previously noted.
>  
>  
> I also installed a pgsql client so that I could access the db:
> sudo apt-get install postgresql-client
>  
> Any tips on how to see if postgres is running from freeswitch ?
>  
> ps auxwww | grep postgres shows postgres is not running under that name.
>  
> Bob
>  
>  
> From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris
> Sent: Monday, April 09, 2018 3:07 PM
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Subject: Re: [Freeswitch-users] sip_presence
>  
> you would have to make sure postgres is setup and configured properly and running so you can access it from psql and from freeswitch itself.
>  
> On Apr 9, 2018, at 5:02 PM, Bob McCarthy <bob.mccarthy at westtel.com <mailto:bob.mccarthy at westtel.com>> wrote:
>  
> Having issues trying to access the db.
>  
> From the wiki PostgreSQL in the core, I have set:
>    <param name="core-db-dsn" value="pgsql://hostaddr=127.0.0.1 <pgsql://hostaddr=127.0.0.1>
>     dbname=freeswitch user=freeswitch password=&apos;&apos; options=&apos;-c
>     client_min_messages=NOTICE&apos; application_name=&apos;freeswitch&apos;" />
>  
> but when I try to access the DB
>  
> è  psql -d freeswitch -U freeswitch
> psql: could not connect to server: No such file or directory
>         Is the server running locally and accepting
>         connections on Unix domain socket "/var/run/postgresql/.s.PGSQL.5432"?
>  
> freeswitch is on debian 9
>  
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> http://www.cluecon.com <http://www.cluecon.com/>
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> http://www.freeswitch.org <http://www.freeswitch.org/>
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From bob.mccarthy at westtel.com  Tue Apr 10 00:39:42 2018
From: bob.mccarthy at westtel.com (Bob McCarthy)
Date: Tue, 10 Apr 2018 00:39:42 +0000
Subject: [Freeswitch-users] sip_presence
In-Reply-To: <8FE85A19-B61A-47A6-B77D-E4EEF22A6AC5@jerris.com>
References: <DM5PR10MB1257706E7E3F69D03C25A9AA87B90@DM5PR10MB1257.namprd10.prod.outlook.com>
 <84296795-1975-424F-81DA-81DE342E69E1@jerris.com>
 <DM5PR10MB1257955865878084F2AEF7C387BF0@DM5PR10MB1257.namprd10.prod.outlook.com>
 <D05BCCDD-84DC-4D72-AC79-DC41F916B89C@jerris.com>
 <DM5PR10MB12572B59AC58971BDA4EA7F387BF0@DM5PR10MB1257.namprd10.prod.outlook.com>
 <1ECF536A-2901-4B64-B894-D80DE5F53FD6@jerris.com>
 <DM5PR10MB12573338F25F8DB35642E62087BF0@DM5PR10MB1257.namprd10.prod.outlook.com>
 <8FE85A19-B61A-47A6-B77D-E4EEF22A6AC5@jerris.com>
Message-ID: <DM5PR10MB12577E9CECA175E52B3664D687BE0@DM5PR10MB1257.namprd10.prod.outlook.com>

Thanks.  I’ll get that set up.

From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris
Sent: Monday, April 09, 2018 5:06 PM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] sip_presence

you still need to setup postgres server.

On Apr 9, 2018, at 6:16 PM, Bob McCarthy <bob.mccarthy at westtel.com<mailto:bob.mccarthy at westtel.com>> wrote:

Just using the freeswitch installation to install the core pgsql in accordance with the wiki.

sudo apt-get install libpq-dev
./configure --enable-core-pgsql-support

make – install et al

Made addition to
autoload_configs/switch.conf.xml as previously noted.


I also installed a pgsql client so that I could access the db:
sudo apt-get install postgresql-client

Any tips on how to see if postgres is running from freeswitch ?

ps auxwww | grep postgres shows postgres is not running under that name.

Bob


From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris
Sent: Monday, April 09, 2018 3:07 PM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
Subject: Re: [Freeswitch-users] sip_presence

you would have to make sure postgres is setup and configured properly and running so you can access it from psql and from freeswitch itself.

On Apr 9, 2018, at 5:02 PM, Bob McCarthy <bob.mccarthy at westtel.com<mailto:bob.mccarthy at westtel.com>> wrote:

Having issues trying to access the db.

From the wiki PostgreSQL in the core, I have set:
   <param name="core-db-dsn" value="pgsql://hostaddr=127.0.0.1
    dbname=freeswitch user=freeswitch password=&apos;&apos; options=&apos;-c
    client_min_messages=NOTICE&apos; application_name=&apos;freeswitch&apos;" />

but when I try to access the DB

==>  psql -d freeswitch -U freeswitch
psql: could not connect to server: No such file or directory
        Is the server running locally and accepting
        connections on Unix domain socket "/var/run/postgresql/.s.PGSQL.5432"?

freeswitch is on debian 9

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

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

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

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From bob.mccarthy at westtel.com  Tue Apr 10 00:41:53 2018
From: bob.mccarthy at westtel.com (Bob McCarthy)
Date: Tue, 10 Apr 2018 00:41:53 +0000
Subject: [Freeswitch-users] SIP Presence errors
Message-ID: <DM5PR10MB125787DF905BC0A3DA6DB79D87BE0@DM5PR10MB1257.namprd10.prod.outlook.com>

Getting a lot of these errors when running in debug presence.
Any idea of what's wrong ?

2018-04-09 18:29:04.906215 [ERR] sofia_presence.c:2308 SEND PRES NOTIFY:
file[sofia_presence.c]
func[broadsoft_sla_notify_callback]
line[3380]
profile[internal]
via[]
ip[192.168.57.101]
port[5060]
route[(null)]
contact["Dispatch 1" <sip:SharedLineFour at 192.168.57.101>]
to[<sip:SharedLineFour at 192.168.57.218:5060>;tag=rBf5t5vnTDXD;tag=rBf5t5vnTDXD;tag=rBf5t5vnTDXD]
from["Dispatch 1" <sip:SharedLineFour at 192.168.57.218>;tag=12E7BD9F-120FDB1E]
url[sip:SharedLineFour at 192.168.57.101]
call_id[2cc83dc799e9fe8fa319dfc0f06948f2]
expires_str[]
event[call-info]
ct[]
pl[]
call_info[<sip:192.168.57.218>;appearance-index=*;appearance-state=idle]
exptime[55]
2018-04-09 18:29:05.486200 [ERR] sofia_presence.c:3775 DELTA 55
2018-04-09 18:29:05.486200 [ERR] sofia_presence.c:3897 check subs sql: select contact from sip_subscriptions where call_id='408b2bcd-50fcd4ce-f6bba5b3 at 192.168.57.103' and profile_name='internal' and hostname='CO-999-9' ["Dispatch 1" <sip:SharedLineTwo at 192.168.57.103>]
2018-04-09 18:29:05.486200 [ERR] sofia_presence.c:3931 re-subscribe event call-info, sql: update sip_subscriptions set expires=1523320200, network_ip='192.168.57.103',network_port='5060',sip_user='SharedLineTwo',sip_host='192.168.57.218',full_via='SIP/2.0/UDP 192.168.57.103;branch=z9hG4bKc04f50ae65841513;rport=5060',full_to='<sip:SharedLineTwo at 192.168.57.218:5060>;tag=5C0qZydXfunD;tag=5C0qZydXfunD;tag=5C0qZydXfunD',full_from='"Dispatch 1" <sip:SharedLineTwo at 192.168.57.218>;tag=DCA332F6-E519577B',contact='"Dispatch 1" <sip:SharedLineTwo at 192.168.57.103>' where call_id='408b2bcd-50fcd4ce-f6bba5b3 at 192.168.57.103' and profile_name='internal' and hostname='CO-999-9'
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From jpinder at outlook.com  Tue Apr 10 01:03:52 2018
From: jpinder at outlook.com (Adrian Pinder)
Date: Tue, 10 Apr 2018 01:03:52 +0000
Subject: [Freeswitch-users] presence_id and call park
Message-ID: <BN7PR13MB230630D976E6F0DD7CE7D14CB2BE0@BN7PR13MB2306.namprd13.prod.outlook.com>

Hi,


Thanks for reading my question.


I'm running the latest version of freeswitch 1.6.20 and now I'm having an issue with call park and presence.


I'm using yealink phones and before 1.6.20 I was able to place a call in predefined parking lot slot and have the call park button assigned to the button go from green to red.


After researching I found out the new version of freeswitch adds the presence_id of the extension that placed the call instead of the presence_id of the park extension which is what I want to change the color of the lamp from green to red.



Any help is appreciated.





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From koketsom at tluka.co.za  Mon Apr  9 18:55:31 2018
From: koketsom at tluka.co.za (koketsom)
Date: Mon, 09 Apr 2018 20:55:31 +0200
Subject: [Freeswitch-users] caller redirect
Message-ID: <20180409185539.DDAA0407EE@relay.mailchannels.net>

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From paul.muaddib83 at gmail.com  Tue Apr 10 08:36:31 2018
From: paul.muaddib83 at gmail.com (Paul Muaddib)
Date: Tue, 10 Apr 2018 10:36:31 +0200
Subject: [Freeswitch-users] caller redirect
In-Reply-To: <20180409185539.DDAA0407EE@relay.mailchannels.net>
References: <20180409185539.DDAA0407EE@relay.mailchannels.net>
Message-ID: <CAEjNe_FW6JM-iNonzPfip2tzvmX9DCZc8rphCqB4FJUAJ_YcmA@mail.gmail.com>

I am not sure if you got me right. I don't want to port our number to
another provider. We have a sip trunk connection with a 100 number block

What I want is: For example someone calls our number with extension +49 XXX
XXX - 23 Then I want to redirect the caller to the US number from our
service provider +1 XXX XXX XXX
When I ask our provider they offered me a dedicated number but with costs
per minuate for us when someone calls this number. This would not be so
bad. But when I could manage the redirection by my own the I would be more
flexiable because I could route callers depending of there origin to
different service providers.

Regards

2018-04-09 20:55 GMT+02:00 koketsom <koketsom at tluka.co.za>:

> Is the number terrestrial number or specific to a provider? Porting is
> possible in South Africa not sure around your country.. porting is possible
> maybe enquire with your current provider is they can't do a porting most
> providers have PoP's on most countries.
>
> Regarfs
>
> Sent from my Huawei Mobile
>
>
> -------- Original Message --------
> Subject: [Freeswitch-users] caller redirect
> From: Paul Muaddib
> To: FreeSWITCH Users Help
> CC:
>
>
> Hi,
>
> one of our products has a service number which we purchase externally from
> a service provider. Now we don't want to change the numbers on our
> documents every time we change the service provider. Our company is located
> in Germany, the current service provider in the USA. Can I simply forward
> the incoming calls to the phone number in the USA with the "deflect"
> application without any costs for us?
>
> What about consumer protection? For example, the customer calls from
> Germany or Europe, but then has to make an international call with
> additional costs.
>
>
> Regards,
>
> Paul
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 bob.mccarthy at westtel.com  Tue Apr 10 11:12:50 2018
From: bob.mccarthy at westtel.com (Bob McCarthy)
Date: Tue, 10 Apr 2018 11:12:50 +0000
Subject: [Freeswitch-users] database schemas
Message-ID: <DM5PR10MB1257C0D71428CA870BD5A6DB87BE0@DM5PR10MB1257.namprd10.prod.outlook.com>

Where are the schemas for the freeswitch databases documented ?
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From ksrigo at gmail.com  Tue Apr 10 17:43:23 2018
From: ksrigo at gmail.com (Srigo Kanapathipillai)
Date: Tue, 10 Apr 2018 17:43:23 +0000
Subject: [Freeswitch-users] caller redirect
In-Reply-To: <CAEjNe_FW6JM-iNonzPfip2tzvmX9DCZc8rphCqB4FJUAJ_YcmA@mail.gmail.com>
References: <20180409185539.DDAA0407EE@relay.mailchannels.net>
 <CAEjNe_FW6JM-iNonzPfip2tzvmX9DCZc8rphCqB4FJUAJ_YcmA@mail.gmail.com>
Message-ID: <CAAiAxuzwV5bGmCdX49GatnkJ7vb7Z4MDSG+=HXzbUmxr2A7Qew@mail.gmail.com>

Hi,

Im not sure to understd everything, if im not you want to send back 302
redirect to your caller to your US number, this mean your sip provider
accepts 302 sip messages and act as your caller and will send back to your
US number a new invite (but not sure all sip provider will do it).

The second things you can do when you get the first call you can use the
dialplan app transfer (or execute in extension) to send your call to
another dialplan extension, there you can do a bridge to your US number
using an external gateway (if this number is not managed by the same
freeswitch)

Regards
Srigo


On Tue, 10 Apr 2018, 19:09 Paul Muaddib, <paul.muaddib83 at gmail.com> wrote:

> I am not sure if you got me right. I don't want to port our number to
> another provider. We have a sip trunk connection with a 100 number block
>
> What I want is: For example someone calls our number with extension +49
> XXX XXX - 23 Then I want to redirect the caller to the US number from our
> service provider +1 XXX XXX XXX
> When I ask our provider they offered me a dedicated number but with costs
> per minuate for us when someone calls this number. This would not be so
> bad. But when I could manage the redirection by my own the I would be more
> flexiable because I could route callers depending of there origin to
> different service providers.
>
> Regards
>
> 2018-04-09 20:55 GMT+02:00 koketsom <koketsom at tluka.co.za>:
>
>> Is the number terrestrial number or specific to a provider? Porting is
>> possible in South Africa not sure around your country.. porting is possible
>> maybe enquire with your current provider is they can't do a porting most
>> providers have PoP's on most countries.
>>
>> Regarfs
>>
>> Sent from my Huawei Mobile
>>
>>
>> -------- Original Message --------
>> Subject: [Freeswitch-users] caller redirect
>> From: Paul Muaddib
>> To: FreeSWITCH Users Help
>> CC:
>>
>>
>> Hi,
>>
>> one of our products has a service number which we purchase externally
>> from a service provider. Now we don't want to change the numbers on our
>> documents every time we change the service provider. Our company is located
>> in Germany, the current service provider in the USA. Can I simply forward
>> the incoming calls to the phone number in the USA with the "deflect"
>> application without any costs for us?
>>
>> What about consumer protection? For example, the customer calls from
>> Germany or Europe, but then has to make an international call with
>> additional costs.
>>
>>
>> Regards,
>>
>> Paul
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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 shakumarsoftware at gmail.com  Tue Apr 10 21:26:54 2018
From: shakumarsoftware at gmail.com (Sharath Kumar)
Date: Tue, 10 Apr 2018 15:26:54 -0600
Subject: [Freeswitch-users] 1 legged call tranfer hacks
Message-ID: <CA+wj8uZmNTukbS30HTX-j_ZLQ0E0m9NcKhSZRhgv5=wSwsqDMw@mail.gmail.com>

Hello,

I use Freeswitch as a client. I am on version 1.6.20. I originate an
outbound call via a SIp gateway the after the call is connected the far end
(B party) sends me(the Freeswitch client-  A party) a REFER to C. I
understand that this is not supported. I was reading about loopback and
also sofia loopback will help to have the extra dummy channel but it is not
clear how to use them in this scenario. I just have an originate command to
work with that I origiate to a SIP gateway. Should I first originate to a
loopback and park it and then bridge it ? Any hints or suggenstions will be
greatly appreciated?
thank you,
Shaks.
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From ksrigo at gmail.com  Wed Apr 11 05:23:20 2018
From: ksrigo at gmail.com (Srigo Kanapathipillai)
Date: Wed, 11 Apr 2018 05:23:20 +0000
Subject: [Freeswitch-users] 1 legged call tranfer hacks
In-Reply-To: <CA+wj8uZmNTukbS30HTX-j_ZLQ0E0m9NcKhSZRhgv5=wSwsqDMw@mail.gmail.com>
References: <CA+wj8uZmNTukbS30HTX-j_ZLQ0E0m9NcKhSZRhgv5=wSwsqDMw@mail.gmail.com>
Message-ID: <CAAiAxuz6W1kEN9ttmLF5kXMdH5u0c81uxA_cym2jgU4dX+rnrQ@mail.gmail.com>

Hi

Did you try to use in your orginate command
{force_transfer_context=transfer_cxt}?

Like this when you get a refer from your B leg, you catch the refer and
generate a bridge to refer_to_uri using your sip gateway in the above
dialplan context.

I know it works with unattended refer message (without Replaces in refer_to
uri) but i guess you can manipulate sip headers to make it work with
attended transfer.

Srigo

On Wed, 11 Apr 2018, 07:47 Sharath Kumar, <shakumarsoftware at gmail.com>
wrote:

> Hello,
>
> I use Freeswitch as a client. I am on version 1.6.20. I originate an
> outbound call via a SIp gateway the after the call is connected the far end
> (B party) sends me(the Freeswitch client-  A party) a REFER to C. I
> understand that this is not supported. I was reading about loopback and
> also sofia loopback will help to have the extra dummy channel but it is not
> clear how to use them in this scenario. I just have an originate command to
> work with that I origiate to a SIP gateway. Should I first originate to a
> loopback and park it and then bridge it ? Any hints or suggenstions will be
> greatly appreciated?
> thank you,
> Shaks.
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 shakumarsoftware at gmail.com  Wed Apr 11 21:19:42 2018
From: shakumarsoftware at gmail.com (Sharath Kumar)
Date: Wed, 11 Apr 2018 15:19:42 -0600
Subject: [Freeswitch-users] 1 legged call tranfer hacks
In-Reply-To: <CAAiAxuz6W1kEN9ttmLF5kXMdH5u0c81uxA_cym2jgU4dX+rnrQ@mail.gmail.com>
References: <CA+wj8uZmNTukbS30HTX-j_ZLQ0E0m9NcKhSZRhgv5=wSwsqDMw@mail.gmail.com>
 <CAAiAxuz6W1kEN9ttmLF5kXMdH5u0c81uxA_cym2jgU4dX+rnrQ@mail.gmail.com>
Message-ID: <CA+wj8ub0utnuXHxWrBAx=6pMVFq=jFNn2x-n55c4UWqjDsaXuA@mail.gmail.com>

Srigo,

Thanks for responding!  I tried it but it didn't work. I am directly
dialing out to a gateway with the originate. Mod_sofia rejects the REFER.
Since there is only 1 leg the transfer fails. I know I have to loop through
using a loopback or use a SIP profile listening on 127.0.0.1 but I am
unsure how the routing works.

On Tue, Apr 10, 2018 at 11:23 PM, Srigo Kanapathipillai <ksrigo at gmail.com>
wrote:

> Hi
>
> Did you try to use in your orginate command {force_transfer_context=
> transfer_cxt}?
>
> Like this when you get a refer from your B leg, you catch the refer and
> generate a bridge to refer_to_uri using your sip gateway in the above
> dialplan context.
>
> I know it works with unattended refer message (without Replaces in
> refer_to uri) but i guess you can manipulate sip headers to make it work
> with attended transfer.
>
> Srigo
>
> On Wed, 11 Apr 2018, 07:47 Sharath Kumar, <shakumarsoftware at gmail.com>
> wrote:
>
>> Hello,
>>
>> I use Freeswitch as a client. I am on version 1.6.20. I originate an
>> outbound call via a SIp gateway the after the call is connected the far end
>> (B party) sends me(the Freeswitch client-  A party) a REFER to C. I
>> understand that this is not supported. I was reading about loopback and
>> also sofia loopback will help to have the extra dummy channel but it is not
>> clear how to use them in this scenario. I just have an originate command to
>> work with that I origiate to a SIP gateway. Should I first originate to a
>> loopback and park it and then bridge it ? Any hints or suggenstions will be
>> greatly appreciated?
>> thank you,
>> Shaks.
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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 bob.mccarthy at westtel.com  Wed Apr 11 21:49:11 2018
From: bob.mccarthy at westtel.com (Bob McCarthy)
Date: Wed, 11 Apr 2018 21:49:11 +0000
Subject: [Freeswitch-users] database schemas
In-Reply-To: <DM5PR10MB1257C0D71428CA870BD5A6DB87BE0@DM5PR10MB1257.namprd10.prod.outlook.com>
References: <DM5PR10MB1257C0D71428CA870BD5A6DB87BE0@DM5PR10MB1257.namprd10.prod.outlook.com>
Message-ID: <MWHPR10MB12610019ACA664B8F2878D2E87BD0@MWHPR10MB1261.namprd10.prod.outlook.com>

Looking specifically for sip_presence .....

From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Bob McCarthy
Sent: Tuesday, April 10, 2018 5:13 AM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: [Freeswitch-users] database schemas


This sender failed our fraud detection checks and may not be who they appear to be. Learn about spoofing<http://aka.ms/LearnAboutSpoofing>

Feedback<http://aka.ms/SafetyTipsFeedback>

Where are the schemas for the freeswitch databases documented ?
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From michael at mailworks.org  Wed Apr 11 21:52:41 2018
From: michael at mailworks.org (Michael Avers)
Date: Wed, 11 Apr 2018 14:52:41 -0700
Subject: [Freeswitch-users] database schemas
In-Reply-To: <MWHPR10MB12610019ACA664B8F2878D2E87BD0@MWHPR10MB1261.namprd10.prod.outlook.com>
References: <DM5PR10MB1257C0D71428CA870BD5A6DB87BE0@DM5PR10MB1257.namprd10.prod.outlook.com>
 <MWHPR10MB12610019ACA664B8F2878D2E87BD0@MWHPR10MB1261.namprd10.prod.outlook.com>
Message-ID: <1523483561.1936113.1334909760.02B35A0E@webmail.messagingengine.com>

Here you go:

https://github.com/fusionpbx/fusionpbx/blob/master/resources/install/sql/switch.sql

On Wed, Apr 11, 2018, at 2:49 PM, Bob McCarthy wrote:
> Looking specifically for sip_presence …..


>  


> *From:* FreeSWITCH-users [mailto:freeswitch-users-
> bounces at lists.freeswitch.org] *On Behalf Of *Bob McCarthy *Sent:*
> Tuesday, April 10, 2018 5:13 AM *To:* FreeSWITCH Users Help <freeswitch-
> users at lists.freeswitch.org> *Subject:* [Freeswitch-users]
> database schemas>  


> 
> This sender failed our fraud detection checks and may not be who they
> appear to be. Learn about spoofing[1]> Feedback[2]


> Where are the schemas for the freeswitch databases documented ?


> ___________________________________________________________________-
> ________> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>  
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>  
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users> http://www.freeswitch.org


Links:

  1. http://aka.ms/LearnAboutSpoofing
  2. http://aka.ms/SafetyTipsFeedback
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From bob.mccarthy at westtel.com  Wed Apr 11 23:29:13 2018
From: bob.mccarthy at westtel.com (Bob McCarthy)
Date: Wed, 11 Apr 2018 23:29:13 +0000
Subject: [Freeswitch-users] database schemas
In-Reply-To: <1523483561.1936113.1334909760.02B35A0E@webmail.messagingengine.com>
References: <DM5PR10MB1257C0D71428CA870BD5A6DB87BE0@DM5PR10MB1257.namprd10.prod.outlook.com>
 <MWHPR10MB12610019ACA664B8F2878D2E87BD0@MWHPR10MB1261.namprd10.prod.outlook.com>
 <1523483561.1936113.1334909760.02B35A0E@webmail.messagingengine.com>
Message-ID: <DM5PR10MB1257A8F80CA56B654B4CE50D87BD0@DM5PR10MB1257.namprd10.prod.outlook.com>

Thanks ☺

From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Avers
Sent: Wednesday, April 11, 2018 3:53 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] database schemas

Here you go:

https://github.com/fusionpbx/fusionpbx/blob/master/resources/install/sql/switch.sql


On Wed, Apr 11, 2018, at 2:49 PM, Bob McCarthy wrote:

Looking specifically for sip_presence …..



From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Bob McCarthy
Sent: Tuesday, April 10, 2018 5:13 AM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
Subject: [Freeswitch-users] database schemas




This sender failed our fraud detection checks and may not be who they appear to be. Learn about spoofing<http://aka.ms/LearnAboutSpoofing>


Feedback<http://aka.ms/SafetyTipsFeedback>


Where are the schemas for the freeswitch databases documented ?
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com

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

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

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From igor.potjevlesch at gmail.com  Thu Apr 12 13:57:19 2018
From: igor.potjevlesch at gmail.com (igor.potjevlesch at gmail.com)
Date: Thu, 12 Apr 2018 15:57:19 +0200
Subject: [Freeswitch-users] RPID replaced by P-Prefered-Identity
Message-ID: <004401d3d266$2c63c6f0$852b54d0$@gmail.com>

Hello,


I have a strange case: a customer who uses a Cisco IP Phone places a call
with Remote-Party-Id. The core network adds P-Asserted-Identity.

So, when the INVITE reach Freeswitch, he has 1 PAI and 1 RPID. Freeswitch
strips both and replaces with PPI. 

 

Any idea how to fix it and keep only PAI? Interfaces are setup with pid as
caller id and before bridging the call, I do:

<action application="unset" data="sip_h_Remote-Party-ID"/>

 

Regards,

 

Igor.

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From wayne at hamilton.net  Thu Apr 12 18:36:11 2018
From: wayne at hamilton.net (Wayne Hahn)
Date: Thu, 12 Apr 2018 13:36:11 -0500
Subject: [Freeswitch-users] RPID replaced by P-Prefered-Identity
In-Reply-To: <004401d3d266$2c63c6f0$852b54d0$@gmail.com>
References: <004401d3d266$2c63c6f0$852b54d0$@gmail.com>
Message-ID: <009b01d3d28d$213e2960$63ba7c20$@hamilton.net>

First I have heard of it.

 

Wayne

 

From: FreeSWITCH-users
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
igor.potjevlesch at gmail.com
Sent: Thursday, April 12, 2018 8:57 AM
To: 'FreeSWITCH Users Help'
Subject: [Freeswitch-users] RPID replaced by P-Prefered-Identity

 

Hello,


I have a strange case: a customer who uses a Cisco IP Phone places a call
with Remote-Party-Id. The core network adds P-Asserted-Identity.

So, when the INVITE reach Freeswitch, he has 1 PAI and 1 RPID. Freeswitch
strips both and replaces with PPI. 

 

Any idea how to fix it and keep only PAI? Interfaces are setup with pid as
caller id and before bridging the call, I do:

<action application="unset" data="sip_h_Remote-Party-ID"/>

 

Regards,

 

Igor.

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From ksrigo at gmail.com  Fri Apr 13 10:57:57 2018
From: ksrigo at gmail.com (Srigo Kanapathipillai)
Date: Fri, 13 Apr 2018 10:57:57 +0000
Subject: [Freeswitch-users] RPID replaced by P-Prefered-Identity
In-Reply-To: <004401d3d266$2c63c6f0$852b54d0$@gmail.com>
References: <004401d3d266$2c63c6f0$852b54d0$@gmail.com>
Message-ID: <CAAiAxuywWv_nJME5cZC+RN9zEmPN1VWHURnwVigo58urBzvrmw@mail.gmail.com>

Hi

In your profile receiving the INVITE, try to put this:
<param name="cid-type" value="pid"/>

Or you can do it manually in your dialplan

https://wiki.freeswitch.org/wiki/Variable_sip_cid_type

Srigo

On Fri, 13 Apr 2018, 00:48 , <igor.potjevlesch at gmail.com> wrote:

> Hello,
>
>
> I have a strange case: a customer who uses a Cisco IP Phone places a call
> with Remote-Party-Id. The core network adds P-Asserted-Identity.
>
> So, when the INVITE reach Freeswitch, he has 1 PAI and 1 RPID. Freeswitch
> strips both and replaces with PPI.
>
>
>
> Any idea how to fix it and keep only PAI? Interfaces are setup with pid as
> caller id and before bridging the call, I do:
>
> <action application="unset" data="sip_h_Remote-Party-ID"/>
>
>
>
> Regards,
>
>
>
> Igor.
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From igor.potjevlesch at gmail.com  Fri Apr 13 16:05:19 2018
From: igor.potjevlesch at gmail.com (igor.potjevlesch at gmail.com)
Date: Fri, 13 Apr 2018 18:05:19 +0200
Subject: [Freeswitch-users] RPID replaced by P-Prefered-Identity
In-Reply-To: <CAAiAxuywWv_nJME5cZC+RN9zEmPN1VWHURnwVigo58urBzvrmw@mail.gmail.com>
References: <004401d3d266$2c63c6f0$852b54d0$@gmail.com>
 <CAAiAxuywWv_nJME5cZC+RN9zEmPN1VWHURnwVigo58urBzvrmw@mail.gmail.com>
Message-ID: <003201d3d341$38673820$a935a860$@gmail.com>

Hi Srigo,

 

Thank you for your suggestion.

Unfortunately, the customer impacted has finally changed his config to replace RPID by PAI directly. So the issue is no more reproductible.

 

Regards,

 

Igor.

 

De : FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> De la part de Srigo Kanapathipillai
Envoyé : vendredi 13 avril 2018 12:58
À : FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Objet : Re: [Freeswitch-users] RPID replaced by P-Prefered-Identity

 

Hi

 

In your profile receiving the INVITE, try to put this:

<param name="cid-type" value="pid"/>

 

Or you can do it manually in your dialplan

 

https://wiki.freeswitch.org/wiki/Variable_sip_cid_type

 

Srigo

 

On Fri, 13 Apr 2018, 00:48 , <igor.potjevlesch at gmail.com <mailto:igor.potjevlesch at gmail.com> > wrote:

Hello,


I have a strange case: a customer who uses a Cisco IP Phone places a call with Remote-Party-Id. The core network adds P-Asserted-Identity.

So, when the INVITE reach Freeswitch, he has 1 PAI and 1 RPID. Freeswitch strips both and replaces with PPI. 

 

Any idea how to fix it and keep only PAI? Interfaces are setup with pid as caller id and before bridging the call, I do:

<action application="unset" data="sip_h_Remote-Party-ID"/>

 

Regards,

 

Igor.

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

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

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

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From tculjaga at gmail.com  Mon Apr 16 11:54:16 2018
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Mon, 16 Apr 2018 13:54:16 +0200
Subject: [Freeswitch-users] mod_xml_curl - serving configuration
Message-ID: <CAJoKKdUW9DWi8MVGTYiLRtjZt+g5uwR6mLn+_c_vzvkLOmZ8og@mail.gmail.com>

Hi all!

I'd like to use mod_xml_curl for serving configurations to freeswitch. Im
wondering how to approach  X-PRE-PROCESS includes properly.



as an example, this is request for sofia.conf:

hostname=FS01&section=configuration&tag_name=configuration&key_name=name&
key_value=sofia.conf&Event-Name=REQUEST_PARAMS&Core-UUID=
118a9f3f-9ee4-4688-9ba0-a510211cd6d3&FreeSWITCH-Hostname=incfswitchtest01.
local&FreeSWITCH-Swi
tchname=FS01&FreeSWITCH-IPv4=192.168.50.66&FreeSWITCH-IPv6=
%3A%3A1&Event-Date-Local=2018-04-11%2015%3A29%3A39&Event-
Date-GMT=Wed,%2011%20Apr%202018%2013%3A29%3A39%20GMT&Event-Date-Timestamp=
1523453379037164&Event-Calling-File=sofia.c
&Event-Calling-Function=config_sofia&Event-Calling-Line-Number=4246&Event-
Sequence=19



this is sofia.conf.xml we have on the file system:


<document type="freeswitch/xml">
  <section name="configuration">
    <configuration name="sofia.conf" description="sofia Endpoint">

      <global_settings>
        <param name="log-level" value="0"/>
        <!--<param name="auto-restart" value="false"/>-->
        <param name="debug-presence" value="0"/>
        <!--<param name="capture-server" value="udp:homer.domain.com:5060
"/>-->
      </global_settings>

      <profiles>
        <X-PRE-PROCESS cmd="include" data="../sip_profiles/*.xml"/>
      </profiles>

    </configuration>
  </section>
</document>



here we have a X-PRE-PROCESS include command

Can i just set the "pointers" to profile names e.g.:

      <profiles>
        <profile name="internal"/>
        <profile name="external"/>

        <!-- any new profile adds here :=) -->
      </profiles>

and wait for a new more specific request from FS with  profile=<profile
name> to serve the entire profile configuration  ?


hostname=FS01&section=configuration&tag_name=configuration&key_name=name&
key_value=sofia.conf&Event-Name=REQUEST_PARAMS&Core-UUID=
118a9f3f-9ee4-4688-9ba0-a510211cd6d3&FreeSWITCH-Hostname=incfswitchtest01.
local&FreeSWITCH-Swi
tchname=FS01&FreeSWITCH-IPv4=192.168.50.66&FreeSWITCH-IPv6=
%3A%3A1&Event-Date-Local=2018-04-11%2015%3A29%3A42&Event-
Date-GMT=Wed,%2011%20Apr%202018%2013%3A29%3A42%20GMT&Event-Date-Timestamp=
1523453382092765&Event-Calling-File=sofia.c
&Event-Calling-Function=launch_sofia_worker_thread&
Event-Calling-Line-Number=2949&Event-Sequence=24&profile=internal


what do i return here ?
Is that correct ?

 <document>
  <section name="configuration">
    <profile name="myProfileName">

        <!-- myProfile data  goes here -->

     </profile>
    </section>
  </document>
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From vladislaus at gmail.com  Mon Apr 16 19:19:23 2018
From: vladislaus at gmail.com (Andres Gomez)
Date: Mon, 16 Apr 2018 14:19:23 -0500
Subject: [Freeswitch-users] Transfer a call direct to agent queue /
	mod_callcenter
Message-ID: <CAHzfA-gM2AUKhZKs9hnmnaK-f=J+V=VM-rU+=HigKo=yh-BA8Q@mail.gmail.com>

 Hi Friends.

With mod_callcenter can I transfer a call direct to agent queue?

Regards

Andrés Gómez
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From alihaider.4189 at gmail.com  Mon Apr 16 20:57:53 2018
From: alihaider.4189 at gmail.com (Ali Haider)
Date: Mon, 16 Apr 2018 13:57:53 -0700
Subject: [Freeswitch-users] where can i fix call recording
Message-ID: <CAB2bS9TWWi5uCUr_zFWRhx+Qm1_34-2mzLj92nfoD7xkJpMqFA@mail.gmail.com>

Hiii
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From alihaider.4189 at gmail.com  Mon Apr 16 20:59:08 2018
From: alihaider.4189 at gmail.com (Ali Haider)
Date: Mon, 16 Apr 2018 13:59:08 -0700
Subject: [Freeswitch-users] where can i fix call recording
In-Reply-To: <CAB2bS9TWWi5uCUr_zFWRhx+Qm1_34-2mzLj92nfoD7xkJpMqFA@mail.gmail.com>
References: <CAB2bS9TWWi5uCUr_zFWRhx+Qm1_34-2mzLj92nfoD7xkJpMqFA@mail.gmail.com>
Message-ID: <CAB2bS9R_iDYj1fX4Gz-SCpfmyYg_H3B1Rh3zGVdmXYBuHAD82g@mail.gmail.com>

hiiii


On 16 April 2018 at 13:57, Ali Haider <alihaider.4189 at gmail.com> wrote:

> Hiii
>
>
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From italo at freeswitch.org  Tue Apr 17 00:39:04 2018
From: italo at freeswitch.org (=?UTF-8?B?w410YWxvIFJvc3Np?=)
Date: Mon, 16 Apr 2018 21:39:04 -0300
Subject: [Freeswitch-users] Transfer a call direct to agent queue /
	mod_callcenter
In-Reply-To: <CAHzfA-gM2AUKhZKs9hnmnaK-f=J+V=VM-rU+=HigKo=yh-BA8Q@mail.gmail.com>
References: <CAHzfA-gM2AUKhZKs9hnmnaK-f=J+V=VM-rU+=HigKo=yh-BA8Q@mail.gmail.com>
Message-ID: <CABMX_w9ouuQ3+KwzVbbOJm_OrFCqS51NV4ztKfXW_Bw+GTwpHA@mail.gmail.com>

Not implemented at this time.

On Mon, Apr 16, 2018 at 4:19 PM, Andres Gomez <vladislaus at gmail.com> wrote:

> Hi Friends.
>
> With mod_callcenter can I transfer a call direct to agent queue?
>
> Regards
>
> Andrés Gómez
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Ítalo Rossi
italo at freeswitch.org
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From david.witham at netsip.com.au  Tue Apr 17 05:08:36 2018
From: david.witham at netsip.com.au (David Witham)
Date: Tue, 17 Apr 2018 15:08:36 +1000
Subject: [Freeswitch-users] 100rel support
Message-ID: <CAPY30xYiUaeXafjZmshEN2bNvgkhRK4h43_zsrW938x-MTm+wg@mail.gmail.com>

Hi all,

We have had an upstream provider request that we enable 100rel support. I
note in the wiki it says:

enable-100rel

This enable support for 100rel (100% reliability - PRACK message as defined
in RFC3262 <http://tools.ietf.org/html/rfc3262>) This fixes a problem with
SIP where provisional messages like "180 Ringing" are not ACK'd and
therefore could be dropped over a poor connection without retransmission.
*2009-07-08:* Enabling this may cause FreeSWITCH to crash, see FSCORE-392
<http://jira.freeswitch.org/browse/FSCORE-392>.

<param name="enable-100rel" value="true"/>



FSCORE-392 doesn't say if the crashing related to enabling 100rel was ever
fixed. Can anyone confirm whether it is now safe to enable 100rel support?

regards,
David
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From alex at freeswitch.com  Tue Apr 17 05:30:34 2018
From: alex at freeswitch.com (Alexey Sibyakin)
Date: Tue, 17 Apr 2018 14:30:34 +0900
Subject: [Freeswitch-users] 100rel support
In-Reply-To: <CAPY30xYiUaeXafjZmshEN2bNvgkhRK4h43_zsrW938x-MTm+wg@mail.gmail.com>
References: <CAPY30xYiUaeXafjZmshEN2bNvgkhRK4h43_zsrW938x-MTm+wg@mail.gmail.com>
Message-ID: <CAGJLwFjGF2de3w+QTvPpjHssdSuPT440ieNVccMWRPRu-CP1kA@mail.gmail.com>

Hi,

It's not safe but it may work for you. I used it for years and never had an
issue but load were relatively small.

Alex

On Tue, Apr 17, 2018 at 2:08 PM, David Witham <david.witham at netsip.com.au>
wrote:

> Hi all,
>
> We have had an upstream provider request that we enable 100rel support. I
> note in the wiki it says:
>
> enable-100rel
>
> This enable support for 100rel (100% reliability - PRACK message as
> defined in RFC3262 <http://tools.ietf.org/html/rfc3262>) This fixes a
> problem with SIP where provisional messages like "180 Ringing" are not
> ACK'd and therefore could be dropped over a poor connection without
> retransmission. *2009-07-08:* Enabling this may cause FreeSWITCH to crash,
> see FSCORE-392 <http://jira.freeswitch.org/browse/FSCORE-392>.
>
> <param name="enable-100rel" value="true"/>
>
>
>
> FSCORE-392 doesn't say if the crashing related to enabling 100rel was ever
> fixed. Can anyone confirm whether it is now safe to enable 100rel support?
>
> regards,
> David
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 Engineer
FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045
<https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g>
Email: alex at freeswitch.com
Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/>
Need commercial support? Contact sales at freeswitch.com for details.
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From igor.potjevlesch at gmail.com  Tue Apr 17 08:08:22 2018
From: igor.potjevlesch at gmail.com (igor.potjevlesch at gmail.com)
Date: Tue, 17 Apr 2018 10:08:22 +0200
Subject: [Freeswitch-users] Suddenly lost of fs_cli
Message-ID: <003c01d3d623$41492b30$c3db8190$@gmail.com>

Hello,

 

Since yesterday, I cannot access to fs_cli anymore after a traffic peak on
an instance.

 

I got the following error:

[ERROR] fs_cli.c:1610 main() Error Connecting [Socket Connection Error]

Usage: /usr/local/freeswitch/bin/fs_cli [-H <host>] [-P <port>] [-p
<secret>] [-d <level>] [-x command] [-t <timeout_ms>] [profile]

 

Is there something I can do without restarting Freeswitch?

 

Regards,

 

Igor.

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From alex at freeswitch.com  Tue Apr 17 08:45:53 2018
From: alex at freeswitch.com (Alexey Sibyakin)
Date: Tue, 17 Apr 2018 17:45:53 +0900
Subject: [Freeswitch-users] Suddenly lost of fs_cli
In-Reply-To: <003c01d3d623$41492b30$c3db8190$@gmail.com>
References: <003c01d3d623$41492b30$c3db8190$@gmail.com>
Message-ID: <CAGJLwFgiZRdUi_SJBjBb6KCXaNwK7GDYsckG0xow+YTN_S7mZg@mail.gmail.com>

Hello

Try to add -rR flags to command

Alex

On Tue, Apr 17, 2018 at 5:08 PM, <igor.potjevlesch at gmail.com> wrote:

> Hello,
>
>
>
> Since yesterday, I cannot access to fs_cli anymore after a traffic peak on
> an instance.
>
>
>
> I got the following error:
>
> [ERROR] fs_cli.c:1610 main() Error Connecting [Socket Connection Error]
>
> Usage: /usr/local/freeswitch/bin/fs_cli [-H <host>] [-P <port>] [-p
> <secret>] [-d <level>] [-x command] [-t <timeout_ms>] [profile]
>
>
>
> Is there something I can do without restarting Freeswitch?
>
>
>
> Regards,
>
>
>
> Igor.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Alex Sibyakin | Support Engineer
FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045
<https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g>
Email: alex at freeswitch.com
Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/>
Need commercial support? Contact sales at freeswitch.com for details.
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From gmaruzz at gmail.com  Tue Apr 17 09:19:05 2018
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Tue, 17 Apr 2018 11:19:05 +0200
Subject: [Freeswitch-users] Suddenly lost of fs_cli
In-Reply-To: <003c01d3d623$41492b30$c3db8190$@gmail.com>
References: <003c01d3d623$41492b30$c3db8190$@gmail.com>
Message-ID: <CALXCt0pJ36dx9i=TGbV8-qof+-eMcR11iHT3JyVEpcOoufpVvQ@mail.gmail.com>

probably your freeswitch died or is no more correctly working...

On 17 April 2018 at 10:08, <igor.potjevlesch at gmail.com> wrote:

> Hello,
>
>
>
> Since yesterday, I cannot access to fs_cli anymore after a traffic peak on
> an instance.
>
>
>
> I got the following error:
>
> [ERROR] fs_cli.c:1610 main() Error Connecting [Socket Connection Error]
>
> Usage: /usr/local/freeswitch/bin/fs_cli [-H <host>] [-P <port>] [-p
> <secret>] [-d <level>] [-x command] [-t <timeout_ms>] [profile]
>
>
>
> Is there something I can do without restarting Freeswitch?
>
>
>
> Regards,
>
>
>
> Igor.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

Sincerely,

Giovanni Maruzzelli
OpenTelecom.IT
cell: +39 347 266 56 18
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From steveayre at gmail.com  Tue Apr 17 09:41:59 2018
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 17 Apr 2018 10:41:59 +0100
Subject: [Freeswitch-users] 100rel support
In-Reply-To: <CAPY30xYiUaeXafjZmshEN2bNvgkhRK4h43_zsrW938x-MTm+wg@mail.gmail.com>
References: <CAPY30xYiUaeXafjZmshEN2bNvgkhRK4h43_zsrW938x-MTm+wg@mail.gmail.com>
Message-ID: <CAFiqYunfhx0-y+tTvjJtu-MzAeoy1B_T-4iewPiY522uy6e-UA@mail.gmail.com>

I've been wondering this for a few years too. I'm guessing that comment has
scared everyone away from enabling it to test it out.

At a guess if there's any issue it's within the Sofia-SIP stack and
therefore trickier to debug/patch than in mod_sofia.

You could enable it and if there are any problems report them on Jira so
they get looked into.

I believe 100rel might be required for IMS implementations.

On 17 April 2018 at 06:08, David Witham <david.witham at netsip.com.au> wrote:

> Hi all,
>
> We have had an upstream provider request that we enable 100rel support. I
> note in the wiki it says:
>
> enable-100rel
>
> This enable support for 100rel (100% reliability - PRACK message as
> defined in RFC3262 <http://tools.ietf.org/html/rfc3262>) This fixes a
> problem with SIP where provisional messages like "180 Ringing" are not
> ACK'd and therefore could be dropped over a poor connection without
> retransmission. *2009-07-08:* Enabling this may cause FreeSWITCH to crash,
> see FSCORE-392 <http://jira.freeswitch.org/browse/FSCORE-392>.
>
> <param name="enable-100rel" value="true"/>
>
>
>
> FSCORE-392 doesn't say if the crashing related to enabling 100rel was ever
> fixed. Can anyone confirm whether it is now safe to enable 100rel support?
>
> regards,
> David
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 igor.potjevlesch at gmail.com  Tue Apr 17 12:44:28 2018
From: igor.potjevlesch at gmail.com (igor.potjevlesch at gmail.com)
Date: Tue, 17 Apr 2018 14:44:28 +0200
Subject: [Freeswitch-users] Suddenly lost of fs_cli
In-Reply-To: <CAGJLwFgiZRdUi_SJBjBb6KCXaNwK7GDYsckG0xow+YTN_S7mZg@mail.gmail.com>
References: <003c01d3d623$41492b30$c3db8190$@gmail.com>
 <CAGJLwFgiZRdUi_SJBjBb6KCXaNwK7GDYsckG0xow+YTN_S7mZg@mail.gmail.com>
Message-ID: <005201d3d649$d320dfa0$79629ee0$@gmail.com>

Hello Alex,

 

There is a loop:

[ERROR] fs_cli.c:1610 main() Error Connecting [Socket Connection Error]

[INFO] fs_cli.c:1616 main() Retrying

[ERROR] fs_cli.c:1610 main() Error Connecting [Socket Connection Error]

[INFO] fs_cli.c:1616 main() Retrying

[ERROR] fs_cli.c:1610 main() Error Connecting [Socket Connection Error]

[INFO] fs_cli.c:1616 main() Retrying

[ERROR] fs_cli.c:1610 main() Error Connecting [Socket Connection Error]

[INFO] fs_cli.c:1616 main() Retrying

[ERROR] fs_cli.c:1610 main() Error Connecting [Socket Connection Error]

[INFO] fs_cli.c:1616 main() Retrying

[ERROR] fs_cli.c:1610 main() Error Connecting [Socket Connection Error]

[INFO] fs_cli.c:1616 main() Retrying

[ERROR] fs_cli.c:1610 main() Error Connecting [Socket Connection Error]

[INFO] fs_cli.c:1616 main() Retrying

[…]

 

Regards,

 

Igor

 

De : FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> De la part de Alexey Sibyakin
Envoyé : mardi 17 avril 2018 10:46
À : FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Objet : Re: [Freeswitch-users] Suddenly lost of fs_cli

 

Hello

 

Try to add -rR flags to command

Alex

 

On Tue, Apr 17, 2018 at 5:08 PM, <igor.potjevlesch at gmail.com <mailto:igor.potjevlesch at gmail.com> > wrote:

Hello,

 

Since yesterday, I cannot access to fs_cli anymore after a traffic peak on an instance.

 

I got the following error:

[ERROR] fs_cli.c:1610 main() Error Connecting [Socket Connection Error]

Usage: /usr/local/freeswitch/bin/fs_cli [-H <host>] [-P <port>] [-p <secret>] [-d <level>] [-x command] [-t <timeout_ms>] [profile]

 

Is there something I can do without restarting Freeswitch?

 

Regards,

 

Igor.


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

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

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





 

-- 

Alex Sibyakin | Support Engineer

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

Email:  <mailto:alex at freeswitch.com> alex at freeswitch.com

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

Need commercial support? Contact  <mailto:sales at freeswitch.com> sales at freeswitch.com for details.

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From igor.potjevlesch at gmail.com  Tue Apr 17 12:45:12 2018
From: igor.potjevlesch at gmail.com (igor.potjevlesch at gmail.com)
Date: Tue, 17 Apr 2018 14:45:12 +0200
Subject: [Freeswitch-users] Suddenly lost of fs_cli
In-Reply-To: <CALXCt0pJ36dx9i=TGbV8-qof+-eMcR11iHT3JyVEpcOoufpVvQ@mail.gmail.com>
References: <003c01d3d623$41492b30$c3db8190$@gmail.com>
 <CALXCt0pJ36dx9i=TGbV8-qof+-eMcR11iHT3JyVEpcOoufpVvQ@mail.gmail.com>
Message-ID: <005701d3d649$ed810a50$c8831ef0$@gmail.com>

Hello Giovanni,

 

Freeswitch seems to process the calls as usual according to the logs and a tcpdump on interfaces.

 

Regards,

 

Igor.

 

De : FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> De la part de Giovanni Maruzzelli
Envoyé : mardi 17 avril 2018 11:19
À : FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Objet : Re: [Freeswitch-users] Suddenly lost of fs_cli

 

probably your freeswitch died or is no more correctly working...

 

On 17 April 2018 at 10:08, <igor.potjevlesch at gmail.com <mailto:igor.potjevlesch at gmail.com> > wrote:

Hello,

 

Since yesterday, I cannot access to fs_cli anymore after a traffic peak on an instance.

 

I got the following error:

[ERROR] fs_cli.c:1610 main() Error Connecting [Socket Connection Error]

Usage: /usr/local/freeswitch/bin/fs_cli [-H <host>] [-P <port>] [-p <secret>] [-d <level>] [-x command] [-t <timeout_ms>] [profile]

 

Is there something I can do without restarting Freeswitch?

 

Regards,

 

Igor.


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

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

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




-- 


Sincerely,

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

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From raimundo.perez.cuba at gmail.com  Tue Apr 17 13:06:20 2018
From: raimundo.perez.cuba at gmail.com (=?utf-8?Q?Raimundo_P=C3=A9rez_Nieves?=)
Date: Tue, 17 Apr 2018 15:06:20 +0200
Subject: [Freeswitch-users] Suddenly lost of fs_cli
In-Reply-To: <mailman.260.1523969145.1211.freeswitch-users@lists.freeswitch.org>
References: <mailman.260.1523969145.1211.freeswitch-users@lists.freeswitch.org>
Message-ID: <3918B447-A5D3-4C67-9947-4D23015A6CC9@gmail.com>

Try this:
Go to path /usr/bin and execute ./freeswitch
It starts FS in secure mode, and you can debug the error.
In my case, it was bad close tag in html and I just shutdown FS with that error, so It did not open. Doing this gave me the idea of the problem.
Hope helps

> On Apr 17, 2018, at 2:45 PM, freeswitch-users-request at lists.freeswitch.org wrote:
> 
> Send FreeSWITCH-users mailing list submissions to
> 	freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org>
> 
> To subscribe or unsubscribe via the World Wide Web, visit
> 	http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> or, via email, send a message with subject or body 'help' to
> 	freeswitch-users-request at lists.freeswitch.org <mailto:freeswitch-users-request at lists.freeswitch.org>
> 
> You can reach the person managing the list at
> 	freeswitch-users-owner at lists.freeswitch.org <mailto:freeswitch-users-owner at lists.freeswitch.org>
> 
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of FreeSWITCH-users digest..."
> Today's Topics:
> 
>   1. Re: Suddenly lost of fs_cli (igor.potjevlesch at gmail.com <mailto:igor.potjevlesch at gmail.com>)
>   2. Re: Suddenly lost of fs_cli (igor.potjevlesch at gmail.com <mailto:igor.potjevlesch at gmail.com>)
> 
> From: <igor.potjevlesch at gmail.com <mailto:igor.potjevlesch at gmail.com>>
> Subject: Re: [Freeswitch-users] Suddenly lost of fs_cli
> Date: April 17, 2018 at 2:44:28 PM GMT+2
> To: "'FreeSWITCH Users Help'" <freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org>>
> 
> 
> Hello Alex,
>  
> There is a loop:
> [ERROR] fs_cli.c:1610 main() Error Connecting [Socket Connection Error]
> [INFO] fs_cli.c:1616 main() Retrying
> [ERROR] fs_cli.c:1610 main() Error Connecting [Socket Connection Error]
> [INFO] fs_cli.c:1616 main() Retrying
> [ERROR] fs_cli.c:1610 main() Error Connecting [Socket Connection Error]
> [INFO] fs_cli.c:1616 main() Retrying
> [ERROR] fs_cli.c:1610 main() Error Connecting [Socket Connection Error]
> [INFO] fs_cli.c:1616 main() Retrying
> [ERROR] fs_cli.c:1610 main() Error Connecting [Socket Connection Error]
> [INFO] fs_cli.c:1616 main() Retrying
> [ERROR] fs_cli.c:1610 main() Error Connecting [Socket Connection Error]
> [INFO] fs_cli.c:1616 main() Retrying
> [ERROR] fs_cli.c:1610 main() Error Connecting [Socket Connection Error]
> [INFO] fs_cli.c:1616 main() Retrying
> […]
>  
> Regards,
>  
> Igor
>  
> De : FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org>> De la part de Alexey Sibyakin
> Envoyé : mardi 17 avril 2018 10:46
> À : FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org>>
> Objet : Re: [Freeswitch-users] Suddenly lost of fs_cli
>  
> Hello
>  
> Try to add -rR flags to command
> 
> Alex
>  
> On Tue, Apr 17, 2018 at 5:08 PM, <igor.potjevlesch at gmail.com <mailto:igor.potjevlesch at gmail.com>> wrote:
> Hello,
>  
> Since yesterday, I cannot access to fs_cli anymore after a traffic peak on an instance.
>  
> I got the following error:
> [ERROR] fs_cli.c:1610 main() Error Connecting [Socket Connection Error]
> Usage: /usr/local/freeswitch/bin/fs_cli [-H <host>] [-P <port>] [-p <secret>] [-d <level>] [-x command] [-t <timeout_ms>] [profile]
>  
> Is there something I can do without restarting Freeswitch?
>  
> Regards,
>  
> Igor.
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> http://www.cluecon.com <http://www.cluecon.com/>
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> http://www.freeswitch.org <http://www.freeswitch.org/>
> 
> 
>  
> -- 
> Alex Sibyakin | Support Engineer
> FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g>
> Email: alex at freeswitch.com <mailto:alex at freeswitch.com>
> Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/>
> Need commercial support? Contact sales at freeswitch.com <mailto:sales at freeswitch.com> for details.
> 
> 
> 
> 
> From: <igor.potjevlesch at gmail.com <mailto:igor.potjevlesch at gmail.com>>
> Subject: Re: [Freeswitch-users] Suddenly lost of fs_cli
> Date: April 17, 2018 at 2:45:12 PM GMT+2
> To: "'FreeSWITCH Users Help'" <freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org>>
> 
> 
> Hello Giovanni,
>  
> Freeswitch seems to process the calls as usual according to the logs and a tcpdump on interfaces.
>  
> Regards,
>  
> Igor.
>  
> De : FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org>> De la part de Giovanni Maruzzelli
> Envoyé : mardi 17 avril 2018 11:19
> À : FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org>>
> Objet : Re: [Freeswitch-users] Suddenly lost of fs_cli
>  
> probably your freeswitch died or is no more correctly working...
>  
> On 17 April 2018 at 10:08, <igor.potjevlesch at gmail.com <mailto:igor.potjevlesch at gmail.com>> wrote:
> Hello,
>  
> Since yesterday, I cannot access to fs_cli anymore after a traffic peak on an instance.
>  
> I got the following error:
> [ERROR] fs_cli.c:1610 main() Error Connecting [Socket Connection Error]
> Usage: /usr/local/freeswitch/bin/fs_cli [-H <host>] [-P <port>] [-p <secret>] [-d <level>] [-x command] [-t <timeout_ms>] [profile]
>  
> Is there something I can do without restarting Freeswitch?
>  
> Regards,
>  
> Igor.
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> http://www.cluecon.com <http://www.cluecon.com/>
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> http://www.freeswitch.org <http://www.freeswitch.org/>
> 
> 
> 
> -- 
> 
> Sincerely,
> 
> Giovanni Maruzzelli
> OpenTelecom.IT
> cell: +39 347 266 56 18
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> http://www.freeswitch.org <http://www.freeswitch.org/>
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From bilaln018 at gmail.com  Tue Apr 17 13:22:02 2018
From: bilaln018 at gmail.com (Bilal Abbasi)
Date: Tue, 17 Apr 2018 18:22:02 +0500
Subject: [Freeswitch-users] Suddenly lost of fs_cli
In-Reply-To: <005701d3d649$ed810a50$c8831ef0$@gmail.com>
References: <003c01d3d623$41492b30$c3db8190$@gmail.com>
 <CALXCt0pJ36dx9i=TGbV8-qof+-eMcR11iHT3JyVEpcOoufpVvQ@mail.gmail.com>
 <005701d3d649$ed810a50$c8831ef0$@gmail.com>
Message-ID: <CADSVerDkPd2emeMEL7YwM4tFdSt5x=KFEbOCY7CrREimFGMguw@mail.gmail.com>

Can you check your acls for your event socket, that happened to me and it
was the acl changes that i made.

On Tue, Apr 17, 2018 at 5:45 PM, <igor.potjevlesch at gmail.com> wrote:

> Hello Giovanni,
>
>
>
> Freeswitch seems to process the calls as usual according to the logs and a
> tcpdump on interfaces.
>
>
>
> Regards,
>
>
>
> Igor.
>
>
>
> *De :* FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> *De
> la part de* Giovanni Maruzzelli
> *Envoyé :* mardi 17 avril 2018 11:19
> *À :* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Objet :* Re: [Freeswitch-users] Suddenly lost of fs_cli
>
>
>
> probably your freeswitch died or is no more correctly working...
>
>
>
> On 17 April 2018 at 10:08, <igor.potjevlesch at gmail.com> wrote:
>
> Hello,
>
>
>
> Since yesterday, I cannot access to fs_cli anymore after a traffic peak on
> an instance.
>
>
>
> I got the following error:
>
> [ERROR] fs_cli.c:1610 main() Error Connecting [Socket Connection Error]
>
> Usage: /usr/local/freeswitch/bin/fs_cli [-H <host>] [-P <port>] [-p
> <secret>] [-d <level>] [-x command] [-t <timeout_ms>] [profile]
>
>
>
> Is there something I can do without restarting Freeswitch?
>
>
>
> Regards,
>
>
>
> Igor.
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> --
>
>
> Sincerely,
>
> Giovanni Maruzzelli
> OpenTelecom.IT
> cell: +39 347 266 56 18
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From igor.potjevlesch at gmail.com  Tue Apr 17 13:22:17 2018
From: igor.potjevlesch at gmail.com (igor.potjevlesch at gmail.com)
Date: Tue, 17 Apr 2018 15:22:17 +0200
Subject: [Freeswitch-users] Suddenly lost of fs_cli
In-Reply-To: <3918B447-A5D3-4C67-9947-4D23015A6CC9@gmail.com>
References: <mailman.260.1523969145.1211.freeswitch-users@lists.freeswitch.org>
 <3918B447-A5D3-4C67-9947-4D23015A6CC9@gmail.com>
Message-ID: <006401d3d64f$1b9f6df0$52de49d0$@gmail.com>

But Freeswitch is already running and process the calls.

If I try this, it will not work? Isn't? I'm not sure that Freeswitch can be launched twice.

 

Regards,

 

Igor.

 

De : Raimundo Pérez Nieves <raimundo.perez.cuba at gmail.com> 
Envoyé : mardi 17 avril 2018 15:06
À : FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>; igor.potjevlesch at gmail.com
Objet : Re: [Freeswitch-users] Suddenly lost of fs_cli

 

Try this:

Go to path /usr/bin and execute ./freeswitch

It starts FS in secure mode, and you can debug the error.

In my case, it was bad close tag in html and I just shutdown FS with that error, so It did not open. Doing this gave me the idea of the problem.

Hope helps





On Apr 17, 2018, at 2:45 PM, freeswitch-users-request at lists.freeswitch.org <mailto:freeswitch-users-request at lists.freeswitch.org>  wrote:

 

Send FreeSWITCH-users mailing list submissions to
                <mailto:freeswitch-users at lists.freeswitch.org> freeswitch-users at lists.freeswitch.org

To subscribe or unsubscribe via the World Wide Web, visit
                <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
or, via email, send a message with subject or body 'help' to
                <mailto:freeswitch-users-request at lists.freeswitch.org> freeswitch-users-request at lists.freeswitch.org

You can reach the person managing the list at
                <mailto:freeswitch-users-owner at lists.freeswitch.org> freeswitch-users-owner at lists.freeswitch.org

When replying, please edit your Subject line so it is more specific
than "Re: Contents of FreeSWITCH-users digest..."
Today's Topics:

  1. Re: Suddenly lost of fs_cli ( <mailto:igor.potjevlesch at gmail.com> igor.potjevlesch at gmail.com)
  2. Re: Suddenly lost of fs_cli ( <mailto:igor.potjevlesch at gmail.com> igor.potjevlesch at gmail.com)



From: < <mailto:igor.potjevlesch at gmail.com> igor.potjevlesch at gmail.com>

Subject: Re: [Freeswitch-users] Suddenly lost of fs_cli

Date: April 17, 2018 at 2:44:28 PM GMT+2

To: "'FreeSWITCH Users Help'" < <mailto:freeswitch-users at lists.freeswitch.org> freeswitch-users at lists.freeswitch.org>





Hello Alex,

 

There is a loop:

[ERROR] fs_cli.c:1610 main() Error Connecting [Socket Connection Error]

[INFO] fs_cli.c:1616 main() Retrying

[ERROR] fs_cli.c:1610 main() Error Connecting [Socket Connection Error]

[INFO] fs_cli.c:1616 main() Retrying

[ERROR] fs_cli.c:1610 main() Error Connecting [Socket Connection Error]

[INFO] fs_cli.c:1616 main() Retrying

[ERROR] fs_cli.c:1610 main() Error Connecting [Socket Connection Error]

[INFO] fs_cli.c:1616 main() Retrying

[ERROR] fs_cli.c:1610 main() Error Connecting [Socket Connection Error]

[INFO] fs_cli.c:1616 main() Retrying

[ERROR] fs_cli.c:1610 main() Error Connecting [Socket Connection Error]

[INFO] fs_cli.c:1616 main() Retrying

[ERROR] fs_cli.c:1610 main() Error Connecting [Socket Connection Error]

[INFO] fs_cli.c:1616 main() Retrying

[…]

 

Regards,

 

Igor

 

De : FreeSWITCH-users < <mailto:freeswitch-users-bounces at lists.freeswitch.org> freeswitch-users-bounces at lists.freeswitch.org> De la part de Alexey Sibyakin
Envoyé : mardi 17 avril 2018 10:46
À : FreeSWITCH Users Help < <mailto:freeswitch-users at lists.freeswitch.org> freeswitch-users at lists.freeswitch.org>
Objet : Re: [Freeswitch-users] Suddenly lost of fs_cli

 

Hello

 

Try to add -rR flags to command

Alex

 

On Tue, Apr 17, 2018 at 5:08 PM, < <mailto:igor.potjevlesch at gmail.com> igor.potjevlesch at gmail.com> wrote:

Hello,

 

Since yesterday, I cannot access to fs_cli anymore after a traffic peak on an instance.

 

I got the following error:

[ERROR] fs_cli.c:1610 main() Error Connecting [Socket Connection Error]

Usage: /usr/local/freeswitch/bin/fs_cli [-H <host>] [-P <port>] [-p <secret>] [-d <level>] [-x command] [-t <timeout_ms>] [profile]

 

Is there something I can do without restarting Freeswitch?

 

Regards,

 

Igor.


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

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

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





 

-- 

Alex Sibyakin | Support Engineer

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

Email:  <mailto:alex at freeswitch.com> alex at freeswitch.com

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

Need commercial support? Contact  <mailto:sales at freeswitch.com> sales at freeswitch.com for details.






From: < <mailto:igor.potjevlesch at gmail.com> igor.potjevlesch at gmail.com>

Subject: Re: [Freeswitch-users] Suddenly lost of fs_cli

Date: April 17, 2018 at 2:45:12 PM GMT+2

To: "'FreeSWITCH Users Help'" < <mailto:freeswitch-users at lists.freeswitch.org> freeswitch-users at lists.freeswitch.org>





Hello Giovanni,

 

Freeswitch seems to process the calls as usual according to the logs and a tcpdump on interfaces.

 

Regards,

 

Igor.

 

De : FreeSWITCH-users < <mailto:freeswitch-users-bounces at lists.freeswitch.org> freeswitch-users-bounces at lists.freeswitch.org> De la part de Giovanni Maruzzelli
Envoyé : mardi 17 avril 2018 11:19
À : FreeSWITCH Users Help < <mailto:freeswitch-users at lists.freeswitch.org> freeswitch-users at lists.freeswitch.org>
Objet : Re: [Freeswitch-users] Suddenly lost of fs_cli

 

probably your freeswitch died or is no more correctly working...

 

On 17 April 2018 at 10:08, < <mailto:igor.potjevlesch at gmail.com> igor.potjevlesch at gmail.com> wrote:

Hello,

 

Since yesterday, I cannot access to fs_cli anymore after a traffic peak on an instance.

 

I got the following error:

[ERROR] fs_cli.c:1610 main() Error Connecting [Socket Connection Error]

Usage: /usr/local/freeswitch/bin/fs_cli [-H <host>] [-P <port>] [-p <secret>] [-d <level>] [-x command] [-t <timeout_ms>] [profile]

 

Is there something I can do without restarting Freeswitch?

 

Regards,

 

Igor.


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

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

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




-- 


Sincerely,

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



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

 

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From alex at freeswitch.com  Tue Apr 17 13:26:44 2018
From: alex at freeswitch.com (Alexey Sibyakin)
Date: Tue, 17 Apr 2018 22:26:44 +0900
Subject: [Freeswitch-users] Suddenly lost of fs_cli
In-Reply-To: <005201d3d649$d320dfa0$79629ee0$@gmail.com>
References: <003c01d3d623$41492b30$c3db8190$@gmail.com>
 <CAGJLwFgiZRdUi_SJBjBb6KCXaNwK7GDYsckG0xow+YTN_S7mZg@mail.gmail.com>
 <005201d3d649$d320dfa0$79629ee0$@gmail.com>
Message-ID: <CAGJLwFi=cDMbcTYhdO06MnyZZU-1oqR3QrhN8aRamNtt8KXKTA@mail.gmail.com>

Can you see ESL socket in "ss -putan | column -t" or netstat -putan ? Any
errors in the log?

Alex

On Tue, Apr 17, 2018 at 9:44 PM, <igor.potjevlesch at gmail.com> wrote:

> Hello Alex,
>
>
>
> There is a loop:
>
> [ERROR] fs_cli.c:1610 main() Error Connecting [Socket Connection Error]
>
> [INFO] fs_cli.c:1616 main() Retrying
>
> [ERROR] fs_cli.c:1610 main() Error Connecting [Socket Connection Error]
>
> [INFO] fs_cli.c:1616 main() Retrying
>
> [ERROR] fs_cli.c:1610 main() Error Connecting [Socket Connection Error]
>
> [INFO] fs_cli.c:1616 main() Retrying
>
> [ERROR] fs_cli.c:1610 main() Error Connecting [Socket Connection Error]
>
> [INFO] fs_cli.c:1616 main() Retrying
>
> [ERROR] fs_cli.c:1610 main() Error Connecting [Socket Connection Error]
>
> [INFO] fs_cli.c:1616 main() Retrying
>
> [ERROR] fs_cli.c:1610 main() Error Connecting [Socket Connection Error]
>
> [INFO] fs_cli.c:1616 main() Retrying
>
> [ERROR] fs_cli.c:1610 main() Error Connecting [Socket Connection Error]
>
> [INFO] fs_cli.c:1616 main() Retrying
>
> […]
>
>
>
> Regards,
>
>
>
> Igor
>
>
>
> *De :* FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> *De
> la part de* Alexey Sibyakin
> *Envoyé :* mardi 17 avril 2018 10:46
> *À :* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Objet :* Re: [Freeswitch-users] Suddenly lost of fs_cli
>
>
>
> Hello
>
>
>
> Try to add -rR flags to command
>
> Alex
>
>
>
> On Tue, Apr 17, 2018 at 5:08 PM, <igor.potjevlesch at gmail.com> wrote:
>
> Hello,
>
>
>
> Since yesterday, I cannot access to fs_cli anymore after a traffic peak on
> an instance.
>
>
>
> I got the following error:
>
> [ERROR] fs_cli.c:1610 main() Error Connecting [Socket Connection Error]
>
> Usage: /usr/local/freeswitch/bin/fs_cli [-H <host>] [-P <port>] [-p
> <secret>] [-d <level>] [-x command] [-t <timeout_ms>] [profile]
>
>
>
> Is there something I can do without restarting Freeswitch?
>
>
>
> Regards,
>
>
>
> Igor.
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
>
> --
>
> Alex Sibyakin | Support Engineer
>
> FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045
> <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g>
>
> Email: alex at freeswitch.com
>
> Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/>
>
> Need commercial support? Contact sales at freeswitch.com for details.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 Engineer
FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045
<https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g>
Email: alex at freeswitch.com
Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/>
Need commercial support? Contact sales at freeswitch.com for details.
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From igor.potjevlesch at gmail.com  Tue Apr 17 14:47:30 2018
From: igor.potjevlesch at gmail.com (igor.potjevlesch at gmail.com)
Date: Tue, 17 Apr 2018 16:47:30 +0200
Subject: [Freeswitch-users] Suddenly lost of fs_cli
In-Reply-To: <CAGJLwFi=cDMbcTYhdO06MnyZZU-1oqR3QrhN8aRamNtt8KXKTA@mail.gmail.com>
References: <003c01d3d623$41492b30$c3db8190$@gmail.com>
 <CAGJLwFgiZRdUi_SJBjBb6KCXaNwK7GDYsckG0xow+YTN_S7mZg@mail.gmail.com>
 <005201d3d649$d320dfa0$79629ee0$@gmail.com>
 <CAGJLwFi=cDMbcTYhdO06MnyZZU-1oqR3QrhN8aRamNtt8KXKTA@mail.gmail.com>
Message-ID: <008401d3d65b$03649dd0$0a2dd970$@gmail.com>

There is no more log written now. I could see in the last one some errors like this:

2018-04-17 14:41:09.219168 [CRIT] mod_cdr_pg_csv.c:268 Connection to database failed: could not create socket: Too many open files

 

But if I take a network dump, the traffic is processed.

 

Regarding the ESL socket, I'm not sure, do you see it?

Proto Recv-Q Send-Q Local Address               Foreign Address             State       PID/Program name

tcp        0      0 192.168.20.30:5066            0.0.0.0:*                   LISTEN      27314/freeswitch

tcp        0      0 192.168.20.30:7443            0.0.0.0:*                   LISTEN      27314/freeswitch

tcp        0      0 192.168.20.18:5060            0.0.0.0:*                   LISTEN      27314/freeswitch

tcp        0      0 192.168.20.17:5060            0.0.0.0:*                   LISTEN      27314/freeswitch

tcp        0      0 192.168.20.131:5060           0.0.0.0:*                   LISTEN      27314/freeswitch

tcp        0      0 192.168.20.19:5060            0.0.0.0:*                   LISTEN      27314/freeswitch

tcp        0      0 192.168.20.31:5060            0.0.0.0:*                   LISTEN      27314/freeswitch

tcp        0      0 192.168.20.130:5060           0.0.0.0:*                   LISTEN      27314/freeswitch

tcp        0      0 192.168.20.16:5060            0.0.0.0:*                   LISTEN      27314/freeswitch

tcp        0      0 192.168.20.30:5060            0.0.0.0:*                   LISTEN      27314/freeswitch

udp        0      0 192.168.20.30:2408            192.168.20.1:5351             ESTABLISHED 27314/freeswitch

udp        0      0 192.168.20.18:5060            0.0.0.0:*                               27314/freeswitch

udp        0      0 192.168.20.17:5060            0.0.0.0:*                               27314/freeswitch

udp        0      0 192.168.20.131:5060           0.0.0.0:*                               27314/freeswitch

udp        0      0 192.168.20.19:5060            0.0.0.0:*                               27314/freeswitch

udp        0      0 192.168.20.31:5060            0.0.0.0:*                               27314/freeswitch

udp        0      0 192.168.20.130:5060           0.0.0.0:*                               27314/freeswitch

udp        0      0 192.168.20.16:5060            0.0.0.0:*                               27314/freeswitch

udp        0      0 192.168.20.30:5060            0.0.0.0:*                               27314/freeswitch

 

Regards,

 

Igor.

 

De : FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> De la part de Alexey Sibyakin
Envoyé : mardi 17 avril 2018 15:27
À : FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Objet : Re: [Freeswitch-users] Suddenly lost of fs_cli

 

Can you see ESL socket in "ss -putan | column -t" or netstat -putan ? Any errors in the log?

 

Alex

 

On Tue, Apr 17, 2018 at 9:44 PM, <igor.potjevlesch at gmail.com <mailto:igor.potjevlesch at gmail.com> > wrote:

Hello Alex,

 

There is a loop:

[ERROR] fs_cli.c:1610 main() Error Connecting [Socket Connection Error]

[INFO] fs_cli.c:1616 main() Retrying

[ERROR] fs_cli.c:1610 main() Error Connecting [Socket Connection Error]

[INFO] fs_cli.c:1616 main() Retrying

[ERROR] fs_cli.c:1610 main() Error Connecting [Socket Connection Error]

[INFO] fs_cli.c:1616 main() Retrying

[ERROR] fs_cli.c:1610 main() Error Connecting [Socket Connection Error]

[INFO] fs_cli.c:1616 main() Retrying

[ERROR] fs_cli.c:1610 main() Error Connecting [Socket Connection Error]

[INFO] fs_cli.c:1616 main() Retrying

[ERROR] fs_cli.c:1610 main() Error Connecting [Socket Connection Error]

[INFO] fs_cli.c:1616 main() Retrying

[ERROR] fs_cli.c:1610 main() Error Connecting [Socket Connection Error]

[INFO] fs_cli.c:1616 main() Retrying

[…]

 

Regards,

 

Igor

 

De : FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org> > De la part de Alexey Sibyakin
Envoyé : mardi 17 avril 2018 10:46
À : FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org> >
Objet : Re: [Freeswitch-users] Suddenly lost of fs_cli

 

Hello

 

Try to add -rR flags to command

Alex

 

On Tue, Apr 17, 2018 at 5:08 PM, <igor.potjevlesch at gmail.com <mailto:igor.potjevlesch at gmail.com> > wrote:

Hello,

 

Since yesterday, I cannot access to fs_cli anymore after a traffic peak on an instance.

 

I got the following error:

[ERROR] fs_cli.c:1610 main() Error Connecting [Socket Connection Error]

Usage: /usr/local/freeswitch/bin/fs_cli [-H <host>] [-P <port>] [-p <secret>] [-d <level>] [-x command] [-t <timeout_ms>] [profile]

 

Is there something I can do without restarting Freeswitch?

 

Regards,

 

Igor.


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

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

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





 

-- 

Alex Sibyakin | Support Engineer

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

Email:  <mailto:alex at freeswitch.com> alex at freeswitch.com

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

Need commercial support? Contact  <mailto:sales at freeswitch.com> sales at freeswitch.com for details.


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

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

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





 

-- 

Alex Sibyakin | Support Engineer

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

Email:  <mailto:alex at freeswitch.com> alex at freeswitch.com

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

Need commercial support? Contact  <mailto:sales at freeswitch.com> sales at freeswitch.com for details.

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From mike at jerris.com  Tue Apr 17 14:56:53 2018
From: mike at jerris.com (Michael Jerris)
Date: Tue, 17 Apr 2018 10:56:53 -0400
Subject: [Freeswitch-users] 100rel support
In-Reply-To: <CAFiqYunfhx0-y+tTvjJtu-MzAeoy1B_T-4iewPiY522uy6e-UA@mail.gmail.com>
References: <CAPY30xYiUaeXafjZmshEN2bNvgkhRK4h43_zsrW938x-MTm+wg@mail.gmail.com>
 <CAFiqYunfhx0-y+tTvjJtu-MzAeoy1B_T-4iewPiY522uy6e-UA@mail.gmail.com>
Message-ID: <C1DBD690-4797-4E38-8DA4-A47D1BD6C425@jerris.com>

The known crash is specifically with proxy forking with 100rel.  If you never hit that its probably fine, but you have zero control over if that will happen, so enabling it opens you to a remote denial of service issue.


> On Apr 17, 2018, at 5:41 AM, Steven Ayre <steveayre at gmail.com> wrote:
> 
> I've been wondering this for a few years too. I'm guessing that comment has scared everyone away from enabling it to test it out.
> 
> At a guess if there's any issue it's within the Sofia-SIP stack and therefore trickier to debug/patch than in mod_sofia.
> 
> You could enable it and if there are any problems report them on Jira so they get looked into.
> 
> I believe 100rel might be required for IMS implementations.
> 
> On 17 April 2018 at 06:08, David Witham <david.witham at netsip.com.au <mailto:david.witham at netsip.com.au>> wrote:
> Hi all,
> 
> We have had an upstream provider request that we enable 100rel support. I note in the wiki it says:
> 
> enable-100rel
> This enable support for 100rel (100% reliability - PRACK message as defined in RFC3262 <http://tools.ietf.org/html/rfc3262>) This fixes a problem with SIP where provisional messages like "180 Ringing" are not ACK'd and therefore could be dropped over a poor connection without retransmission. *2009-07-08:* Enabling this may cause FreeSWITCH to crash, see FSCORE-392 <http://jira.freeswitch.org/browse/FSCORE-392>.
> <param name="enable-100rel" value="true"/>
> 
> 
> FSCORE-392 doesn't say if the crashing related to enabling 100rel was ever fixed. Can anyone confirm whether it is now safe to enable 100rel support?
> 
> regards,
> David
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> http://www.cluecon.com <http://www.cluecon.com/>
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> http://www.freeswitch.org <http://www.freeswitch.org/>
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From grcamauer at gmail.com  Tue Apr 17 16:12:12 2018
From: grcamauer at gmail.com (Guillermo Ruiz Camauer)
Date: Tue, 17 Apr 2018 13:12:12 -0300
Subject: [Freeswitch-users] Suddenly lost of fs_cli
In-Reply-To: <008401d3d65b$03649dd0$0a2dd970$@gmail.com>
References: <003c01d3d623$41492b30$c3db8190$@gmail.com>
 <CAGJLwFgiZRdUi_SJBjBb6KCXaNwK7GDYsckG0xow+YTN_S7mZg@mail.gmail.com>
 <005201d3d649$d320dfa0$79629ee0$@gmail.com>
 <CAGJLwFi=cDMbcTYhdO06MnyZZU-1oqR3QrhN8aRamNtt8KXKTA@mail.gmail.com>
 <008401d3d65b$03649dd0$0a2dd970$@gmail.com>
Message-ID: <CAPNTnnp1mHpxLshGVrzTsyn3ZQGxonKehVcyYhEgb8ppW3igpQ@mail.gmail.com>

If it is not writing to tthe log or to the CDR, I think its high time you
stopped/kill that instance and start a new one...

Guillermo

On Tue, Apr 17, 2018 at 11:47 AM, <igor.potjevlesch at gmail.com> wrote:

> There is no more log written now. I could see in the last one some errors
> like this:
>
> 2018-04-17 14:41:09.219168 [CRIT] mod_cdr_pg_csv.c:268 Connection to
> database failed: could not create socket: Too many open files
>
>
>
> But if I take a network dump, the traffic is processed.
>
>
>
> Regarding the ESL socket, I'm not sure, do you see it?
>
> Proto Recv-Q Send-Q Local Address               Foreign
> Address             State       PID/Program name
>
> tcp        0      0 192.168.20.30:5066            0.0.0.0:*
> LISTEN      27314/freeswitch
>
> tcp        0      0 192.168.20.30:7443            0.0.0.0:*
> LISTEN      27314/freeswitch
>
> tcp        0      0 192.168.20.18:5060            0.0.0.0:*
> LISTEN      27314/freeswitch
>
> tcp        0      0 192.168.20.17:5060            0.0.0.0:*
> LISTEN      27314/freeswitch
>
> tcp        0      0 192.168.20.131:5060           0.0.0.0:*
> LISTEN      27314/freeswitch
>
> tcp        0      0 192.168.20.19:5060            0.0.0.0:*
> LISTEN      27314/freeswitch
>
> tcp        0      0 192.168.20.31:5060            0.0.0.0:*
> LISTEN      27314/freeswitch
>
> tcp        0      0 192.168.20.130:5060           0.0.0.0:*
> LISTEN      27314/freeswitch
>
> tcp        0      0 192.168.20.16:5060            0.0.0.0:*
> LISTEN      27314/freeswitch
>
> tcp        0      0 192.168.20.30:5060            0.0.0.0:*
> LISTEN      27314/freeswitch
>
> udp        0      0 192.168.20.30:2408            192.168.20.1:5351
> ESTABLISHED 27314/freeswitch
>
> udp        0      0 192.168.20.18:5060            0.0.0.0:
> *                               27314/freeswitch
>
> udp        0      0 192.168.20.17:5060            0.0.0.0:
> *                               27314/freeswitch
>
> udp        0      0 192.168.20.131:5060           0.0.0.0:
> *                               27314/freeswitch
>
> udp        0      0 192.168.20.19:5060            0.0.0.0:*
>               27314/freeswitch
>
> udp        0      0 192.168.20.31:5060            0.0.0.0:
> *                               27314/freeswitch
>
> udp        0      0 192.168.20.130:5060           0.0.0.0:
> *                               27314/freeswitch
>
> udp        0      0 192.168.20.16:5060            0.0.0.0:
> *                               27314/freeswitch
>
> udp        0      0 192.168.20.30:5060            0.0.0.0:
> *                               27314/freeswitch
>
>
>
> Regards,
>
>
>
> Igor.
>
>
>
> *De :* FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> *De
> la part de* Alexey Sibyakin
> *Envoyé :* mardi 17 avril 2018 15:27
>
> *À :* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Objet :* Re: [Freeswitch-users] Suddenly lost of fs_cli
>
>
>
> Can you see ESL socket in "ss -putan | column -t" or netstat -putan ? Any
> errors in the log?
>
>
>
> Alex
>
>
>
> On Tue, Apr 17, 2018 at 9:44 PM, <igor.potjevlesch at gmail.com> wrote:
>
> Hello Alex,
>
>
>
> There is a loop:
>
> [ERROR] fs_cli.c:1610 main() Error Connecting [Socket Connection Error]
>
> [INFO] fs_cli.c:1616 main() Retrying
>
> [ERROR] fs_cli.c:1610 main() Error Connecting [Socket Connection Error]
>
> [INFO] fs_cli.c:1616 main() Retrying
>
> [ERROR] fs_cli.c:1610 main() Error Connecting [Socket Connection Error]
>
> [INFO] fs_cli.c:1616 main() Retrying
>
> [ERROR] fs_cli.c:1610 main() Error Connecting [Socket Connection Error]
>
> [INFO] fs_cli.c:1616 main() Retrying
>
> [ERROR] fs_cli.c:1610 main() Error Connecting [Socket Connection Error]
>
> [INFO] fs_cli.c:1616 main() Retrying
>
> [ERROR] fs_cli.c:1610 main() Error Connecting [Socket Connection Error]
>
> [INFO] fs_cli.c:1616 main() Retrying
>
> [ERROR] fs_cli.c:1610 main() Error Connecting [Socket Connection Error]
>
> [INFO] fs_cli.c:1616 main() Retrying
>
> […]
>
>
>
> Regards,
>
>
>
> Igor
>
>
>
> *De :* FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> *De
> la part de* Alexey Sibyakin
> *Envoyé :* mardi 17 avril 2018 10:46
> *À :* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Objet :* Re: [Freeswitch-users] Suddenly lost of fs_cli
>
>
>
> Hello
>
>
>
> Try to add -rR flags to command
>
> Alex
>
>
>
> On Tue, Apr 17, 2018 at 5:08 PM, <igor.potjevlesch at gmail.com> wrote:
>
> Hello,
>
>
>
> Since yesterday, I cannot access to fs_cli anymore after a traffic peak on
> an instance.
>
>
>
> I got the following error:
>
> [ERROR] fs_cli.c:1610 main() Error Connecting [Socket Connection Error]
>
> Usage: /usr/local/freeswitch/bin/fs_cli [-H <host>] [-P <port>] [-p
> <secret>] [-d <level>] [-x command] [-t <timeout_ms>] [profile]
>
>
>
> Is there something I can do without restarting Freeswitch?
>
>
>
> Regards,
>
>
>
> Igor.
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
>
> --
>
> Alex Sibyakin | Support Engineer
>
> FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045
> <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g>
>
> Email: alex at freeswitch.com
>
> Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/>
>
> Need commercial support? Contact sales at freeswitch.com for details.
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 Engineer
>
> FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045
> <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g>
>
> Email: alex at freeswitch.com
>
> Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/>
>
> Need commercial support? Contact sales at freeswitch.com for details.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Guillermo Ruiz Camauer
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From nneul at mst.edu  Tue Apr 17 17:12:16 2018
From: nneul at mst.edu (Nathan Neulinger)
Date: Tue, 17 Apr 2018 12:12:16 -0500
Subject: [Freeswitch-users] Suddenly lost of fs_cli
In-Reply-To: <CAPNTnnp1mHpxLshGVrzTsyn3ZQGxonKehVcyYhEgb8ppW3igpQ@mail.gmail.com>
References: <003c01d3d623$41492b30$c3db8190$@gmail.com>
 <CAGJLwFgiZRdUi_SJBjBb6KCXaNwK7GDYsckG0xow+YTN_S7mZg@mail.gmail.com>
 <005201d3d649$d320dfa0$79629ee0$@gmail.com>
 <CAGJLwFi=cDMbcTYhdO06MnyZZU-1oqR3QrhN8aRamNtt8KXKTA@mail.gmail.com>
 <008401d3d65b$03649dd0$0a2dd970$@gmail.com>
 <CAPNTnnp1mHpxLshGVrzTsyn3ZQGxonKehVcyYhEgb8ppW3igpQ@mail.gmail.com>
Message-ID: <53d3149a-3257-3939-c4ea-66a9f1870a3a@mst.edu>

Would also be a really good time to check your /etc/security/limits.conf or limits.d/ settings for freeswitch for file 
descriptor limit settings.

-- Nathan

On 4/17/18 11:12 AM, Guillermo Ruiz Camauer wrote:
> If it is not writing to tthe log or to the CDR, I think its high time you stopped/kill that instance and start a new 
> one...
>
> Guillermo
>
> On Tue, Apr 17, 2018 at 11:47 AM, <igor.potjevlesch at gmail.com <mailto:igor.potjevlesch at gmail.com>> wrote:
>
>     There is no more log written now. I could see in the last one some errors like this:
>
>     2018-04-17 14:41:09.219168 [CRIT] mod_cdr_pg_csv.c:268 Connection to database failed: could not create socket: Too
>     many open files
>
>     But if I take a network dump, the traffic is processed.
>
>     Regarding the ESL socket, I'm not sure, do you see it?
>
>     Proto Recv-Q Send-Q Local Address               Foreign Address             State       PID/Program name
>
>     tcp 0      0 192.168.20.30:5066 <http://192.168.20.30:5066>           0.0.0.0:* LISTEN      27314/freeswitch
>
>     tcp 0      0 192.168.20.30:7443 <http://192.168.20.30:7443> 0.0.0.0:*                   LISTEN 27314/freeswitch
>
>     tcp 0      0 192.168.20.18:5060 <http://192.168.20.18:5060> 0.0.0.0:*                   LISTEN 27314/freeswitch
>
>     tcp 0      0 192.168.20.17:5060 <http://192.168.20.17:5060> 0.0.0.0:*                   LISTEN 27314/freeswitch
>
>     tcp 0      0 192.168.20.131:5060 <http://192.168.20.131:5060> 0.0.0.0:*                   LISTEN 27314/freeswitch
>
>     tcp        0  0 192.168.20.19:5060 <http://192.168.20.19:5060> 0.0.0.0:*                   LISTEN 27314/freeswitch
>
>     tcp 0      0 192.168.20.31:5060 <http://192.168.20.31:5060> 0.0.0.0:*                   LISTEN 27314/freeswitch
>
>     tcp 0      0 192.168.20.130:5060 <http://192.168.20.130:5060> 0.0.0.0:*                   LISTEN 27314/freeswitch
>
>     tcp 0      0 192.168.20.16:5060 <http://192.168.20.16:5060> 0.0.0.0:*                   LISTEN 27314/freeswitch
>
>     tcp 0      0 192.168.20.30:5060 <http://192.168.20.30:5060> 0.0.0.0:*                   LISTEN 27314/freeswitch
>
>     udp 0      0 192.168.20.30:2408 <http://192.168.20.30:2408> 192.168.20.1:5351 <http://192.168.20.1:5351>
>     ESTABLISHED 27314/freeswitch
>
>     udp 0      0 192.168.20.18:5060 <http://192.168.20.18:5060> 0.0.0.0:* 27314/freeswitch
>
>     udp 0      0 192.168.20.17:5060 <http://192.168.20.17:5060> 0.0.0.0:* 27314/freeswitch
>
>     udp 0      0 192.168.20.131:5060 <http://192.168.20.131:5060> 0.0.0.0:* 27314/freeswitch
>
>     udp 0      0 192.168.20.19:5060 <http://192.168.20.19:5060> 0.0.0.0:*               27314/freeswitch
>
>     udp 0      0 192.168.20.31:5060 <http://192.168.20.31:5060> 0.0.0.0:* 27314/freeswitch
>
>     udp 0      0 192.168.20.130:5060 <http://192.168.20.130:5060> 0.0.0.0:* 27314/freeswitch
>
>     udp 0      0 192.168.20.16:5060 <http://192.168.20.16:5060> 0.0.0.0:* 27314/freeswitch
>
>     udp 0      0 192.168.20.30:5060 <http://192.168.20.30:5060> 0.0.0.0:* 27314/freeswitch
>
>     Regards,
>
>     Igor.
>
>     *De :* FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org
>     <mailto:freeswitch-users-bounces at lists.freeswitch.org>> *De la part de* Alexey Sibyakin
>     *Envoyé :* mardi 17 avril 2018 15:27
>
>
>     *À :* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org>>
>     *Objet :* Re: [Freeswitch-users] Suddenly lost of fs_cli
>
>     Can you see ESL socket in "ss -putan | column -t" or netstat -putan ? Any errors in the log?
>
>     Alex
>
>     On Tue, Apr 17, 2018 at 9:44 PM, <igor.potjevlesch at gmail.com <mailto:igor.potjevlesch at gmail.com>> wrote:
>
>         Hello Alex,
>
>         There is a loop:
>
>         [ERROR] fs_cli.c:1610 main() Error Connecting [Socket Connection Error]
>
>         [INFO] fs_cli.c:1616 main() Retrying
>
>         [ERROR] fs_cli.c:1610 main() Error Connecting [Socket Connection Error]
>
>         [INFO] fs_cli.c:1616 main() Retrying
>
>         [ERROR] fs_cli.c:1610 main() Error Connecting [Socket Connection Error]
>
>         [INFO] fs_cli.c:1616 main() Retrying
>
>         [ERROR] fs_cli.c:1610 main() Error Connecting [Socket Connection Error]
>
>         [INFO] fs_cli.c:1616 main() Retrying
>
>         [ERROR] fs_cli.c:1610 main() Error Connecting [Socket Connection Error]
>
>         [INFO] fs_cli.c:1616 main() Retrying
>
>         [ERROR] fs_cli.c:1610 main() Error Connecting [Socket Connection Error]
>
>         [INFO] fs_cli.c:1616 main() Retrying
>
>         [ERROR] fs_cli.c:1610 main() Error Connecting [Socket Connection Error]
>
>         [INFO] fs_cli.c:1616 main() Retrying
>
>         […]
>
>         Regards,
>
>         Igor
>
>         *De :* FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org
>         <mailto:freeswitch-users-bounces at lists.freeswitch.org>> *De la part de* Alexey Sibyakin
>         *Envoyé :* mardi 17 avril 2018 10:46
>         *À :* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org>>
>         *Objet :* Re: [Freeswitch-users] Suddenly lost of fs_cli
>
>         Hello
>
>         Try to add -rR flags to command
>
>         Alex
>
>         On Tue, Apr 17, 2018 at 5:08 PM, <igor.potjevlesch at gmail.com <mailto:igor.potjevlesch at gmail.com>> wrote:
>
>             Hello,
>
>             Since yesterday, I cannot access to fs_cli anymore after a traffic peak on an instance.
>
>             I got the following error:
>
>             [ERROR] fs_cli.c:1610 main() Error Connecting [Socket Connection Error]
>
>             Usage: /usr/local/freeswitch/bin/fs_cli [-H <host>] [-P <port>] [-p <secret>] [-d <level>] [-x command]
>             [-t <timeout_ms>] [profile]
>
>             Is there something I can do without restarting Freeswitch?
>
>             Regards,
>
>             Igor.
>
>
>             _________________________________________________________________________
>             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
>
>
>
>         -- 
>
>         Alex Sibyakin | Support Engineer
>
>         FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045
>         <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g>
>
>         Email: alex at freeswitch.com <mailto:alex at freeswitch.com>
>
>         Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/>
>
>         Need commercial support? Contactsales at freeswitch.com <mailto:sales at freeswitch.com>for details.
>
>
>         _________________________________________________________________________
>         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
>
>
>
>     -- 
>
>     Alex Sibyakin | Support Engineer
>
>     FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045
>     <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g>
>
>     Email: alex at freeswitch.com <mailto:alex at freeswitch.com>
>
>     Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/>
>
>     Need commercial support? Contactsales at freeswitch.com <mailto:sales at freeswitch.com>for details.
>
>
>     _________________________________________________________________________
>     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
>
>
>
>
> -- 
> Guillermo Ruiz Camauer
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
------------------------------------------------------------
Nathan Neulinger                       nneul at mst.edu
Missouri S&T Information Technology    (573) 612-1412
System Administrator - Architect

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From nneul at mst.edu  Tue Apr 17 17:24:07 2018
From: nneul at mst.edu (Nathan Neulinger)
Date: Tue, 17 Apr 2018 12:24:07 -0500
Subject: [Freeswitch-users] Suddenly lost of fs_cli
In-Reply-To: <53d3149a-3257-3939-c4ea-66a9f1870a3a@mst.edu>
References: <003c01d3d623$41492b30$c3db8190$@gmail.com>
 <CAGJLwFgiZRdUi_SJBjBb6KCXaNwK7GDYsckG0xow+YTN_S7mZg@mail.gmail.com>
 <005201d3d649$d320dfa0$79629ee0$@gmail.com>
 <CAGJLwFi=cDMbcTYhdO06MnyZZU-1oqR3QrhN8aRamNtt8KXKTA@mail.gmail.com>
 <008401d3d65b$03649dd0$0a2dd970$@gmail.com>
 <CAPNTnnp1mHpxLshGVrzTsyn3ZQGxonKehVcyYhEgb8ppW3igpQ@mail.gmail.com>
 <53d3149a-3257-3939-c4ea-66a9f1870a3a@mst.edu>
Message-ID: <1ed3ca1e-f394-0c23-343d-406bf4442755@mst.edu>

Example on my box:

freeswitch soft nproc 50000
freeswitch hard nproc 50000
freeswitch soft nofile 100000
freeswitch hard nofile 100000

Which is a bit overkill -- right now my freeswitch process has 1638 open files, and 1216 active threads.

-- Nathan


On 4/17/18 12:12 PM, Nathan Neulinger wrote:
>
> Would also be a really good time to check your /etc/security/limits.conf or limits.d/ settings for freeswitch for file 
> descriptor limit settings.
>
> -- Nathan
>
> On 4/17/18 11:12 AM, Guillermo Ruiz Camauer wrote:
>> If it is not writing to tthe log or to the CDR, I think its high time you stopped/kill that instance and start a new 
>> one...
>>
>> Guillermo
>>
>> On Tue, Apr 17, 2018 at 11:47 AM, <igor.potjevlesch at gmail.com <mailto:igor.potjevlesch at gmail.com>> wrote:
>>
>>     There is no more log written now. I could see in the last one some errors like this:
>>
>>     2018-04-17 14:41:09.219168 [CRIT] mod_cdr_pg_csv.c:268 Connection to database failed: could not create socket:
>>     Too many open files
>>
>>     But if I take a network dump, the traffic is processed.
>>
>>     Regarding the ESL socket, I'm not sure, do you see it?
>>
>>     Proto Recv-Q Send-Q Local Address               Foreign Address             State       PID/Program name
>>
>>     tcp 0      0 192.168.20.30:5066 <http://192.168.20.30:5066>           0.0.0.0:* LISTEN      27314/freeswitch
>>
>>     tcp 0      0 192.168.20.30:7443 <http://192.168.20.30:7443> 0.0.0.0:*                   LISTEN 27314/freeswitch
>>
>>     tcp 0      0 192.168.20.18:5060 <http://192.168.20.18:5060> 0.0.0.0:*                   LISTEN 27314/freeswitch
>>
>>     tcp 0      0 192.168.20.17:5060 <http://192.168.20.17:5060> 0.0.0.0:*                   LISTEN 27314/freeswitch
>>
>>     tcp 0      0 192.168.20.131:5060 <http://192.168.20.131:5060> 0.0.0.0:*                   LISTEN 27314/freeswitch
>>
>>     tcp 0      0 192.168.20.19:5060 <http://192.168.20.19:5060> 0.0.0.0:*                   LISTEN 27314/freeswitch
>>
>>     tcp 0      0 192.168.20.31:5060 <http://192.168.20.31:5060> 0.0.0.0:*                   LISTEN 27314/freeswitch
>>
>>     tcp 0      0 192.168.20.130:5060 <http://192.168.20.130:5060> 0.0.0.0:*                   LISTEN 27314/freeswitch
>>
>>     tcp 0      0 192.168.20.16:5060 <http://192.168.20.16:5060> 0.0.0.0:*                   LISTEN 27314/freeswitch
>>
>>     tcp 0      0 192.168.20.30:5060 <http://192.168.20.30:5060> 0.0.0.0:*                   LISTEN 27314/freeswitch
>>
>>     udp 0      0 192.168.20.30:2408 <http://192.168.20.30:2408> 192.168.20.1:5351 <http://192.168.20.1:5351>
>>     ESTABLISHED 27314/freeswitch
>>
>>     udp 0      0 192.168.20.18:5060 <http://192.168.20.18:5060> 0.0.0.0:* 27314/freeswitch
>>
>>     udp 0      0 192.168.20.17:5060 <http://192.168.20.17:5060> 0.0.0.0:* 27314/freeswitch
>>
>>     udp 0      0 192.168.20.131:5060 <http://192.168.20.131:5060> 0.0.0.0:* 27314/freeswitch
>>
>>     udp 0      0 192.168.20.19:5060 <http://192.168.20.19:5060> 0.0.0.0:*               27314/freeswitch
>>
>>     udp 0      0 192.168.20.31:5060 <http://192.168.20.31:5060> 0.0.0.0:* 27314/freeswitch
>>
>>     udp 0      0 192.168.20.130:5060 <http://192.168.20.130:5060> 0.0.0.0:* 27314/freeswitch
>>
>>     udp 0      0 192.168.20.16:5060 <http://192.168.20.16:5060> 0.0.0.0:* 27314/freeswitch
>>
>>     udp 0      0 192.168.20.30:5060 <http://192.168.20.30:5060> 0.0.0.0:* 27314/freeswitch
>>
>>     Regards,
>>
>>     Igor.
>>
>>     *De :* FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org
>>     <mailto:freeswitch-users-bounces at lists.freeswitch.org>> *De la part de* Alexey Sibyakin
>>     *Envoyé :* mardi 17 avril 2018 15:27
>>
>>
>>     *À :* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org>>
>>     *Objet :* Re: [Freeswitch-users] Suddenly lost of fs_cli
>>
>>     Can you see ESL socket in "ss -putan | column -t" or netstat -putan ? Any errors in the log?
>>
>>     Alex
>>
>>     On Tue, Apr 17, 2018 at 9:44 PM, <igor.potjevlesch at gmail.com <mailto:igor.potjevlesch at gmail.com>> wrote:
>>
>>         Hello Alex,
>>
>>         There is a loop:
>>
>>         [ERROR] fs_cli.c:1610 main() Error Connecting [Socket Connection Error]
>>
>>         [INFO] fs_cli.c:1616 main() Retrying
>>
>>         [ERROR] fs_cli.c:1610 main() Error Connecting [Socket Connection Error]
>>
>>         [INFO] fs_cli.c:1616 main() Retrying
>>
>>         [ERROR] fs_cli.c:1610 main() Error Connecting [Socket Connection Error]
>>
>>         [INFO] fs_cli.c:1616 main() Retrying
>>
>>         [ERROR] fs_cli.c:1610 main() Error Connecting [Socket Connection Error]
>>
>>         [INFO] fs_cli.c:1616 main() Retrying
>>
>>         [ERROR] fs_cli.c:1610 main() Error Connecting [Socket Connection Error]
>>
>>         [INFO] fs_cli.c:1616 main() Retrying
>>
>>         [ERROR] fs_cli.c:1610 main() Error Connecting [Socket Connection Error]
>>
>>         [INFO] fs_cli.c:1616 main() Retrying
>>
>>         [ERROR] fs_cli.c:1610 main() Error Connecting [Socket Connection Error]
>>
>>         [INFO] fs_cli.c:1616 main() Retrying
>>
>>         […]
>>
>>         Regards,
>>
>>         Igor
>>
>>         *De :* FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org
>>         <mailto:freeswitch-users-bounces at lists.freeswitch.org>> *De la part de* Alexey Sibyakin
>>         *Envoyé :* mardi 17 avril 2018 10:46
>>         *À :* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org
>>         <mailto:freeswitch-users at lists.freeswitch.org>>
>>         *Objet :* Re: [Freeswitch-users] Suddenly lost of fs_cli
>>
>>         Hello
>>
>>         Try to add -rR flags to command
>>
>>         Alex
>>
>>         On Tue, Apr 17, 2018 at 5:08 PM, <igor.potjevlesch at gmail.com <mailto:igor.potjevlesch at gmail.com>> wrote:
>>
>>             Hello,
>>
>>             Since yesterday, I cannot access to fs_cli anymore after a traffic peak on an instance.
>>
>>             I got the following error:
>>
>>             [ERROR] fs_cli.c:1610 main() Error Connecting [Socket Connection Error]
>>
>>             Usage: /usr/local/freeswitch/bin/fs_cli [-H <host>] [-P <port>] [-p <secret>] [-d <level>] [-x command]
>>             [-t <timeout_ms>] [profile]
>>
>>             Is there something I can do without restarting Freeswitch?
>>
>>             Regards,
>>
>>             Igor.
>>
>>
>>             _________________________________________________________________________
>>             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
>>
>>
>>
>>         -- 
>>
>>         Alex Sibyakin | Support Engineer
>>
>>         FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045
>>         <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g>
>>
>>         Email: alex at freeswitch.com <mailto:alex at freeswitch.com>
>>
>>         Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/>
>>
>>         Need commercial support? Contactsales at freeswitch.com <mailto:sales at freeswitch.com>for details.
>>
>>
>>         _________________________________________________________________________
>>         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
>>
>>
>>
>>     -- 
>>
>>     Alex Sibyakin | Support Engineer
>>
>>     FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045
>>     <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g>
>>
>>     Email: alex at freeswitch.com <mailto:alex at freeswitch.com>
>>
>>     Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/>
>>
>>     Need commercial support? Contactsales at freeswitch.com <mailto:sales at freeswitch.com>for details.
>>
>>
>>     _________________________________________________________________________
>>     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
>>
>>
>>
>>
>> -- 
>> Guillermo Ruiz Camauer
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> -- 
> ------------------------------------------------------------
> Nathan Neulingernneul at mst.edu
> Missouri S&T Information Technology    (573) 612-1412
> System Administrator - Architect
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
------------------------------------------------------------
Nathan Neulinger                       nneul at mst.edu
Missouri S&T Information Technology    (573) 612-1412
System Administrator - Architect

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From grcamauer at gmail.com  Tue Apr 17 23:21:51 2018
From: grcamauer at gmail.com (Guillermo Ruiz Camauer)
Date: Tue, 17 Apr 2018 20:21:51 -0300
Subject: [Freeswitch-users] Problem with FROM in Gateway comgig with one
	leggged call
Message-ID: <CAPNTnnoQr0bsd2jw_fZ3p4vHFZKBt+6HPGAHCanh5AiU-cAP1g@mail.gmail.com>

All,

I am having a problem with my trunk provider.  They are telling me that I
am sending invalid data in the FROM field in my INVITEs.

send 1105 bytes to udp/[190.210.242.212]:5060 at 20:08:19.303853:
   ------------------------------------------------------------------------
   INVITE sip:111111111#5401155557558 at 190.210.242.212 SIP/2.0
   Via: SIP/2.0/UDP 201.216.208.162;rport;branch=z9hG4bKaUty18S0r7rHH
   Max-Forwards: 70
   From: "Caller Name" <sip:1144444444@*190.210.242.212*>;tag=cUHKHN7pQBKUS
   To: <sip:151885#5401155557558 at 190.210.242.212>
   Call-ID: 0f230991-bd37-1236-769b-1cc1de7988ec
   CSeq: 121667185 INVITE
   Contact: <sip:1144444444 at 201.216.208.162:5060;transport=udp;gw=iplan4>
   User-Agent:
FreeSWITCH-mod_sofia/1.8.0+git~20160928T221047Z~b8c65fb455~64bit
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
REGISTER, REFER, NOTIFY
   Supported: timer, path, replaces
   Allow-Events: talk, hold, conference, refer
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 226
   X-FS-Support: update_display,send_info
   Remote-Party-ID: "Caller Name" <sip:5551231234 at 190.210.242.212
>;party=calling;screen=yes;privacy=off

   v=0
   o=FreeSWITCH 1523985303 1523985304 IN IP4 201.216.208.162
   s=FreeSWITCH
   c=IN IP4 201.216.208.162
   t=0 0
   m=audio 21196 RTP/AVP 0 101
   a=rtpmap:0 PCMU/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=ptime:20


They object to the FROM DOMAIN ( *190.210.242.212) *being equal to the TO
domain.

These are one legged calls, and my gateway configuration looks like this:

<include>
  <gateway name="iplan4">
  <param name="username" value="1144444444"/>
  <param name="realm" value="190.210.242.212"/>
  <param name="from_user" value="1143211234"/>
*  <param name="from_domain" value="201.216.208.162"/>*
  <param name="password" value="ninguna"/>
  <param name="extension" value="1144444444"/>
  <param name="extension-in-contact" value="true"/>
  <param name="proxy" value="190.210.242.212"/>
  <param name="rfc2833-pt" value="101"/>
  <param name="dtmf-type" value="rfc2833"/>
  <param name="liberal-dtmf" value="false"/>
  <param name="apply-inbound-acl" value="iPlan"/>
  <param name="accept-blind-auth" value="true"/>
  <param name="register" value="false"/>
  <param name="register-transport" value="udp"/>
  <param name="retry-seconds" value="30"/>
  </gateway>
</include>


FS is not honoring this line:  *  <param name="from_domain"
value="201.216.208.162"/> *and is instead using the proxy value in the FROM
field.
How do I configure the gateway to use my public IP ( *201.216.208.162*) in
the FROM field?

BTW, it is also not honoring the    *<param name="from_user"
value="1143211234"/>* line.

Thanks,

Guillermo

-- 
Guillermo Ruiz Camauer
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From cong.wang.itsherpa at gmail.com  Wed Apr 18 01:18:37 2018
From: cong.wang.itsherpa at gmail.com (=?utf-8?B?546L6IGh?=)
Date: Wed, 18 Apr 2018 10:18:37 +0900
Subject: [Freeswitch-users] Is it possible to access into an active call by
	admin?
Message-ID: <108D6841-AEA4-4340-8CF5-6AC3CC83A9D0@gmail.com>

Hey all,

I got a new mission about monitor the call. 
We want to access into a certain call and got the video/audio from both 2 legs without be spotted by caller.
Firstly I considered to transmit bridge to conference to support 3-way calling, 
but caller would turn into telephone view immediately after enter the conference room, without waiting callee accept the phone.
Is there any solution for the conference or any suggestion for this function?

Regards.

From alex at freeswitch.com  Wed Apr 18 02:05:05 2018
From: alex at freeswitch.com (Alexey Sibyakin)
Date: Wed, 18 Apr 2018 11:05:05 +0900
Subject: [Freeswitch-users] Is it possible to access into an active call
 by admin?
In-Reply-To: <108D6841-AEA4-4340-8CF5-6AC3CC83A9D0@gmail.com>
References: <108D6841-AEA4-4340-8CF5-6AC3CC83A9D0@gmail.com>
Message-ID: <CAGJLwFhFFdiJa01_1aFzTcV_G05GNPP_gwdpi6qiLKzxYkRUDA@mail.gmail.com>

Hey,

You can try
https://freeswitch.org/confluence/display/FREESWITCH/mod_spy
https://freeswitch.org/confluence/display/FREESWITCH/mod_dptools:+eavesdrop

Alex

On Wed, Apr 18, 2018 at 10:18 AM, 王聡 <cong.wang.itsherpa at gmail.com> wrote:

> Hey all,
>
> I got a new mission about monitor the call.
> We want to access into a certain call and got the video/audio from both 2
> legs without be spotted by caller.
> Firstly I considered to transmit bridge to conference to support 3-way
> calling,
> but caller would turn into telephone view immediately after enter the
> conference room, without waiting callee accept the phone.
> Is there any solution for the conference or any suggestion for this
> function?
>
> Regards.
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




-- 
Alex Sibyakin | Support Engineer
FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045
<https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g>
Email: alex at freeswitch.com
Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/>
Need commercial support? Contact sales at freeswitch.com for details.
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From cong.wang.itsherpa at gmail.com  Wed Apr 18 02:44:48 2018
From: cong.wang.itsherpa at gmail.com (=?utf-8?B?546L6IGh?=)
Date: Wed, 18 Apr 2018 11:44:48 +0900
Subject: [Freeswitch-users] Is it possible to access into an active call
 by admin?
In-Reply-To: <CAGJLwFhFFdiJa01_1aFzTcV_G05GNPP_gwdpi6qiLKzxYkRUDA@mail.gmail.com>
References: <108D6841-AEA4-4340-8CF5-6AC3CC83A9D0@gmail.com>
 <CAGJLwFhFFdiJa01_1aFzTcV_G05GNPP_gwdpi6qiLKzxYkRUDA@mail.gmail.com>
Message-ID: <E617ADF4-F318-4983-99B9-0691C6E4A935@gmail.com>

Hey Alex,

Thanks for your advice, but is it possible for mod_spy to monitor a video call? 
I’ll have a try later. Thanks:)

Regards.

> 2018/04/18 11:05、Alexey Sibyakin <alex at freeswitch.com>のメール:
> 
> Hey,
> 
> You can try 
> https://freeswitch.org/confluence/display/FREESWITCH/mod_spy <https://freeswitch.org/confluence/display/FREESWITCH/mod_spy>
> https://freeswitch.org/confluence/display/FREESWITCH/mod_dptools:+eavesdrop <https://freeswitch.org/confluence/display/FREESWITCH/mod_dptools:+eavesdrop>
> 
> Alex
> 
> On Wed, Apr 18, 2018 at 10:18 AM, 王聡 <cong.wang.itsherpa at gmail.com <mailto:cong.wang.itsherpa at gmail.com>> wrote:
> Hey all,
> 
> I got a new mission about monitor the call. 
> We want to access into a certain call and got the video/audio from both 2 legs without be spotted by caller.
> Firstly I considered to transmit bridge to conference to support 3-way calling, 
> but caller would turn into telephone view immediately after enter the conference room, without waiting callee accept the phone.
> Is there any solution for the conference or any suggestion for this function?
> 
> Regards.
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> http://www.cluecon.com <http://www.cluecon.com/>
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> http://www.freeswitch.org <http://www.freeswitch.org/>
> 
> 
> -- 
> Alex Sibyakin | Support Engineer
> FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g>
> Email: alex at freeswitch.com <mailto:alex at freeswitch.com>
> Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/>
> Need commercial support? Contact sales at freeswitch.com <mailto:sales at freeswitch.com> for details.
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 igor.potjevlesch at gmail.com  Wed Apr 18 08:20:25 2018
From: igor.potjevlesch at gmail.com (igor.potjevlesch at gmail.com)
Date: Wed, 18 Apr 2018 10:20:25 +0200
Subject: [Freeswitch-users] Suddenly lost of fs_cli
In-Reply-To: <CAPNTnnp1mHpxLshGVrzTsyn3ZQGxonKehVcyYhEgb8ppW3igpQ@mail.gmail.com>
References: <003c01d3d623$41492b30$c3db8190$@gmail.com>
 <CAGJLwFgiZRdUi_SJBjBb6KCXaNwK7GDYsckG0xow+YTN_S7mZg@mail.gmail.com>
 <005201d3d649$d320dfa0$79629ee0$@gmail.com>
 <CAGJLwFi=cDMbcTYhdO06MnyZZU-1oqR3QrhN8aRamNtt8KXKTA@mail.gmail.com>
 <008401d3d65b$03649dd0$0a2dd970$@gmail.com>
 <CAPNTnnp1mHpxLshGVrzTsyn3ZQGxonKehVcyYhEgb8ppW3igpQ@mail.gmail.com>
Message-ID: <008401d3d6ee$1a682630$4f387290$@gmail.com>

Indeed. I had to do that. Freeswitch was still processing the calls but replied systematically with Bad Gateway to one leg and sent a BYE to the other after the call has been hang up.

 

De : FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> De la part de Guillermo Ruiz Camauer
Envoyé : mardi 17 avril 2018 18:12
À : FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Objet : Re: [Freeswitch-users] Suddenly lost of fs_cli

 

If it is not writing to tthe log or to the CDR, I think its high time you stopped/kill that instance and start a new one...

 

Guillermo

 

On Tue, Apr 17, 2018 at 11:47 AM, <igor.potjevlesch at gmail.com <mailto:igor.potjevlesch at gmail.com> > wrote:

There is no more log written now. I could see in the last one some errors like this:

2018-04-17 14:41:09.219168 [CRIT] mod_cdr_pg_csv.c:268 Connection to database failed: could not create socket: Too many open files

 

But if I take a network dump, the traffic is processed.

 

Regarding the ESL socket, I'm not sure, do you see it?

Proto Recv-Q Send-Q Local Address               Foreign Address             State       PID/Program name

tcp        0      0 192.168.20.30:5066 <http://192.168.20.30:5066>             0.0.0.0:*                   LISTEN      27314/freeswitch

tcp        0      0 192.168.20.30:7443 <http://192.168.20.30:7443>             0.0.0.0:*                   LISTEN      27314/freeswitch

tcp        0      0 192.168.20.18:5060 <http://192.168.20.18:5060>             0.0.0.0:*                   LISTEN      27314/freeswitch

tcp        0      0 192.168.20.17:5060 <http://192.168.20.17:5060>             0.0.0.0:*                   LISTEN      27314/freeswitch

tcp        0      0 192.168.20.131:5060 <http://192.168.20.131:5060>            0.0.0.0:*                   LISTEN      27314/freeswitch

tcp        0      0 192.168.20.19:5060 <http://192.168.20.19:5060>             0.0.0.0:*                   LISTEN      27314/freeswitch

tcp        0      0 192.168.20.31:5060 <http://192.168.20.31:5060>             0.0.0.0:*                   LISTEN      27314/freeswitch

tcp        0      0 192.168.20.130:5060 <http://192.168.20.130:5060>            0.0.0.0:*                   LISTEN      27314/freeswitch

tcp        0      0 192.168.20.16:5060 <http://192.168.20.16:5060>             0.0.0.0:*                   LISTEN      27314/freeswitch

tcp        0      0 192.168.20.30:5060 <http://192.168.20.30:5060>             0.0.0.0:*                   LISTEN      27314/freeswitch

udp        0      0 192.168.20.30:2408 <http://192.168.20.30:2408>             192.168.20.1:5351 <http://192.168.20.1:5351>              ESTABLISHED 27314/freeswitch

udp        0      0 192.168.20.18:5060 <http://192.168.20.18:5060>             0.0.0.0:*                               27314/freeswitch

udp        0      0 192.168.20.17:5060 <http://192.168.20.17:5060>             0.0.0.0:*                               27314/freeswitch

udp        0      0 192.168.20.131:5060 <http://192.168.20.131:5060>            0.0.0.0:*                               27314/freeswitch

udp        0      0 192.168.20.19:5060 <http://192.168.20.19:5060>             0.0.0.0:*                               27314/freeswitch

udp        0      0 192.168.20.31:5060 <http://192.168.20.31:5060>             0.0.0.0:*                               27314/freeswitch

udp        0      0 192.168.20.130:5060 <http://192.168.20.130:5060>            0.0.0.0:*                               27314/freeswitch

udp        0      0 192.168.20.16:5060 <http://192.168.20.16:5060>             0.0.0.0:*                               27314/freeswitch

udp        0      0 192.168.20.30:5060 <http://192.168.20.30:5060>             0.0.0.0:*                               27314/freeswitch

 

Regards,

 

Igor.

 

De : FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org> > De la part de Alexey Sibyakin
Envoyé : mardi 17 avril 2018 15:27


À : FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org> >
Objet : Re: [Freeswitch-users] Suddenly lost of fs_cli

 

Can you see ESL socket in "ss -putan | column -t" or netstat -putan ? Any errors in the log?

 

Alex

 

On Tue, Apr 17, 2018 at 9:44 PM, <igor.potjevlesch at gmail.com <mailto:igor.potjevlesch at gmail.com> > wrote:

Hello Alex,

 

There is a loop:

[ERROR] fs_cli.c:1610 main() Error Connecting [Socket Connection Error]

[INFO] fs_cli.c:1616 main() Retrying

[ERROR] fs_cli.c:1610 main() Error Connecting [Socket Connection Error]

[INFO] fs_cli.c:1616 main() Retrying

[ERROR] fs_cli.c:1610 main() Error Connecting [Socket Connection Error]

[INFO] fs_cli.c:1616 main() Retrying

[ERROR] fs_cli.c:1610 main() Error Connecting [Socket Connection Error]

[INFO] fs_cli.c:1616 main() Retrying

[ERROR] fs_cli.c:1610 main() Error Connecting [Socket Connection Error]

[INFO] fs_cli.c:1616 main() Retrying

[ERROR] fs_cli.c:1610 main() Error Connecting [Socket Connection Error]

[INFO] fs_cli.c:1616 main() Retrying

[ERROR] fs_cli.c:1610 main() Error Connecting [Socket Connection Error]

[INFO] fs_cli.c:1616 main() Retrying

[…]

 

Regards,

 

Igor

 

De : FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org> > De la part de Alexey Sibyakin
Envoyé : mardi 17 avril 2018 10:46
À : FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org> >
Objet : Re: [Freeswitch-users] Suddenly lost of fs_cli

 

Hello

 

Try to add -rR flags to command

Alex

 

On Tue, Apr 17, 2018 at 5:08 PM, <igor.potjevlesch at gmail.com <mailto:igor.potjevlesch at gmail.com> > wrote:

Hello,

 

Since yesterday, I cannot access to fs_cli anymore after a traffic peak on an instance.

 

I got the following error:

[ERROR] fs_cli.c:1610 main() Error Connecting [Socket Connection Error]

Usage: /usr/local/freeswitch/bin/fs_cli [-H <host>] [-P <port>] [-p <secret>] [-d <level>] [-x command] [-t <timeout_ms>] [profile]

 

Is there something I can do without restarting Freeswitch?

 

Regards,

 

Igor.


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

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

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





 

-- 

Alex Sibyakin | Support Engineer

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

Email:  <mailto:alex at freeswitch.com> alex at freeswitch.com

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

Need commercial support? Contact  <mailto:sales at freeswitch.com> sales at freeswitch.com for details.


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

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

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





 

-- 

Alex Sibyakin | Support Engineer

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

Email:  <mailto:alex at freeswitch.com> alex at freeswitch.com

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

Need commercial support? Contact  <mailto:sales at freeswitch.com> sales at freeswitch.com for details.


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

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

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





 

-- 

Guillermo Ruiz Camauer

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From igor.potjevlesch at gmail.com  Wed Apr 18 08:26:19 2018
From: igor.potjevlesch at gmail.com (igor.potjevlesch at gmail.com)
Date: Wed, 18 Apr 2018 10:26:19 +0200
Subject: [Freeswitch-users] Suddenly lost of fs_cli
In-Reply-To: <53d3149a-3257-3939-c4ea-66a9f1870a3a@mst.edu>
References: <003c01d3d623$41492b30$c3db8190$@gmail.com>
 <CAGJLwFgiZRdUi_SJBjBb6KCXaNwK7GDYsckG0xow+YTN_S7mZg@mail.gmail.com>
 <005201d3d649$d320dfa0$79629ee0$@gmail.com>
 <CAGJLwFi=cDMbcTYhdO06MnyZZU-1oqR3QrhN8aRamNtt8KXKTA@mail.gmail.com>
 <008401d3d65b$03649dd0$0a2dd970$@gmail.com>
 <CAPNTnnp1mHpxLshGVrzTsyn3ZQGxonKehVcyYhEgb8ppW3igpQ@mail.gmail.com>
 <53d3149a-3257-3939-c4ea-66a9f1870a3a@mst.edu>
Message-ID: <008901d3d6ee$eda6abc0$c8f40340$@gmail.com>

You're right. I got a new "crash" today and I checked the limits.conf for this instance. There is nothing define for freeswitch, so default values are applied.

I will update with these values:

 

freeswitch       hard    nofile          500000

freeswitch       soft    nofile          500000

freeswitch       hard    nproc           62815

freeswitch       soft    nproc           62815

 

 

De : FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> De la part de Nathan Neulinger
Envoyé : mardi 17 avril 2018 19:12
À : FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>; Guillermo Ruiz Camauer <grcamauer at gmail.com>
Objet : Re: [Freeswitch-users] Suddenly lost of fs_cli

 

Would also be a really good time to check your /etc/security/limits.conf or limits.d/ settings for freeswitch for file descriptor limit settings.

-- Nathan

On 4/17/18 11:12 AM, Guillermo Ruiz Camauer wrote:

If it is not writing to tthe log or to the CDR, I think its high time you stopped/kill that instance and start a new one... 

 

Guillermo

 

On Tue, Apr 17, 2018 at 11:47 AM, <igor.potjevlesch at gmail.com <mailto:igor.potjevlesch at gmail.com> > wrote:

There is no more log written now. I could see in the last one some errors like this:

2018-04-17 14:41:09.219168 [CRIT] mod_cdr_pg_csv.c:268 Connection to database failed: could not create socket: Too many open files

 

But if I take a network dump, the traffic is processed.

 

Regarding the ESL socket, I'm not sure, do you see it?

Proto Recv-Q Send-Q Local Address               Foreign Address             State       PID/Program name

tcp        0      0 192.168.20.30:5066 <http://192.168.20.30:5066>             0.0.0.0:*                   LISTEN      27314/freeswitch

tcp        0      0 192.168.20.30:7443 <http://192.168.20.30:7443>             0.0.0.0:*                   LISTEN      27314/freeswitch

tcp        0      0 192.168.20.18:5060 <http://192.168.20.18:5060>             0.0.0.0:*                   LISTEN      27314/freeswitch

tcp        0      0 192.168.20.17:5060 <http://192.168.20.17:5060>             0.0.0.0:*                   LISTEN      27314/freeswitch

tcp        0      0 192.168.20.131:5060 <http://192.168.20.131:5060>            0.0.0.0:*                   LISTEN      27314/freeswitch

tcp        0      0 192.168.20.19:5060 <http://192.168.20.19:5060>             0.0.0.0:*                   LISTEN      27314/freeswitch

tcp        0      0 192.168.20.31:5060 <http://192.168.20.31:5060>             0.0.0.0:*                   LISTEN      27314/freeswitch

tcp        0      0 192.168.20.130:5060 <http://192.168.20.130:5060>            0.0.0.0:*                   LISTEN      27314/freeswitch

tcp        0      0 192.168.20.16:5060 <http://192.168.20.16:5060>             0.0.0.0:*                   LISTEN      27314/freeswitch

tcp        0      0 192.168.20.30:5060 <http://192.168.20.30:5060>             0.0.0.0:*                   LISTEN      27314/freeswitch

udp        0      0 192.168.20.30:2408 <http://192.168.20.30:2408>             192.168.20.1:5351 <http://192.168.20.1:5351>              ESTABLISHED 27314/freeswitch

udp        0      0 192.168.20.18:5060 <http://192.168.20.18:5060>             0.0.0.0:*                               27314/freeswitch

udp        0      0 192.168.20.17:5060 <http://192.168.20.17:5060>             0.0.0.0:*                               27314/freeswitch

udp        0      0 192.168.20.131:5060 <http://192.168.20.131:5060>            0.0.0.0:*                               27314/freeswitch

udp        0      0 192.168.20.19:5060 <http://192.168.20.19:5060>             0.0.0.0:*                               27314/freeswitch

udp        0      0 192.168.20.31:5060 <http://192.168.20.31:5060>             0.0.0.0:*                               27314/freeswitch

udp        0      0 192.168.20.130:5060 <http://192.168.20.130:5060>            0.0.0.0:*                               27314/freeswitch

udp        0      0 192.168.20.16:5060 <http://192.168.20.16:5060>             0.0.0.0:*                               27314/freeswitch

udp        0      0 192.168.20.30:5060            0.0.0.0:*                               27314/freeswitch

 

Regards,

 

Igor.

 

De : FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org> > De la part de Alexey Sibyakin
Envoyé : mardi 17 avril 2018 15:27


À : FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org> >
Objet : Re: [Freeswitch-users] Suddenly lost of fs_cli

 

Can you see ESL socket in "ss -putan | column -t" or netstat -putan ? Any errors in the log?

 

Alex

 

On Tue, Apr 17, 2018 at 9:44 PM, <igor.potjevlesch at gmail.com <mailto:igor.potjevlesch at gmail.com> > wrote:

Hello Alex,

 

There is a loop:

[ERROR] fs_cli.c:1610 main() Error Connecting [Socket Connection Error]

[INFO] fs_cli.c:1616 main() Retrying

[ERROR] fs_cli.c:1610 main() Error Connecting [Socket Connection Error]

[INFO] fs_cli.c:1616 main() Retrying

[ERROR] fs_cli.c:1610 main() Error Connecting [Socket Connection Error]

[INFO] fs_cli.c:1616 main() Retrying

[ERROR] fs_cli.c:1610 main() Error Connecting [Socket Connection Error]

[INFO] fs_cli.c:1616 main() Retrying

[ERROR] fs_cli.c:1610 main() Error Connecting [Socket Connection Error]

[INFO] fs_cli.c:1616 main() Retrying

[ERROR] fs_cli.c:1610 main() Error Connecting [Socket Connection Error]

[INFO] fs_cli.c:1616 main() Retrying

[ERROR] fs_cli.c:1610 main() Error Connecting [Socket Connection Error]

[INFO] fs_cli.c:1616 main() Retrying

[…]

 

Regards,

 

Igor

 

De : FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org> > De la part de Alexey Sibyakin
Envoyé : mardi 17 avril 2018 10:46
À : FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org> >
Objet : Re: [Freeswitch-users] Suddenly lost of fs_cli

 

Hello

 

Try to add -rR flags to command

Alex

 

On Tue, Apr 17, 2018 at 5:08 PM, <igor.potjevlesch at gmail.com <mailto:igor.potjevlesch at gmail.com> > wrote:

Hello,

 

Since yesterday, I cannot access to fs_cli anymore after a traffic peak on an instance.

 

I got the following error:

[ERROR] fs_cli.c:1610 main() Error Connecting [Socket Connection Error]

Usage: /usr/local/freeswitch/bin/fs_cli [-H <host>] [-P <port>] [-p <secret>] [-d <level>] [-x command] [-t <timeout_ms>] [profile]

 

Is there something I can do without restarting Freeswitch?

 

Regards,

 

Igor.


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

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

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





 

-- 

Alex Sibyakin | Support Engineer

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

Email:  <mailto:alex at freeswitch.com> alex at freeswitch.com

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

Need commercial support? Contact  <mailto:sales at freeswitch.com> sales at freeswitch.com for details.


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

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

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





 

-- 

Alex Sibyakin | Support Engineer

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

Email:  <mailto:alex at freeswitch.com> alex at freeswitch.com

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

Need commercial support? Contact  <mailto:sales at freeswitch.com> sales at freeswitch.com for details.


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

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

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





 

-- 

Guillermo Ruiz Camauer





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

-- 
------------------------------------------------------------
Nathan Neulinger                       nneul at mst.edu <mailto:nneul at mst.edu> 
Missouri S&T Information Technology    (573) 612-1412
System Administrator - Architect       
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From ksrigo at gmail.com  Wed Apr 18 13:38:18 2018
From: ksrigo at gmail.com (Srigo Kanapathipillai)
Date: Wed, 18 Apr 2018 13:38:18 +0000
Subject: [Freeswitch-users] Problem with FROM in Gateway comgig with one
 leggged call
In-Reply-To: <CAPNTnnoQr0bsd2jw_fZ3p4vHFZKBt+6HPGAHCanh5AiU-cAP1g@mail.gmail.com>
References: <CAPNTnnoQr0bsd2jw_fZ3p4vHFZKBt+6HPGAHCanh5AiU-cAP1g@mail.gmail.com>
Message-ID: <CAAiAxuy5KqxD7Q6Q0FNZyMCw4+UbVUeq7qan__sPHWu5sJcJUg@mail.gmail.com>

Hi,

It should be: - instead of _ between from and domain.

*  <param name="from-domain" value="201.216.208.162"/>*

Srigo

On Wed, 18 Apr 2018, 10:06 Guillermo Ruiz Camauer, <grcamauer at gmail.com>
wrote:

> All,
>
> I am having a problem with my trunk provider.  They are telling me that I
> am sending invalid data in the FROM field in my INVITEs.
>
> send 1105 bytes to udp/[190.210.242.212]:5060 at 20:08:19.303853:
>    ------------------------------------------------------------------------
>    INVITE sip:111111111#5401155557558 at 190.210.242.212 SIP/2.0
>    Via: SIP/2.0/UDP 201.216.208.162;rport;branch=z9hG4bKaUty18S0r7rHH
>    Max-Forwards: 70
>    From: "Caller Name" <sip:1144444444@*190.210.242.212*
> >;tag=cUHKHN7pQBKUS
>    To: <sip:151885#5401155557558 at 190.210.242.212>
>    Call-ID: 0f230991-bd37-1236-769b-1cc1de7988ec
>    CSeq: 121667185 INVITE
>    Contact: <sip:1144444444 at 201.216.208.162:5060;transport=udp;gw=iplan4>
>    User-Agent:
> FreeSWITCH-mod_sofia/1.8.0+git~20160928T221047Z~b8c65fb455~64bit
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
> REGISTER, REFER, NOTIFY
>    Supported: timer, path, replaces
>    Allow-Events: talk, hold, conference, refer
>    Content-Type: application/sdp
>    Content-Disposition: session
>    Content-Length: 226
>    X-FS-Support: update_display,send_info
>    Remote-Party-ID: "Caller Name" <sip:5551231234 at 190.210.242.212
> >;party=calling;screen=yes;privacy=off
>
>    v=0
>    o=FreeSWITCH 1523985303 1523985304 IN IP4 201.216.208.162
>    s=FreeSWITCH
>    c=IN IP4 201.216.208.162
>    t=0 0
>    m=audio 21196 RTP/AVP 0 101
>    a=rtpmap:0 PCMU/8000
>    a=rtpmap:101 telephone-event/8000
>    a=fmtp:101 0-16
>    a=ptime:20
>
>
> They object to the FROM DOMAIN ( *190.210.242.212) *being equal to the TO
> domain.
>
> These are one legged calls, and my gateway configuration looks like this:
>
> <include>
>   <gateway name="iplan4">
>   <param name="username" value="1144444444"/>
>   <param name="realm" value="190.210.242.212"/>
>   <param name="from_user" value="1143211234"/>
> *  <param name="from_domain" value="201.216.208.162"/>*
>   <param name="password" value="ninguna"/>
>   <param name="extension" value="1144444444"/>
>   <param name="extension-in-contact" value="true"/>
>   <param name="proxy" value="190.210.242.212"/>
>   <param name="rfc2833-pt" value="101"/>
>   <param name="dtmf-type" value="rfc2833"/>
>   <param name="liberal-dtmf" value="false"/>
>   <param name="apply-inbound-acl" value="iPlan"/>
>   <param name="accept-blind-auth" value="true"/>
>   <param name="register" value="false"/>
>   <param name="register-transport" value="udp"/>
>   <param name="retry-seconds" value="30"/>
>   </gateway>
> </include>
>
>
> FS is not honoring this line:  *  <param name="from_domain"
> value="201.216.208.162"/> *and is instead using the proxy value in the
> FROM field.
> How do I configure the gateway to use my public IP ( *201.216.208.162*)
> in the FROM field?
>
> BTW, it is also not honoring the    *<param name="from_user"
> value="1143211234"/>* line.
>
> Thanks,
>
> Guillermo
>
> --
> Guillermo Ruiz Camauer
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From shaun.stokes at itec-support.co.uk  Wed Apr 18 14:08:59 2018
From: shaun.stokes at itec-support.co.uk (Shaun Stokes)
Date: Wed, 18 Apr 2018 14:08:59 +0000
Subject: [Freeswitch-users] LUA menu on H264 device causes 500 Server
	Internal Error
In-Reply-To: <CAAiAxuy5KqxD7Q6Q0FNZyMCw4+UbVUeq7qan__sPHWu5sJcJUg@mail.gmail.com>
References: <CAPNTnnoQr0bsd2jw_fZ3p4vHFZKBt+6HPGAHCanh5AiU-cAP1g@mail.gmail.com>,
 <CAAiAxuy5KqxD7Q6Q0FNZyMCw4+UbVUeq7qan__sPHWu5sJcJUg@mail.gmail.com>
Message-ID: <1524060539124.73181@itec-support.co.uk>

Hi All,


We have an issue when calling a LUA script with menu options from an H264 video enabled device the call drops part way into the call while navigating a menu due to '500 Server Internal Error'.


Trying to establish if this is a problem our end or a bug, has anyone else experienced this issue before?


Logs from an example below, our IP address is masked as x.x.x.x


nua.c:633 nua_invite() nua: nua_invite: entering
nua_stack.c:529 nua_signal() nua(0x7f25b00e8e10): sent signal r_invite
nua_stack.c:569 nua_stack_signal() nua(0x7f25b00e8e10): recv signal r_invite
nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering
soa.c:403 soa_set_params() soa_set_params(static::0x7f25b00f2ce0, ...) called
soa.c:1302 soa_init_offer_answer() soa_init_offer_answer(static::0x7f25b00f2ce0) called
nta.c:2665 nta_tpn_by_url() nta: selecting scheme sip
tport.c:4588 tport_by_name() tport(0x7f25b0005100): found 0x7f25b0147320 by name tls/x.x.x.x:7859
tport.c:3257 tport_tsend() tport_tsend(0x7f25b0147320) tpn = tls/x.x.x.x:7859
tport_type_tls.c:534 tport_tls_send() tport_tls_writevec: vec 0x7f25b00e40c0 0x7f25b001fe60 755 (755)
EXECUTE sofia/internal/2001 at domain sleep(2000)
tport_type_tls.c:534 tport_tls_send() tport_tls_writevec: vec 0x7f25b00e40c0 0x7f25b01ca460 9 (9)
tport.c:3594 tport_vsend() tport_vsend(0x7f25b0147320): 764 bytes of 764 to tls/x.x.x.x:7859
tport.c:3492 tport_send_msg() tport_vsend returned 764
tport.c:2296 tport_set_secondary_timer() tport(0x7f25b0147320): reset timer
nta.c:8304 outgoing_send() nta: sent INVITE (121693762) to tls/x.x.x.x:7859
tport.c:4160 tport_pend() tport_pend(0x7f25b0147320): pending 0x7f25b013b770 for tls/x.x.x.x:7859 (already 0)
nta.c:1350 set_timeout() nta: timer set to 32000 ms
nua_session.c:4145 signal_call_state_change() nua(0x7f25b00e8e10): ready call updated: calling
nua_stack.c:269 nua_stack_event() nua(0x7f25b00e8e10): event i_state INVITE sent
nua_stack.c:359 nua_application_event() nua: nua_application_event: entering
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
2018-04-18 14:54:13.139241 [DEBUG] sofia.c:7084 Channel sofia/internal/2001 at domain entering state [calling][0]
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
tport.c:2773 tport_wakeup() tport_wakeup(0x7f25b0147320): events IN
tport.c:2864 tport_recv_event() tport_recv_event(0x7f25b0147320)
tport_type_tls.c:434 tport_tls_recv() tport_tls_recv(0x7f25b0147320): tls_read() returned 473
tport.c:3205 tport_recv_iovec() tport_recv_iovec(0x7f25b0147320) msg 0x7f25b00d65f0 from (tls/x.x.x.x:7859) has 473 bytes, veclen = 1
tport.c:3023 tport_deliver() tport_deliver(0x7f25b0147320): msg 0x7f25b00d65f0 (473 bytes) from tls/x.x.x.x:7859/sips next=(nil)
nta.c:3299 agent_recv_response() nta: received 500 Server Internal Error for INVITE (121693762)
nta.c:3366 agent_recv_response() nta: 500 Server Internal Error is going to a transaction
nta.c:9564 outgoing_estimate_delay() nta_outgoing: RTT is 78.612 ms
tport.c:4222 tport_release() tport_release(0x7f25b0147320): 0x7f25b013b770 by 0x7f25b01d3eb0 with 0x7f25b00d65f0
tport.c:4588 tport_by_name() tport(0x7f25b0005100): found 0x7f25b0147320 by name tls/x.x.x.x:7859
tport.c:3257 tport_tsend() tport_tsend(0x7f25b0147320) tpn = tls/x.x.x.x:7859
tport_type_tls.c:534 tport_tls_send() tport_tls_writevec: vec 0x7f25b00e40c0 0x7f25b011b590 403 (403)
tport.c:3594 tport_vsend() tport_vsend(0x7f25b0147320): 403 bytes of 403 to tls/x.x.x.x:7859
tport.c:3492 tport_send_msg() tport_vsend returned 403
tport.c:2296 tport_set_secondary_timer() tport(0x7f25b0147320): reset timer
nta.c:8304 outgoing_send() nta: sent ACK (121693762) to tls/x.x.x.x:7859
nta.c:8722 outgoing_free() nta: outgoing_free(0x7f25b0146630)
nua_stack.c:271 nua_stack_event() nua(0x7f25b00e8e10): event r_invite 500 Server Internal Error
nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering
soa.c:1784 soa_terminate() soa_terminate(static::0x7f25b00f2ce0) called
nua_stack.c:359 nua_application_event() nua: nua_application_event: entering
soa.c:1302 soa_init_offer_answer() soa_init_offer_answer(static::0x7f25b00f2ce0) called
nta.c:2665 nta_tpn_by_url() nta: selecting scheme sip
tport.c:4588 tport_by_name() tport(0x7f25b0005100): found 0x7f25b0147320 by name tls/x.x.x.x:7859
tport.c:3257 tport_tsend() tport_tsend(0x7f25b0147320) tpn = tls/x.x.x.x:7859
tport_type_tls.c:534 tport_tls_send() tport_tls_writevec: vec 0x7f25b00e40c0 0x7f25b011b590 625 (625)
tport.c:3594 tport_vsend() tport_vsend(0x7f25b0147320): 625 bytes of 625 to tls/x.x.x.x:7859
tport.c:3492 tport_send_msg() tport_vsend returned 625
tport.c:2296 tport_set_secondary_timer() tport(0x7f25b0147320): reset timer
nta.c:8304 outgoing_send() nta: sent BYE (121693763) to tls/x.x.x.x:7859
tport.c:4160 tport_pend() tport_pend(0x7f25b0147320): pending 0x7f25b01128c0 for tls/x.x.x.x:7859 (already 0)
nua_session.c:4139 signal_call_state_change() nua(0x7f25b00e8e10): call state changed: ready -> terminating
nua_stack.c:271 nua_stack_event() nua(0x7f25b00e8e10): event i_state 500 Server Internal Error
nua_stack.c:359 nua_application_event() nua: nua_application_event: entering
tport.c:2296 tport_set_secondary_timer() tport(0x7f25b0147320): reset timer
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
2018-04-18 14:54:13.239253 [DEBUG] sofia.c:7084 Channel sofia/internal/2001 at domain entering state [terminating][500]
2018-04-18 14:54:13.239253 [NOTICE] sofia.c:8273 Hangup sofia/internal/2001 at domain [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE]
nua.c:342 nua_handle_bind() nua: nua_handle_bind: entering
nua.c:921 nua_handle_destroy() nua: nua_handle_destroy: entering
nua_stack.c:569 nua_stack_signal() nua(0x7f25b00e8e10): recv signal r_destroy
nua_dialog.c:397 nua_dialog_usage_remove_at() nua(0x7f25b00e8e10): removing session usage
nua_stack.c:529 nua_signal() nua(0x7f25b00e8e10): sent signal r_destroy
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
nua_session.c:4139 signal_call_state_change() nua(0x7f25b00e8e10): call state changed: terminating -> terminated
nua.c:342 nua_handle_bind() nua: nua_handle_bind: entering
nua.c:921 nua_handle_destroy() nua: nua_handle_destroy: entering
nua_stack.c:269 nua_stack_event() nua(0x7f25b00e8e10): event i_state Terminated
nua_stack.c:269 nua_stack_event() nua(0x7f25b00e8e10): event i_terminated Terminated
soa.c:356 soa_destroy() soa_destroy(static::0x7f25b00f2ce0) called
nta.c:4470 nta_leg_destroy() nta_leg_destroy(0x7f25b013afd0)
2018-04-18 14:54:13.239253 [DEBUG] switch_core_session.c:2815 sofia/internal/2001 at domain? skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
2018-04-18 14:54:13.239253 [DEBUG] freeswitch_lua.cpp:382 DBH handle 0x7f256404f450 released.
2018-04-18 14:54:13.239253 [DEBUG] switch_cpp.cpp:1112 sofia/internal/2001 at domain destroy/unlink session from object
2018-04-18 14:54:13.239253 [DEBUG] switch_core_media.c:5970 sofia/internal/2001 at domain Video thread ended


Thanks,

Shaun


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From grcamauer at gmail.com  Wed Apr 18 14:41:25 2018
From: grcamauer at gmail.com (Guillermo Ruiz Camauer)
Date: Wed, 18 Apr 2018 11:41:25 -0300
Subject: [Freeswitch-users] Problem with FROM in Gateway comgig with one
 leggged call
In-Reply-To: <CAAiAxuy5KqxD7Q6Q0FNZyMCw4+UbVUeq7qan__sPHWu5sJcJUg@mail.gmail.com>
References: <CAPNTnnoQr0bsd2jw_fZ3p4vHFZKBt+6HPGAHCanh5AiU-cAP1g@mail.gmail.com>
 <CAAiAxuy5KqxD7Q6Q0FNZyMCw4+UbVUeq7qan__sPHWu5sJcJUg@mail.gmail.com>
Message-ID: <CAPNTnnqFTbVx1bU1FchTe-AZi5b+NdnGqht3cNGHD5wosdGnjw@mail.gmail.com>

That was it, thanks!


Guillermo


On Wed, Apr 18, 2018 at 10:38 AM, Srigo Kanapathipillai <ksrigo at gmail.com>
wrote:

> Hi,
>
> It should be: - instead of _ between from and domain.
>
> *  <param name="from-domain" value="201.216.208.162"/>*
>
> Srigo
>
> On Wed, 18 Apr 2018, 10:06 Guillermo Ruiz Camauer, <grcamauer at gmail.com>
> wrote:
>
>> All,
>>
>> I am having a problem with my trunk provider.  They are telling me that I
>> am sending invalid data in the FROM field in my INVITEs.
>>
>> send 1105 bytes to udp/[190.210.242.212]:5060 at 20:08:19.303853:
>>    -----------------------------------------------------------
>> -------------
>>    INVITE sip:111111111#5401155557558 at 190.210.242.212 SIP/2.0
>>    Via: SIP/2.0/UDP 201.216.208.162;rport;branch=z9hG4bKaUty18S0r7rHH
>>    Max-Forwards: 70
>>    From: "Caller Name" <sip:1144444444@*190.210.242.212*
>> >;tag=cUHKHN7pQBKUS
>>    To: <sip:151885#5401155557558 at 190.210.242.212>
>>    Call-ID: 0f230991-bd37-1236-769b-1cc1de7988ec
>>    CSeq: 121667185 INVITE
>>    Contact: <sip:1144444444 at 201.216.208.162:5060;transport=udp;gw=iplan4>
>>    User-Agent: FreeSWITCH-mod_sofia/1.8.0+git~20160928T221047Z~
>> b8c65fb455~64bit
>>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
>> REGISTER, REFER, NOTIFY
>>    Supported: timer, path, replaces
>>    Allow-Events: talk, hold, conference, refer
>>    Content-Type: application/sdp
>>    Content-Disposition: session
>>    Content-Length: 226
>>    X-FS-Support: update_display,send_info
>>    Remote-Party-ID: "Caller Name" <sip:5551231234 at 190.210.242.212
>> >;party=calling;screen=yes;privacy=off
>>
>>    v=0
>>    o=FreeSWITCH 1523985303 1523985304 IN IP4 201.216.208.162
>>    s=FreeSWITCH
>>    c=IN IP4 201.216.208.162
>>    t=0 0
>>    m=audio 21196 RTP/AVP 0 101
>>    a=rtpmap:0 PCMU/8000
>>    a=rtpmap:101 telephone-event/8000
>>    a=fmtp:101 0-16
>>    a=ptime:20
>>
>>
>> They object to the FROM DOMAIN ( *190.210.242.212) *being equal to the
>> TO domain.
>>
>> These are one legged calls, and my gateway configuration looks like this:
>>
>> <include>
>>   <gateway name="iplan4">
>>   <param name="username" value="1144444444"/>
>>   <param name="realm" value="190.210.242.212"/>
>>   <param name="from_user" value="1143211234"/>
>> *  <param name="from_domain" value="201.216.208.162"/>*
>>   <param name="password" value="ninguna"/>
>>   <param name="extension" value="1144444444"/>
>>   <param name="extension-in-contact" value="true"/>
>>   <param name="proxy" value="190.210.242.212"/>
>>   <param name="rfc2833-pt" value="101"/>
>>   <param name="dtmf-type" value="rfc2833"/>
>>   <param name="liberal-dtmf" value="false"/>
>>   <param name="apply-inbound-acl" value="iPlan"/>
>>   <param name="accept-blind-auth" value="true"/>
>>   <param name="register" value="false"/>
>>   <param name="register-transport" value="udp"/>
>>   <param name="retry-seconds" value="30"/>
>>   </gateway>
>> </include>
>>
>>
>> FS is not honoring this line:  *  <param name="from_domain"
>> value="201.216.208.162"/> *and is instead using the proxy value in the
>> FROM field.
>> How do I configure the gateway to use my public IP ( *201.216.208.162*)
>> in the FROM field?
>>
>> BTW, it is also not honoring the    *<param name="from_user"
>> value="1143211234"/>* line.
>>
>> Thanks,
>>
>> Guillermo
>>
>> --
>> Guillermo Ruiz Camauer
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Guillermo Ruiz Camauer
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From mickael at winlux.fr  Wed Apr 18 15:21:30 2018
From: mickael at winlux.fr (Mickael Hubert)
Date: Wed, 18 Apr 2018 17:21:30 +0200
Subject: [Freeswitch-users] Send RTP to external server
In-Reply-To: <CA+fRVSXFwwqQHoPiM7Dv0rKMy=E2_hAwUZA8Y6e5sOwk6n4tQw@mail.gmail.com>
References: <CY1PR07MB23771896AA8A186E0D283560BEC70@CY1PR07MB2377.namprd07.prod.outlook.com>
 <1760115799.197.1519921479661.JavaMail.sfa@EDISTAR-SFA>
 <CA+fRVSXFwwqQHoPiM7Dv0rKMy=E2_hAwUZA8Y6e5sOwk6n4tQw@mail.gmail.com>
Message-ID: <CA+fRVSV6LvYeYaEdzvKmdQEL4GwYRpxmxOEq8=9SAqYXuHkS4A@mail.gmail.com>

Hi all,
I worked on this POC script:
https://github.com/Mickaelh51/rtp_parser/blob/master/pyshark_rtp_test.py
It can parse pcap file with real sip calls.

I can create a raw file with audio data (one file by leg), It's exactly I
want do do ;)

But now I wan to stream this to kaldi with websocket... it's hard :( (for
me)

Do you think I'm on the good way with this POC ?

I can see other ways maybe:
- https://freeswitch.org/confluence/display/FREESWITCH/mod_vlc
- https://freeswitch.org/confluence/display/FREESWITCH/mod_shout

Maybe it's easier to create a "gateway script" to read vlc or shout flow
and send it to kaldi ?

What do you think about this new idea ?

The last chance: hack one of theses modules to put audio stream to kaldi...
it's not the good way for my c++ skills ;)
https://freeswitch.org/stash/projects/FS/repos/freeswitch/browse/src/mod/asr_tts

thanks in advance

2018-03-02 11:09 GMT+01:00 Mickael Hubert <mickael at winlux.fr>:

> Hi
> thanks a lot for your answers.
> Do you split media stream ? I want to stream only caller voice for
> exemple. My ASR works better if it recognizes only one stream ;)
> thanks in advance
>
> 2018-03-01 17:24 GMT+01:00 Stefano Favaro <stefano.favaro at edistar.com>:
>
>>
>> Hello, I'm currently using:
>>
>>
>> Freeswitch with mod_unimrcp and the plugin from UniMRCP (
>> http://unimrcp.org/gsr)
>>
>>
>> It connects to Google Speech API for audio transcription in realtime
>> using mrcp.
>>
>>
>> Stefano.
>>
>>
>>
>>
>>
>> ------------------------------
>> *Da: *"Robert Mundkowsky" <rmundkowsky at ets.org>
>> *A: *"FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>> *Inviato: *Mercoledì, 28 febbraio 2018 20:17:01
>> *Oggetto: *Re: [Freeswitch-users] Send RTP to external server
>>
>>
>> It is actually fairly common to want to stream audio/video to somewhere
>> to process the data in real time, rather than waiting until recording is
>> finished.
>>
>>
>>
>> For example, if want to convert audio to text (ASR) then you do not want
>> to wait until the conference is over before you start the ASR.  Imagine,
>> your ASR takes 1 minute to convert 1 minute of audio then if a conference
>> is 30 minutes long, you would have to wait another 30 minutes for the ASR
>> to finish before you could do something with the text.
>>
>>
>>
>> I believe you can use gstreamer (https://gstreamer.freedesktop.org/) to
>> handle receiving the RTP from a FreeSWITCH conference.  I think it allows
>> you to sample frames if you want to get frames from video granted I think
>> you are only interested in audio, but I think you still have to develop a
>> daemon that understand SIP/RTP in order to talk to FreeSWITCH.  Another
>> approach might be to create a plugin for the Unimrcp (MRCPv2 server
>> http://www.unimrcp.org/ ) and talk to the FreeSWITCH using their mrcp
>> module and tight that to the conference call.  You might be able to hack
>> the Kaldi plugin for Unimrcp to do what you want.
>>
>>
>>
>> Robert
>>
>>
>>
>> *From:* FreeSWITCH-users [mailto:freeswitch-users-bounc
>> es at lists.freeswitch.org] *On Behalf Of *Tihomir Culjaga
>> *Sent:* Wednesday, February 28, 2018 1:47 PM
>> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> *Subject:* Re: [Freeswitch-users] Send RTP to external server
>>
>>
>>
>> why do you want to stream... isn't it enough just to rsync the file once
>> the call is finished ?
>>
>>
>>
>> On 28 February 2018 at 16:16, Mickael Hubert <mickael at winlux.fr> wrote:
>>
>> Hi
>> thanks a lot for your answer.
>>
>> But I want to send stream to external server, not record in audio file. i
>> can use record to "capture" the voice, but to stream it, it's more
>> complicated ;)
>>
>> thanks in advance
>>
>>
>>
>> 2018-02-21 0:50 GMT+01:00 Brian West <brian at freeswitch.com>:
>>
>> you can already do this without SIPREC in freeswitch. By setting the
>> RECORD_READ_ONLY or RECORD_WRITE_ONLY variables.
>>
>>
>>
>> /b
>>
>>
>>
>>
>>
>> On Fri, Feb 16, 2018 at 10:33 AM, Mickael Hubert <mickael at winlux.fr>
>> wrote:
>>
>> Hi list,
>> I want to record each call through freeswitch. But i want record only
>> caller (SSRC 1) OR callee (SSRC 2) voice (not both).
>>
>> I read about SIPREC, Jack, etc ... not interesting
>>
>> Do you have a idea for me please ?
>>
>> Thanks in advance
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.freeswitchsolutions.com&data=02%7C01%7Crmundkowsky%40ets.org%7C4f0c6786eb0f40725f8a08d57edbe173%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636554405236219602&sdata=6wPzLzP8i%2FDBCXxEyl%2F7xIPJ2hYQy9C2NNGXSiRvfKc%3D&reserved=0>
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.freeswitch.org&data=02%7C01%7Crmundkowsky%40ets.org%7C4f0c6786eb0f40725f8a08d57edbe173%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636554405236219602&sdata=36oG%2FMWaCRXZ%2BfGlz79y3Uv80cWbJvOvf3UtJrUZgKU%3D&reserved=0>
>> http://confluence.freeswitch.org
>> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fconfluence.freeswitch.org&data=02%7C01%7Crmundkowsky%40ets.org%7C4f0c6786eb0f40725f8a08d57edbe173%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636554405236219602&sdata=Fvt9Vq6L9eMPg6cfmpWZURJdemUPRGBOTkXsCKDC5yk%3D&reserved=0>
>> http://www.cluecon.com
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From babak.freeswitch at gmail.com  Wed Apr 18 09:45:06 2018
From: babak.freeswitch at gmail.com (Babak Yakhchali)
Date: Wed, 18 Apr 2018 14:15:06 +0430
Subject: [Freeswitch-users] What happens to a call when running event_socket
	after call is transferred?
Message-ID: <CAEVfcaYHiOY6tTUv_r2=DKa47NRkeHM7FK6G7x-g+QUaLNkEtA@mail.gmail.com>

Hi
this is my simple dial plan to run an outbound esl application:

<extension name="intelligent">
<condition field="destination_number" expression=".*$">
<action application="sleep" data="200"/>
<action application="socket" data="127.0.0.1:3003 async full"/>
</condition>
</extension>

the application is a nodejs process which answers the call and plays some
files and then bridges the call to a user. As you can see in my dialplan
all calls are handled in this same way using esl. Problem occurs when a
bridged call is transferred to another number. at this point there will be
two socket connections for the same call,if I close the first socket the
call is hanged up with this message on logs:
"... has executed the last dialplan instruction, hanging up"
 and after this the second socket will close too. Is this expected or I'm
doing something wrong?
thanks
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From mike at jerris.com  Wed Apr 18 20:58:03 2018
From: mike at jerris.com (Michael Jerris)
Date: Wed, 18 Apr 2018 16:58:03 -0400
Subject: [Freeswitch-users] LUA menu on H264 device causes 500 Server
 Internal Error
In-Reply-To: <1524060539124.73181@itec-support.co.uk>
References: <CAPNTnnoQr0bsd2jw_fZ3p4vHFZKBt+6HPGAHCanh5AiU-cAP1g@mail.gmail.com>
 <CAAiAxuy5KqxD7Q6Q0FNZyMCw4+UbVUeq7qan__sPHWu5sJcJUg@mail.gmail.com>
 <1524060539124.73181@itec-support.co.uk>
Message-ID: <93A99129-2BC1-46C0-BA7E-759C73922F5C@jerris.com>

this full log of the call UNMODIFIED with a full sip trace may be able to help answer your questions.

> On Apr 18, 2018, at 10:08 AM, Shaun Stokes <shaun.stokes at itec-support.co.uk> wrote:
> 
> Hi All,
> 
> We have an issue when calling a LUA script with menu options from an H264 video enabled device the call drops part way into the call while navigating a menu due to '500 Server Internal Error'.
> 
> Trying to establish if this is a problem our end or a bug, has anyone else experienced this issue before?
> 
> Logs from an example below, our IP address is masked as x.x.x.x
> 
> nua.c:633 nua_invite() nua: nua_invite: entering 
> nua_stack.c:529 nua_signal() nua(0x7f25b00e8e10): sent signal r_invite 
> nua_stack.c:569 nua_stack_signal() nua(0x7f25b00e8e10): recv signal r_invite 
> nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering 
> soa.c:403 soa_set_params() soa_set_params(static::0x7f25b00f2ce0, ...) called 
> soa.c:1302 soa_init_offer_answer() soa_init_offer_answer(static::0x7f25b00f2ce0) called 
> nta.c:2665 nta_tpn_by_url() nta: selecting scheme sip 
> tport.c:4588 tport_by_name() tport(0x7f25b0005100): found 0x7f25b0147320 by name tls/x.x.x.x:7859 
> tport.c:3257 tport_tsend() tport_tsend(0x7f25b0147320) tpn = tls/x.x.x.x:7859 
> tport_type_tls.c:534 tport_tls_send() tport_tls_writevec: vec 0x7f25b00e40c0 0x7f25b001fe60 755 (755) 
> EXECUTE sofia/internal/2001 at domain sleep(2000) 
> tport_type_tls.c:534 tport_tls_send() tport_tls_writevec: vec 0x7f25b00e40c0 0x7f25b01ca460 9 (9) 
> tport.c:3594 tport_vsend() tport_vsend(0x7f25b0147320): 764 bytes of 764 to tls/x.x.x.x:7859 
> tport.c:3492 tport_send_msg() tport_vsend returned 764 
> tport.c:2296 tport_set_secondary_timer() tport(0x7f25b0147320): reset timer 
> nta.c:8304 outgoing_send() nta: sent INVITE (121693762) to tls/x.x.x.x:7859 
> tport.c:4160 tport_pend() tport_pend(0x7f25b0147320): pending 0x7f25b013b770 for tls/x.x.x.x:7859 (already 0) 
> nta.c:1350 set_timeout() nta: timer set to 32000 ms 
> nua_session.c:4145 signal_call_state_change() nua(0x7f25b00e8e10): ready call updated: calling 
> nua_stack.c:269 nua_stack_event() nua(0x7f25b00e8e10): event i_state INVITE sent 
> nua_stack.c:359 nua_application_event() nua: nua_application_event: entering 
> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering 
> 2018-04-18 14:54:13.139241 [DEBUG] sofia.c:7084 Channel sofia/internal/2001 at domain entering state [calling][0] 
> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering 
> tport.c:2773 tport_wakeup() tport_wakeup(0x7f25b0147320): events IN 
> tport.c:2864 tport_recv_event() tport_recv_event(0x7f25b0147320) 
> tport_type_tls.c:434 tport_tls_recv() tport_tls_recv(0x7f25b0147320): tls_read() returned 473 
> tport.c:3205 tport_recv_iovec() tport_recv_iovec(0x7f25b0147320) msg 0x7f25b00d65f0 from (tls/x.x.x.x:7859) has 473 bytes, veclen = 1 
> tport.c:3023 tport_deliver() tport_deliver(0x7f25b0147320): msg 0x7f25b00d65f0 (473 bytes) from tls/x.x.x.x:7859/sips next=(nil) 
> nta.c:3299 agent_recv_response() nta: received 500 Server Internal Error for INVITE (121693762) 
> nta.c:3366 agent_recv_response() nta: 500 Server Internal Error is going to a transaction 
> nta.c:9564 outgoing_estimate_delay() nta_outgoing: RTT is 78.612 ms 
> tport.c:4222 tport_release() tport_release(0x7f25b0147320): 0x7f25b013b770 by 0x7f25b01d3eb0 with 0x7f25b00d65f0 
> tport.c:4588 tport_by_name() tport(0x7f25b0005100): found 0x7f25b0147320 by name tls/x.x.x.x:7859 
> tport.c:3257 tport_tsend() tport_tsend(0x7f25b0147320) tpn = tls/x.x.x.x:7859 
> tport_type_tls.c:534 tport_tls_send() tport_tls_writevec: vec 0x7f25b00e40c0 0x7f25b011b590 403 (403) 
> tport.c:3594 tport_vsend() tport_vsend(0x7f25b0147320): 403 bytes of 403 to tls/x.x.x.x:7859 
> tport.c:3492 tport_send_msg() tport_vsend returned 403 
> tport.c:2296 tport_set_secondary_timer() tport(0x7f25b0147320): reset timer 
> nta.c:8304 outgoing_send() nta: sent ACK (121693762) to tls/x.x.x.x:7859 
> nta.c:8722 outgoing_free() nta: outgoing_free(0x7f25b0146630) 
> nua_stack.c:271 nua_stack_event() nua(0x7f25b00e8e10): event r_invite 500 Server Internal Error 
> nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering 
> soa.c:1784 soa_terminate() soa_terminate(static::0x7f25b00f2ce0) called 
> nua_stack.c:359 nua_application_event() nua: nua_application_event: entering 
> soa.c:1302 soa_init_offer_answer() soa_init_offer_answer(static::0x7f25b00f2ce0) called 
> nta.c:2665 nta_tpn_by_url() nta: selecting scheme sip 
> tport.c:4588 tport_by_name() tport(0x7f25b0005100): found 0x7f25b0147320 by name tls/x.x.x.x:7859 
> tport.c:3257 tport_tsend() tport_tsend(0x7f25b0147320) tpn = tls/x.x.x.x:7859 
> tport_type_tls.c:534 tport_tls_send() tport_tls_writevec: vec 0x7f25b00e40c0 0x7f25b011b590 625 (625) 
> tport.c:3594 tport_vsend() tport_vsend(0x7f25b0147320): 625 bytes of 625 to tls/x.x.x.x:7859 
> tport.c:3492 tport_send_msg() tport_vsend returned 625 
> tport.c:2296 tport_set_secondary_timer() tport(0x7f25b0147320): reset timer 
> nta.c:8304 outgoing_send() nta: sent BYE (121693763) to tls/x.x.x.x:7859 
> tport.c:4160 tport_pend() tport_pend(0x7f25b0147320): pending 0x7f25b01128c0 for tls/x.x.x.x:7859 (already 0) 
> nua_session.c:4139 signal_call_state_change() nua(0x7f25b00e8e10): call state changed: ready -> terminating 
> nua_stack.c:271 nua_stack_event() nua(0x7f25b00e8e10): event i_state 500 Server Internal Error 
> nua_stack.c:359 nua_application_event() nua: nua_application_event: entering 
> tport.c:2296 tport_set_secondary_timer() tport(0x7f25b0147320): reset timer 
> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering 
> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering 
> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering 
> 2018-04-18 14:54:13.239253 [DEBUG] sofia.c:7084 Channel sofia/internal/2001 at domain entering state [terminating][500] 
> 2018-04-18 14:54:13.239253 [NOTICE] sofia.c:8273 Hangup sofia/internal/2001 at domain [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE] 
> nua.c:342 nua_handle_bind() nua: nua_handle_bind: entering 
> nua.c:921 nua_handle_destroy() nua: nua_handle_destroy: entering 
> nua_stack.c:569 nua_stack_signal() nua(0x7f25b00e8e10): recv signal r_destroy 
> nua_dialog.c:397 nua_dialog_usage_remove_at() nua(0x7f25b00e8e10): removing session usage 
> nua_stack.c:529 nua_signal() nua(0x7f25b00e8e10): sent signal r_destroy 
> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering 
> nua_session.c:4139 signal_call_state_change() nua(0x7f25b00e8e10): call state changed: terminating -> terminated 
> nua.c:342 nua_handle_bind() nua: nua_handle_bind: entering 
> nua.c:921 nua_handle_destroy() nua: nua_handle_destroy: entering 
> nua_stack.c:269 nua_stack_event() nua(0x7f25b00e8e10): event i_state Terminated 
> nua_stack.c:269 nua_stack_event() nua(0x7f25b00e8e10): event i_terminated Terminated 
> soa.c:356 soa_destroy() soa_destroy(static::0x7f25b00f2ce0) called 
> nta.c:4470 nta_leg_destroy() nta_leg_destroy(0x7f25b013afd0) 
> 2018-04-18 14:54:13.239253 [DEBUG] switch_core_session.c:2815 sofia/internal/2001 at domain​ skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) 
> 2018-04-18 14:54:13.239253 [DEBUG] freeswitch_lua.cpp:382 DBH handle 0x7f256404f450 released. 
> 2018-04-18 14:54:13.239253 [DEBUG] switch_cpp.cpp:1112 sofia/internal/2001 at domain destroy/unlink session from object 
> 2018-04-18 14:54:13.239253 [DEBUG] switch_core_media.c:5970 sofia/internal/2001 at domain Video thread ended
> 
> 
> Thanks,
> Shaun
> 
> 
> ______________________________________________________________________
> This message has been checked for all known viruses by MessageLabs Virus Scanning Service.
> ______________________________________________________________________
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> http://www.cluecon.com <http://www.cluecon.com/>
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> http://www.freeswitch.org <http://www.freeswitch.org/>
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From ksrigo at gmail.com  Thu Apr 19 00:05:19 2018
From: ksrigo at gmail.com (Srigo Kanapathipillai)
Date: Thu, 19 Apr 2018 00:05:19 +0000
Subject: [Freeswitch-users] Suddenly lost of fs_cli
In-Reply-To: <003c01d3d623$41492b30$c3db8190$@gmail.com>
References: <003c01d3d623$41492b30$c3db8190$@gmail.com>
Message-ID: <CAAiAxuxdddpvrivmzFt24Rdt04TB4kQKOm4dDV85AwMn3aWMuw@mail.gmail.com>

Probably your module event has been stopped.

Srigo

On Tue, 17 Apr 2018, 18:21 , <igor.potjevlesch at gmail.com> wrote:

> Hello,
>
>
>
> Since yesterday, I cannot access to fs_cli anymore after a traffic peak on
> an instance.
>
>
>
> I got the following error:
>
> [ERROR] fs_cli.c:1610 main() Error Connecting [Socket Connection Error]
>
> Usage: /usr/local/freeswitch/bin/fs_cli [-H <host>] [-P <port>] [-p
> <secret>] [-d <level>] [-x command] [-t <timeout_ms>] [profile]
>
>
>
> Is there something I can do without restarting Freeswitch?
>
>
>
> Regards,
>
>
>
> Igor.
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From shaun.stokes at itec-support.co.uk  Thu Apr 19 07:46:50 2018
From: shaun.stokes at itec-support.co.uk (Shaun Stokes)
Date: Thu, 19 Apr 2018 07:46:50 +0000
Subject: [Freeswitch-users] LUA menu on H264 device causes 500 Server
 Internal Error
In-Reply-To: <93A99129-2BC1-46C0-BA7E-759C73922F5C@jerris.com>
References: <CAPNTnnoQr0bsd2jw_fZ3p4vHFZKBt+6HPGAHCanh5AiU-cAP1g@mail.gmail.com>
 <CAAiAxuy5KqxD7Q6Q0FNZyMCw4+UbVUeq7qan__sPHWu5sJcJUg@mail.gmail.com>
 <1524060539124.73181@itec-support.co.uk>,
 <93A99129-2BC1-46C0-BA7E-759C73922F5C@jerris.com>
Message-ID: <1524124009442.76302@itec-support.co.uk>

Thanks Mike,


We've identified the problem, it is an issue on our end.


________________________________
From: FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> on behalf of Michael Jerris <mike at jerris.com>
Sent: 18 April 2018 21:58
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] LUA menu on H264 device causes 500 Server Internal Error

this full log of the call UNMODIFIED with a full sip trace may be able to help answer your questions.

On Apr 18, 2018, at 10:08 AM, Shaun Stokes <shaun.stokes at itec-support.co.uk<mailto:shaun.stokes at itec-support.co.uk>> wrote:


Hi All,

We have an issue when calling a LUA script with menu options from an H264 video enabled device the call drops part way into the call while navigating a menu due to '500 Server Internal Error'.

Trying to establish if this is a problem our end or a bug, has anyone else experienced this issue before?

Logs from an example below, our IP address is masked as x.x.x.x

nua.c:633 nua_invite() nua: nua_invite: entering
nua_stack.c:529 nua_signal() nua(0x7f25b00e8e10): sent signal r_invite
nua_stack.c:569 nua_stack_signal() nua(0x7f25b00e8e10): recv signal r_invite
nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering
soa.c:403 soa_set_params() soa_set_params(static::0x7f25b00f2ce0, ...) called
soa.c:1302 soa_init_offer_answer() soa_init_offer_answer(static::0x7f25b00f2ce0) called
nta.c:2665 nta_tpn_by_url() nta: selecting scheme sip
tport.c:4588 tport_by_name() tport(0x7f25b0005100): found 0x7f25b0147320 by name tls/x.x.x.x:7859
tport.c:3257 tport_tsend() tport_tsend(0x7f25b0147320) tpn = tls/x.x.x.x:7859
tport_type_tls.c:534 tport_tls_send() tport_tls_writevec: vec 0x7f25b00e40c0 0x7f25b001fe60 755 (755)
EXECUTE sofia/internal/2001 at domain sleep(2000)
tport_type_tls.c:534 tport_tls_send() tport_tls_writevec: vec 0x7f25b00e40c0 0x7f25b01ca460 9 (9)
tport.c:3594 tport_vsend() tport_vsend(0x7f25b0147320): 764 bytes of 764 to tls/x.x.x.x:7859
tport.c:3492 tport_send_msg() tport_vsend returned 764
tport.c:2296 tport_set_secondary_timer() tport(0x7f25b0147320): reset timer
nta.c:8304 outgoing_send() nta: sent INVITE (121693762) to tls/x.x.x.x:7859
tport.c:4160 tport_pend() tport_pend(0x7f25b0147320): pending 0x7f25b013b770 for tls/x.x.x.x:7859 (already 0)
nta.c:1350 set_timeout() nta: timer set to 32000 ms
nua_session.c:4145 signal_call_state_change() nua(0x7f25b00e8e10): ready call updated: calling
nua_stack.c:269 nua_stack_event() nua(0x7f25b00e8e10): event i_state INVITE sent
nua_stack.c:359 nua_application_event() nua: nua_application_event: entering
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
2018-04-18 14:54:13.139241 [DEBUG] sofia.c:7084 Channel sofia/internal/2001 at domain entering state [calling][0]
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
tport.c:2773 tport_wakeup() tport_wakeup(0x7f25b0147320): events IN
tport.c:2864 tport_recv_event() tport_recv_event(0x7f25b0147320)
tport_type_tls.c:434 tport_tls_recv() tport_tls_recv(0x7f25b0147320): tls_read() returned 473
tport.c:3205 tport_recv_iovec() tport_recv_iovec(0x7f25b0147320) msg 0x7f25b00d65f0 from (tls/x.x.x.x:7859) has 473 bytes, veclen = 1
tport.c:3023 tport_deliver() tport_deliver(0x7f25b0147320): msg 0x7f25b00d65f0 (473 bytes) from tls/x.x.x.x:7859/sips next=(nil)
nta.c:3299 agent_recv_response() nta: received 500 Server Internal Error for INVITE (121693762)
nta.c:3366 agent_recv_response() nta: 500 Server Internal Error is going to a transaction
nta.c:9564 outgoing_estimate_delay() nta_outgoing: RTT is 78.612 ms
tport.c:4222 tport_release() tport_release(0x7f25b0147320): 0x7f25b013b770 by 0x7f25b01d3eb0 with 0x7f25b00d65f0
tport.c:4588 tport_by_name() tport(0x7f25b0005100): found 0x7f25b0147320 by name tls/x.x.x.x:7859
tport.c:3257 tport_tsend() tport_tsend(0x7f25b0147320) tpn = tls/x.x.x.x:7859
tport_type_tls.c:534 tport_tls_send() tport_tls_writevec: vec 0x7f25b00e40c0 0x7f25b011b590 403 (403)
tport.c:3594 tport_vsend() tport_vsend(0x7f25b0147320): 403 bytes of 403 to tls/x.x.x.x:7859
tport.c:3492 tport_send_msg() tport_vsend returned 403
tport.c:2296 tport_set_secondary_timer() tport(0x7f25b0147320): reset timer
nta.c:8304 outgoing_send() nta: sent ACK (121693762) to tls/x.x.x.x:7859
nta.c:8722 outgoing_free() nta: outgoing_free(0x7f25b0146630)
nua_stack.c:271 nua_stack_event() nua(0x7f25b00e8e10): event r_invite 500 Server Internal Error
nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering
soa.c:1784 soa_terminate() soa_terminate(static::0x7f25b00f2ce0) called
nua_stack.c:359 nua_application_event() nua: nua_application_event: entering
soa.c:1302 soa_init_offer_answer() soa_init_offer_answer(static::0x7f25b00f2ce0) called
nta.c:2665 nta_tpn_by_url() nta: selecting scheme sip
tport.c:4588 tport_by_name() tport(0x7f25b0005100): found 0x7f25b0147320 by name tls/x.x.x.x:7859
tport.c:3257 tport_tsend() tport_tsend(0x7f25b0147320) tpn = tls/x.x.x.x:7859
tport_type_tls.c:534 tport_tls_send() tport_tls_writevec: vec 0x7f25b00e40c0 0x7f25b011b590 625 (625)
tport.c:3594 tport_vsend() tport_vsend(0x7f25b0147320): 625 bytes of 625 to tls/x.x.x.x:7859
tport.c:3492 tport_send_msg() tport_vsend returned 625
tport.c:2296 tport_set_secondary_timer() tport(0x7f25b0147320): reset timer
nta.c:8304 outgoing_send() nta: sent BYE (121693763) to tls/x.x.x.x:7859
tport.c:4160 tport_pend() tport_pend(0x7f25b0147320): pending 0x7f25b01128c0 for tls/x.x.x.x:7859 (already 0)
nua_session.c:4139 signal_call_state_change() nua(0x7f25b00e8e10): call state changed: ready -> terminating
nua_stack.c:271 nua_stack_event() nua(0x7f25b00e8e10): event i_state 500 Server Internal Error
nua_stack.c:359 nua_application_event() nua: nua_application_event: entering
tport.c:2296 tport_set_secondary_timer() tport(0x7f25b0147320): reset timer
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
2018-04-18 14:54:13.239253 [DEBUG] sofia.c:7084 Channel sofia/internal/2001 at domain entering state [terminating][500]
2018-04-18 14:54:13.239253 [NOTICE] sofia.c:8273 Hangup sofia/internal/2001 at domain [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE]
nua.c:342 nua_handle_bind() nua: nua_handle_bind: entering
nua.c:921 nua_handle_destroy() nua: nua_handle_destroy: entering
nua_stack.c:569 nua_stack_signal() nua(0x7f25b00e8e10): recv signal r_destroy
nua_dialog.c:397 nua_dialog_usage_remove_at() nua(0x7f25b00e8e10): removing session usage
nua_stack.c:529 nua_signal() nua(0x7f25b00e8e10): sent signal r_destroy
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
nua_session.c:4139 signal_call_state_change() nua(0x7f25b00e8e10): call state changed: terminating -> terminated
nua.c:342 nua_handle_bind() nua: nua_handle_bind: entering
nua.c:921 nua_handle_destroy() nua: nua_handle_destroy: entering
nua_stack.c:269 nua_stack_event() nua(0x7f25b00e8e10): event i_state Terminated
nua_stack.c:269 nua_stack_event() nua(0x7f25b00e8e10): event i_terminated Terminated
soa.c:356 soa_destroy() soa_destroy(static::0x7f25b00f2ce0) called
nta.c:4470 nta_leg_destroy() nta_leg_destroy(0x7f25b013afd0)
2018-04-18 14:54:13.239253 [DEBUG] switch_core_session.c:2815 sofia/internal/2001 at domain​ skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
2018-04-18 14:54:13.239253 [DEBUG] freeswitch_lua.cpp:382 DBH handle 0x7f256404f450 released.
2018-04-18 14:54:13.239253 [DEBUG] switch_cpp.cpp:1112 sofia/internal/2001 at domain destroy/unlink session from object
2018-04-18 14:54:13.239253 [DEBUG] switch_core_media.c:5970 sofia/internal/2001 at domain Video thread ended


Thanks,
Shaun


______________________________________________________________________
This message has been checked for all known viruses by MessageLabs Virus Scanning Service.
______________________________________________________________________
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com<http://www.freeswitchsolutions.com/>

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

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


______________________________________________________________________
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______________________________________________________________________

______________________________________________________________________
This message has been checked for all known viruses by  MessageLabs Virus Scanning Service.
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From mickael at winlux.fr  Thu Apr 19 08:31:52 2018
From: mickael at winlux.fr (Mickael Hubert)
Date: Thu, 19 Apr 2018 10:31:52 +0200
Subject: [Freeswitch-users] RTP reception before sending any RTP packet
Message-ID: <CA+fRVSWHki+k28uVTyFVHK1Z93YDGaB05LoZtStzcSVXBnraDg@mail.gmail.com>

Hi list,
I'd like to know if there's a way to tell FS to wait for RTP reception
before sending any RTP packet ?
We're facing a problem regarding early media with a PBX, NATed behind a
dumb router : actually, if FS first sends RTP to the PBX, without waiting
for RTP coming from the PBX, there's no audio. In some cases (related to
the provider we use behind FS to establish call to the PSTN) FS isn't the
'first-shooter' of the RTP flow, and everything works fine, but when FS is
the first to send RTP, it doesn't work...
Because we can't change anything at our providers configs, we'd like to
know if FS can "delay" RTP sending until it receives some RTP from the
calling party.

The schematic is following :
PBX ===> FS ===> ...voice backbone... ===> Providers
(FS is acting as an SBC)

Thanks for your help,
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From vma at vallimamod.org  Thu Apr 19 09:35:43 2018
From: vma at vallimamod.org (Vallimamod Abdullah)
Date: Thu, 19 Apr 2018 11:35:43 +0200
Subject: [Freeswitch-users] RTP reception before sending any RTP packet
In-Reply-To: <CA+fRVSWHki+k28uVTyFVHK1Z93YDGaB05LoZtStzcSVXBnraDg@mail.gmail.com>
References: <CA+fRVSWHki+k28uVTyFVHK1Z93YDGaB05LoZtStzcSVXBnraDg@mail.gmail.com>
Message-ID: <36896921-538C-4353-8920-C80FA763FC09@vallimamod.org>

Hi,

To my knowledge, it is the default beahviour for freeswitch to wait for incoming rtp from nated endpoints before starting to transmit.

You can explicitely force it with the profile param disable-rtp-auto-adjust=false.
You can also play with the channel variable rtp_auto_adjust_threshold which defines the number of rtp packets to wait for initially (default is 10 iirc)

Best Regards,
-- 
Vallimamod Abdullah
SIP Solutions
linkedin.com/in/vallimamod
.

> On 19 Apr 2018, at 10:31, Mickael Hubert <mickael at winlux.fr> wrote:
> 
> Hi list, 
> I'd like to know if there's a way to tell FS to wait for RTP reception before sending any RTP packet ? 
> We're facing a problem regarding early media with a PBX, NATed behind a dumb router : actually, if FS first sends RTP to the PBX, without waiting for RTP coming from the PBX, there's no audio. In some cases (related to the provider we use behind FS to establish call to the PSTN) FS isn't the 'first-shooter' of the RTP flow, and everything works fine, but when FS is the first to send RTP, it doesn't work... 
> Because we can't change anything at our providers configs, we'd like to know if FS can "delay" RTP sending until it receives some RTP from the calling party.
>  
> The schematic is following :
> PBX ===> FS ===> ...voice backbone... ===> Providers
> (FS is acting as an SBC)
>  
> Thanks for your help, 
> 



From infos at madovsky.org  Thu Apr 19 14:04:07 2018
From: infos at madovsky.org (Madovsky)
Date: Thu, 19 Apr 2018 07:04:07 -0700
Subject: [Freeswitch-users] Ekiga sotfphone codecs and FreeSWITCH
Message-ID: <d483e6f8-3ff4-7715-cd5a-d63505959825@madovsky.org>

Hi Folks,

just noticed that FS does not compare rightly Speex (maybe other codec) 
with Ekiga codecs

the log shows

Audio Codec Compare [SPEEX:93:16000:20:0:1]/[SPEEX:99:16000:20:42200:1]

although FS considers to not be the right codec.

is it because of the bitrate?


thanks

Franck



From lapa at novatec.de  Thu Apr 19 07:28:10 2018
From: lapa at novatec.de (Lars Paulsen)
Date: Thu, 19 Apr 2018 09:28:10 +0200
Subject: [Freeswitch-users] How to add headers to an outgoing REGISTER
Message-ID: <000801d3d7af$f81068f0$e8313ad0$@novatec.de>

Hello,

 

I have read the documentation under
https://wiki.freeswitch.org/wiki/Sofia-SIP#Adding_Request_Headers and as I
understood it is possible to add

Headers to SIP requests simply by changing the configuration.

 

But the documentation refers to the dialing plan.

I tried to add headers to an outgoing REGISTER but failed.

 

Because the documentation refers to the dialplan and changing headers for
existing calls, I am not sure if adding headers to a REGISTER request is
possible or not.

Can anyone please tell me if it is possible or not and if yes, how to
formulate the correct "<action application="set" .>" and "<condition>"
sections in the configuration.

 

Thanks in advance for all answers.

 

Best Regards,
Lars

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From abaci64 at gmail.com  Thu Apr 19 16:09:01 2018
From: abaci64 at gmail.com (Abaci B)
Date: Thu, 19 Apr 2018 12:09:01 -0400
Subject: [Freeswitch-users] Question about large confeerence
Message-ID: <CAFmZFiOZmdxtbYQJU9Ye1su0aRShqDWU+T1-j0iS32yy6pKeiw@mail.gmail.com>

Hi,
I'm trying to set up a large audio conference of over 1000 users where only
1 or 2 users are talking and the rest are just listening, but after 600-700
users the CPU usage gets too high and the audio gets choppy, I understand
that mod_conference is not optimized for my use case but was wondering if
someone has any suggestions on what can be done to optimize for my use case.
I was also thinking that maybe there is a  better alternative for what I
look for, such as having just the talkers on the conference and the
listeners should listen to it via mod_local_stream or maybe use something
like eavesdrop for the listeners but not sure if and how much performance I
would gain.
Thanks for any help or input
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From gmaruzz at gmail.com  Thu Apr 19 19:20:29 2018
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Thu, 19 Apr 2018 19:20:29 +0000
Subject: [Freeswitch-users] Question about large confeerence
In-Reply-To: <CAFmZFiOZmdxtbYQJU9Ye1su0aRShqDWU+T1-j0iS32yy6pKeiw@mail.gmail.com>
References: <CAFmZFiOZmdxtbYQJU9Ye1su0aRShqDWU+T1-j0iS32yy6pKeiw@mail.gmail.com>
Message-ID: <CALXCt0rS4Mfcm8ObL8nCmOnYbMkxor9FcQy0AzPkar5B+JnUWg@mail.gmail.com>

definitely conference with two participant, and 598 on localstream

-giovanni

On Thu, Apr 19, 2018, 18:38 Abaci B <abaci64 at gmail.com> wrote:

> Hi,
> I'm trying to set up a large audio conference of over 1000 users where
> only 1 or 2 users are talking and the rest are just listening, but after
> 600-700 users the CPU usage gets too high and the audio gets choppy, I
> understand that mod_conference is not optimized for my use case but was
> wondering if someone has any suggestions on what can be done to optimize
> for my use case.
> I was also thinking that maybe there is a  better alternative for what I
> look for, such as having just the talkers on the conference and the
> listeners should listen to it via mod_local_stream or maybe use something
> like eavesdrop for the listeners but not sure if and how much performance I
> would gain.
> Thanks for any help or input
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 abaci64 at gmail.com  Thu Apr 19 19:55:59 2018
From: abaci64 at gmail.com (Abaci B)
Date: Thu, 19 Apr 2018 15:55:59 -0400
Subject: [Freeswitch-users] Question about large confeerence
In-Reply-To: <CALXCt0rS4Mfcm8ObL8nCmOnYbMkxor9FcQy0AzPkar5B+JnUWg@mail.gmail.com>
References: <CAFmZFiOZmdxtbYQJU9Ye1su0aRShqDWU+T1-j0iS32yy6pKeiw@mail.gmail.com>
 <CALXCt0rS4Mfcm8ObL8nCmOnYbMkxor9FcQy0AzPkar5B+JnUWg@mail.gmail.com>
Message-ID: <CAFmZFiMBqsRuPwXYwPQho9fMR080XQRDoaPNcdkm-323YjTL_w@mail.gmail.com>

Thanks for the reply.
Another idea I'm thinking of is to add a LISTEN_ONLY mflag to
mod_conference and only run conference_loop_input thread on channels that
don't have this flag, that way I don't need to set up a shoutcast server
and have everything contained within freeswitch, does that make sense?

On Thu, Apr 19, 2018 at 3:20 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>
wrote:

> definitely conference with two participant, and 598 on localstream
>
> -giovanni
>
> On Thu, Apr 19, 2018, 18:38 Abaci B <abaci64 at gmail.com> wrote:
>
>> Hi,
>> I'm trying to set up a large audio conference of over 1000 users where
>> only 1 or 2 users are talking and the rest are just listening, but after
>> 600-700 users the CPU usage gets too high and the audio gets choppy, I
>> understand that mod_conference is not optimized for my use case but was
>> wondering if someone has any suggestions on what can be done to optimize
>> for my use case.
>> I was also thinking that maybe there is a  better alternative for what I
>> look for, such as having just the talkers on the conference and the
>> listeners should listen to it via mod_local_stream or maybe use something
>> like eavesdrop for the listeners but not sure if and how much performance I
>> would gain.
>> Thanks for any help or input
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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 steveayre at gmail.com  Thu Apr 19 21:19:06 2018
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 19 Apr 2018 22:19:06 +0100
Subject: [Freeswitch-users] Question about large confeerence
In-Reply-To: <CAFmZFiMBqsRuPwXYwPQho9fMR080XQRDoaPNcdkm-323YjTL_w@mail.gmail.com>
References: <CAFmZFiOZmdxtbYQJU9Ye1su0aRShqDWU+T1-j0iS32yy6pKeiw@mail.gmail.com>
 <CALXCt0rS4Mfcm8ObL8nCmOnYbMkxor9FcQy0AzPkar5B+JnUWg@mail.gmail.com>
 <CAFmZFiMBqsRuPwXYwPQho9fMR080XQRDoaPNcdkm-323YjTL_w@mail.gmail.com>
Message-ID: <CAFiqYukYkTkTM2vZ2e9iyZMwqn-fcQXOasVg0JwHHPp-fhiutw@mail.gmail.com>

There's already a muted flag you could use for that.

On 19 April 2018 at 20:55, Abaci B <abaci64 at gmail.com> wrote:

> Thanks for the reply.
> Another idea I'm thinking of is to add a LISTEN_ONLY mflag to
> mod_conference and only run conference_loop_input thread on channels that
> don't have this flag, that way I don't need to set up a shoutcast server
> and have everything contained within freeswitch, does that make sense?
>
> On Thu, Apr 19, 2018 at 3:20 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>
> wrote:
>
>> definitely conference with two participant, and 598 on localstream
>>
>> -giovanni
>>
>> On Thu, Apr 19, 2018, 18:38 Abaci B <abaci64 at gmail.com> wrote:
>>
>>> Hi,
>>> I'm trying to set up a large audio conference of over 1000 users where
>>> only 1 or 2 users are talking and the rest are just listening, but after
>>> 600-700 users the CPU usage gets too high and the audio gets choppy, I
>>> understand that mod_conference is not optimized for my use case but was
>>> wondering if someone has any suggestions on what can be done to optimize
>>> for my use case.
>>> I was also thinking that maybe there is a  better alternative for what I
>>> look for, such as having just the talkers on the conference and the
>>> listeners should listen to it via mod_local_stream or maybe use something
>>> like eavesdrop for the listeners but not sure if and how much performance I
>>> would gain.
>>> Thanks for any help or input
>>>
>>> ____________________________________________________________
>>> _____________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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 abaci64 at gmail.com  Thu Apr 19 21:20:46 2018
From: abaci64 at gmail.com (Abaci B)
Date: Thu, 19 Apr 2018 17:20:46 -0400
Subject: [Freeswitch-users] mod_conference scalability
In-Reply-To: <1261084008.6396.89.camel@local.freepabx.com>
References: <00aa01ca7e99$9901f9a0$cb05ece0$@com>
 <FDEDE676-A6DA-48BA-8F83-53ADFA5D40A3@jerris.com>
 <049601ca7f4f$37da5580$a78f0080$@com>
 <191c3a030912171142j24e64cb9r1f811d76db8a4918@mail.gmail.com>
 <04b501ca7f58$0a188870$1e499950$@com>
 <1261084008.6396.89.camel@local.freepabx.com>
Message-ID: <CAFmZFiMMhoFXok-OSqds3yt0TSv00TcsPk9_FP_egSMUrkzUzQ@mail.gmail.com>

Hi,
Came across this thread wile having the exact same issue, do you mind
sharing some info about your solution?
Thanks

On Thu, Dec 17, 2009 at 4:06 PM, David Knell <dave at 3c.co.uk> wrote:

> Hi Brian,
>
> I imagine that one of the issues is that you're using a complex
> sledgehammer (mod_conference) to crack a simple nut - that of having
> multiple listeners listening to a single speaker.
>
> As far as I am aware, FreeSWITCH doesn't have anything built in which
> will allow this kind of simple audio path switching - maybe someone more
> knowledgeable than me will correct me if I'm wrong?
>
> I presented some stuff at ClueCon which would address this kind of
> simple application and ought to scale well beyond what you've seen with
> FS or Asterisk.  It's still pretty basic [I'd do more with it if I
> wasn't so busy joshing with the other Brian on Facebook], and has never
> been deployed in anger but, if you're interested, drop me a note
> off-list.
>
> --Dave
>
> > I didn’t realize there was a policy about load testing questions. What
> > forum should I have used for this?
> >
> >
> >
> > I didn’t get the chance to test on FS trunk yet, but when I do I will
> > provide you with the feedback when I do. Just let me know what forum
> > to use for this topic from now on.
> >
> >
> >
> > Thanks,
> >
> >
> >
> > Brian.
> >
> >
> >
> > From: Anthony Minessale [mailto:anthony.minessale at gmail.com]
> > Sent: Thursday, December 17, 2009 2:42 PM
> > To: freeswitch-users at lists.freeswitch.org
> > Subject: Re: [Freeswitch-users] mod_conference scalability
> >
> >
> >
> >
> > One man's stable release is another man's 6 month old release with
> > hundreds of known fixed bugs.
> > If one of the core developers tells you to try it, you may as well
> > take the time to try it now that you have opened a forum questioning
> > the scalability.
> >
> > When you tested asterisk did you actually use 600 phones and verify
> > that each one can hear the audio perfectly and in time with what the
> > speaker was saying?  Did you try same on FS?
> >
> > Did you optimize your dialplan on FS to deal with a load test or
> > follow any of the recommended performance tuning page.
> >
> > All of the answers to these questions are really moot because we have
> > a policy against entertaining load testing questions but if you like
> > asterisk, by all means, use it, and good luck to you if those numbers
> > you are testing at are what you plan to put in real
> > production.........
> >
> >
> >
> > On Thu, Dec 17, 2009 at 1:29 PM, Brian <brian at proximosystems.com>
> > wrote:
> >
> > Hi Mike,
> >
> >
> >
> > I didn’t get around to testing on the FreeSWITCH trunk yet. Are there
> > substantial fixes to mod_conference in the FreeSWITCH trunk that might
> > increase capacity for my scenario of one speaker and many listeners?
> > If I want to put this into a production environment, I would need a
> > stable version, which as far as I know is the 1.0.4 version.
> >
> >
> >
> > However, I did test on Asterisk 1.4 using app_conference, and doing
> > the same scenario was able to get 1 speaker and 600 listeners on a
> > single conference with no audio issues. The CPU at that point was just
> > over 300%, same as where the single conference scenario failed on
> > FreeSWITCH with 300 listeners.  I was able to push it to over 700
> > listeners before I reached 400% CPU usage (I guess maxing out my
> > quad-core processors), and asterisk finally crashed. But up until that
> > point, there were no audio problems.
> >
> >
> >
> > I’ve read a lot about how FreeSWITCH is supposed to be more scalable
> > than Asterisk, but unless there is something wrong with my FreeSWITCH
> > setup, Asterisk was clearly the winner in this test – more than
> > doubling FreeSWITCH capacity in this case. Again, maybe there is
> > something on the FreeSWITCH side that I’m doing wrong, but I don’t see
> > what it could be.
> >
> >
> >
> > Brian.
> >
> >
> >
> >
> >
> > From: Michael Jerris [mailto:mike at jerris.com]
> > Sent: Thursday, December 17, 2009 10:18 AM
> > To: freeswitch-users at lists.freeswitch.org
> > Subject: Re: [Freeswitch-users] mod_conference scalability
> >
> >
> >
> >
> > I would be curious what the same tests produce with svn trunk of
> > FreeSWITCH.
> >
> >
> >
> >
> > Mike
> >
> >
> >
> >
> > On Dec 16, 2009, at 4:49 PM, Brian wrote:
> >
> >
> >
> >
> > Hi,
> >
> >
> >
> >
> >
> > I’m new to FreeSWITCH and I’m testing the scalability of
> > mod_conference to see if it will scale better that other solutions. My
> > scenario is to have one speaker, and many listeners (mute). Since I
> > have only one speaker, I was expecting this to scale well because
> > there is no audio mixing required, just send each frame of the single
> > speaker to each listener. Unfortunately, my testing was disappointing,
> > and it didn’t scale nearly as well as I’d hoped (based on what I’ve
> > read on how FreeSWITCH is supposed to be generally very scalable).
> >
> >
> >
> >
> >
> > Here’s my server setup is this:
> >
> >
> >
> >
> >
> > FreeSWITCH 1.0.4, 64 bit CentOS 5.3, on a quad-core Xeon server, 4 Gig
> > of RAM. I’ve set file logging to “notice” level. My conference profile
> > is configured to suppress several events, hoping that it would improve
> > performance.
> >
> >
> >
> >
> >
> > Here are a few scenarios I tested, and roughly where I reached the
> > point of audio failure on the conferences:
> >
> >
> >
> >
> >
> > Scenario 1:
> >
> >
> > 1 conference, 1 speaker, audio failed at approx 300 listeners (mute)
> >
> >
> >
> >
> >
> > Scenario 2:
> >
> >
> > 4 conferences, 1 speaker per conference, audio failed approx 110
> > listeners per conference (so just over 400 total channels on the
> > system).
> >
> >
> >
> >
> >
> > Scenario 3:
> >
> >
> > 16 conferences, 1 speaker per conference, audio failed at 32 listeners
> > per conference (so just over 500 total channels on the system).
> >
> >
> >
> >
> >
> >
> >
> >
> > Looking at the output from “top”, it seems that in all 3 scenarios,
> > the audio quality failed when the % CPU for the FreeSWITCH process
> > exceeded 300%.
> >
> >
> >
> >
> >
> > I was hoping maybe someone else might have done similar testing, or
> > maybe has suggestions on how to improve the performance. Or perhaps an
> > alternate solution to the one speaker, many listener case?
> >
> >
> >
> >
> >
> > Thanks,
> >
> >
> >
> >
> >
> > Brian.
> >
> >
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> >
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> >
> >
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> > Twitter: http://twitter.com/FreeSWITCH_wire
> >
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> > IRC: irc.freenode.net #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org
> > iax:guest at conference.freeswitch.org/888
> > googletalk:conf+888 at conference.freeswitch.org
> > pstn:+19193869900
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 abaci64 at gmail.com  Thu Apr 19 21:37:45 2018
From: abaci64 at gmail.com (Abaci B)
Date: Thu, 19 Apr 2018 17:37:45 -0400
Subject: [Freeswitch-users] Question about large confeerence
In-Reply-To: <CAFiqYukYkTkTM2vZ2e9iyZMwqn-fcQXOasVg0JwHHPp-fhiutw@mail.gmail.com>
References: <CAFmZFiOZmdxtbYQJU9Ye1su0aRShqDWU+T1-j0iS32yy6pKeiw@mail.gmail.com>
 <CALXCt0rS4Mfcm8ObL8nCmOnYbMkxor9FcQy0AzPkar5B+JnUWg@mail.gmail.com>
 <CAFmZFiMBqsRuPwXYwPQho9fMR080XQRDoaPNcdkm-323YjTL_w@mail.gmail.com>
 <CAFiqYukYkTkTM2vZ2e9iyZMwqn-fcQXOasVg0JwHHPp-fhiutw@mail.gmail.com>
Message-ID: <CAFmZFiMp05KUdkYSoSvd-tnGCrHgU56v6Ar5_rp6V2iLALYX_A@mail.gmail.com>

I was thinking that the muted flag (CAN_SPEAK?) is something that can
change mid conference which means there is a chance of loosing a single
frame (the last one right before muting). I was more interested in knowing
as far as performance if it makes sense.
not sure if this is safe but something like only starting
conference_loop_launch_input if that flag is not set.
I still wish there would be module that would do exactly this (something
like shoutcast but local and without the shoutcast latency)

On Thu, Apr 19, 2018 at 5:19 PM, Steven Ayre <steveayre at gmail.com> wrote:

> There's already a muted flag you could use for that.
>
> On 19 April 2018 at 20:55, Abaci B <abaci64 at gmail.com> wrote:
>
>> Thanks for the reply.
>> Another idea I'm thinking of is to add a LISTEN_ONLY mflag to
>> mod_conference and only run conference_loop_input thread on channels that
>> don't have this flag, that way I don't need to set up a shoutcast server
>> and have everything contained within freeswitch, does that make sense?
>>
>> On Thu, Apr 19, 2018 at 3:20 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>
>> wrote:
>>
>>> definitely conference with two participant, and 598 on localstream
>>>
>>> -giovanni
>>>
>>> On Thu, Apr 19, 2018, 18:38 Abaci B <abaci64 at gmail.com> wrote:
>>>
>>>> Hi,
>>>> I'm trying to set up a large audio conference of over 1000 users where
>>>> only 1 or 2 users are talking and the rest are just listening, but after
>>>> 600-700 users the CPU usage gets too high and the audio gets choppy, I
>>>> understand that mod_conference is not optimized for my use case but was
>>>> wondering if someone has any suggestions on what can be done to optimize
>>>> for my use case.
>>>> I was also thinking that maybe there is a  better alternative for what
>>>> I look for, such as having just the talkers on the conference and the
>>>> listeners should listen to it via mod_local_stream or maybe use something
>>>> like eavesdrop for the listeners but not sure if and how much performance I
>>>> would gain.
>>>> Thanks for any help or input
>>>>
>>>> ____________________________________________________________
>>>> _____________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free
>>>> switch-users
>>>> http://www.freeswitch.org
>>>
>>>
>>> ____________________________________________________________
>>> _____________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From ksrigo at gmail.com  Fri Apr 20 01:39:31 2018
From: ksrigo at gmail.com (Srigo Kanapathipillai)
Date: Fri, 20 Apr 2018 01:39:31 +0000
Subject: [Freeswitch-users] How to add headers to an outgoing REGISTER
In-Reply-To: <000801d3d7af$f81068f0$e8313ad0$@novatec.de>
References: <000801d3d7af$f81068f0$e8313ad0$@novatec.de>
Message-ID: <CAAiAxuz6VfiJR5rro946ruQPjhGCYSXQ0MMW8vLNXb=X-DV4tA@mail.gmail.com>

Hi,

What you mean by outgoing register?a Gateway registering?

If yes, I'm not sure you can do it.

Srigo

On Fri, 20 Apr 2018, 01:33 Lars Paulsen, <lapa at novatec.de> wrote:

> Hello,
>
>
>
> I have read the documentation under
> https://wiki.freeswitch.org/wiki/Sofia-SIP#Adding_Request_Headers and as
> I understood it is possible to add
>
> Headers to SIP requests simply by changing the configuration.
>
>
>
> But the documentation refers to the dialing plan.
>
> I tried to add headers to an outgoing REGISTER but failed.
>
>
>
> Because the documentation refers to the dialplan and changing headers for
> existing calls, I am not sure if adding headers to a REGISTER request is
> possible or not.
>
> Can anyone please tell me if it is possible or not and if yes, how to
> formulate the correct “<action application=”set” …>” and “<condition>”
> sections in the configuration.
>
>
>
> Thanks in advance for all answers.
>
>
>
> Best Regards,
> Lars
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 gmaruzz at gmail.com  Fri Apr 20 05:30:52 2018
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Fri, 20 Apr 2018 07:30:52 +0200
Subject: [Freeswitch-users] Question about large confeerence
In-Reply-To: <CAFmZFiMp05KUdkYSoSvd-tnGCrHgU56v6Ar5_rp6V2iLALYX_A@mail.gmail.com>
References: <CAFmZFiOZmdxtbYQJU9Ye1su0aRShqDWU+T1-j0iS32yy6pKeiw@mail.gmail.com>
 <CALXCt0rS4Mfcm8ObL8nCmOnYbMkxor9FcQy0AzPkar5B+JnUWg@mail.gmail.com>
 <CAFmZFiMBqsRuPwXYwPQho9fMR080XQRDoaPNcdkm-323YjTL_w@mail.gmail.com>
 <CAFiqYukYkTkTM2vZ2e9iyZMwqn-fcQXOasVg0JwHHPp-fhiutw@mail.gmail.com>
 <CAFmZFiMp05KUdkYSoSvd-tnGCrHgU56v6Ar5_rp6V2iLALYX_A@mail.gmail.com>
Message-ID: <CALXCt0q2Y_vD8tBtrBicYD0ki8uBNZpypuOj3-NjMmpbS03NDw@mail.gmail.com>

you definitely want to have only the speakers into conference, and stream
to all others.

-giovanni

On 19 April 2018 at 23:37, Abaci B <abaci64 at gmail.com> wrote:

> I was thinking that the muted flag (CAN_SPEAK?) is something that can
> change mid conference which means there is a chance of loosing a single
> frame (the last one right before muting). I was more interested in knowing
> as far as performance if it makes sense.
> not sure if this is safe but something like only starting
> conference_loop_launch_input if that flag is not set.
> I still wish there would be module that would do exactly this (something
> like shoutcast but local and without the shoutcast latency)
>
> On Thu, Apr 19, 2018 at 5:19 PM, Steven Ayre <steveayre at gmail.com> wrote:
>
>> There's already a muted flag you could use for that.
>>
>> On 19 April 2018 at 20:55, Abaci B <abaci64 at gmail.com> wrote:
>>
>>> Thanks for the reply.
>>> Another idea I'm thinking of is to add a LISTEN_ONLY mflag to
>>> mod_conference and only run conference_loop_input thread on channels that
>>> don't have this flag, that way I don't need to set up a shoutcast server
>>> and have everything contained within freeswitch, does that make sense?
>>>
>>> On Thu, Apr 19, 2018 at 3:20 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>
>>> wrote:
>>>
>>>> definitely conference with two participant, and 598 on localstream
>>>>
>>>> -giovanni
>>>>
>>>> On Thu, Apr 19, 2018, 18:38 Abaci B <abaci64 at gmail.com> wrote:
>>>>
>>>>> Hi,
>>>>> I'm trying to set up a large audio conference of over 1000 users where
>>>>> only 1 or 2 users are talking and the rest are just listening, but after
>>>>> 600-700 users the CPU usage gets too high and the audio gets choppy, I
>>>>> understand that mod_conference is not optimized for my use case but was
>>>>> wondering if someone has any suggestions on what can be done to optimize
>>>>> for my use case.
>>>>> I was also thinking that maybe there is a  better alternative for what
>>>>> I look for, such as having just the talkers on the conference and the
>>>>> listeners should listen to it via mod_local_stream or maybe use something
>>>>> like eavesdrop for the listeners but not sure if and how much performance I
>>>>> would gain.
>>>>> Thanks for any help or input
>>>>>
>>>>> ____________________________________________________________
>>>>> _____________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free
>>>>> switch-users
>>>>> http://www.freeswitch.org
>>>>
>>>>
>>>> ____________________________________________________________
>>>> _____________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free
>>>> switch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> ____________________________________________________________
>>> _____________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

Sincerely,

Giovanni Maruzzelli
OpenTelecom.IT
cell: +39 347 266 56 18
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From lapa at novatec.de  Fri Apr 20 06:29:38 2018
From: lapa at novatec.de (Lars Paulsen)
Date: Fri, 20 Apr 2018 08:29:38 +0200
Subject: [Freeswitch-users] How to add headers to an outgoing REGISTER
In-Reply-To: <CAAiAxuz6VfiJR5rro946ruQPjhGCYSXQ0MMW8vLNXb=X-DV4tA@mail.gmail.com>
References: <000801d3d7af$f81068f0$e8313ad0$@novatec.de>
 <CAAiAxuz6VfiJR5rro946ruQPjhGCYSXQ0MMW8vLNXb=X-DV4tA@mail.gmail.com>
Message-ID: <003601d3d870$f5342b50$df9c81f0$@novatec.de>

Hello Srigo,

 

yes, a gateway sending out a REGISTER. And I want to add headers to it before it’s send out.
I DO NOT mean adding headers to an REGISTER response send to a local extension.

 

To be precise I wanted to add “Require” and “Proxy-Require” headers.

But I failed on adding completely new headers.

 

Regards,
Lars

 

Von: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Srigo Kanapathipillai
Gesendet: Freitag, 20. April 2018 03:40
An: FreeSWITCH Users Help
Betreff: Re: [Freeswitch-users] How to add headers to an outgoing REGISTER

 

Hi,

 

What you mean by outgoing register?a Gateway registering? 

 

If yes, I'm not sure you can do it.

 

Srigo

 

On Fri, 20 Apr 2018, 01:33 Lars Paulsen, <lapa at novatec.de> wrote:

Hello,

 

I have read the documentation under https://wiki.freeswitch.org/wiki/Sofia-SIP#Adding_Request_Headers and as I understood it is possible to add

Headers to SIP requests simply by changing the configuration.

 

But the documentation refers to the dialing plan.

I tried to add headers to an outgoing REGISTER but failed.

 

Because the documentation refers to the dialplan and changing headers for existing calls, I am not sure if adding headers to a REGISTER request is possible or not.

Can anyone please tell me if it is possible or not and if yes, how to formulate the correct “<action application=”set” …>” and “<condition>” sections in the configuration.

 

Thanks in advance for all answers.

 

Best Regards,
Lars

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

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

FreeSWITCH-users mailing list
FreeSWITCH-users at 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 mickael at winlux.fr  Fri Apr 20 08:47:33 2018
From: mickael at winlux.fr (Mickael Hubert)
Date: Fri, 20 Apr 2018 10:47:33 +0200
Subject: [Freeswitch-users] RTP reception before sending any RTP packet
In-Reply-To: <36896921-538C-4353-8920-C80FA763FC09@vallimamod.org>
References: <CA+fRVSWHki+k28uVTyFVHK1Z93YDGaB05LoZtStzcSVXBnraDg@mail.gmail.com>
 <36896921-538C-4353-8920-C80FA763FC09@vallimamod.org>
Message-ID: <CA+fRVSXm0gWgmTQu5KNQ-76iT3aau4=0WZFGc5OXCv96E4V51g@mail.gmail.com>

Thanks for your answer Vallimamod,

You can see our dialplan bellow:
We tried these commands "rtp auto adjust, etc ...", but the issue continues.

is there a way to show if FS knows when the call is natted or not ? FYI we
have an opensips of front for SIP, but RTP is send directly to FS.

SIP call --> Opensips (nat detection) --> FS
RTP flow ------------------------------------------> FS

<condition field="${destinationAuthorized}" expression="^yes$"
break="never">
         <action application="export"
data="codec_string=${ep_codec_string}"/>
         <action application="export" data="sip_h_X-cid=${uuid}"/>
         <action application="set" data="hangup_after_bridge=true"/>
          <action application="set" data="ignore_early_media=false"/>
          <action application="set" data="ignore_display_updates=true"/>
          <action application="set" data="inherit_codec=true"/>
          <action application="set" data="fax_enable_t38=false"/>
          <action application="export" data="suppress_cng=true"/>
          <action application="set" data="disable_rtp_auto_adjust=false"/>
          <action application="set" data="rtp_auto_adjust_threshold=1"/>
          <action application="set"
data="rtp_manual_rtp_bugs=accept_any_packets"/>
          <action application="unset" data="sip_h_X-Username"/>
          <action application="bridge" data="
          {
              sip_invite_params=user=phone,
              sip_cid_type=none,
              ignore_display_updates=true,
               sip_contact_user=${caller_id_number}
          }
          sofia/internal/${destination_number}@${distributor(LISTINTERNAL
${sofia(profile internal gwlist down)})}"/>

Thanks in advance


2018-04-19 11:35 GMT+02:00 Vallimamod Abdullah <vma at vallimamod.org>:

> Hi,
>
> To my knowledge, it is the default beahviour for freeswitch to wait for
> incoming rtp from nated endpoints before starting to transmit.
>
> You can explicitely force it with the profile param
> disable-rtp-auto-adjust=false.
> You can also play with the channel variable rtp_auto_adjust_threshold
> which defines the number of rtp packets to wait for initially (default is
> 10 iirc)
>
> Best Regards,
> --
> Vallimamod Abdullah
> SIP Solutions
> linkedin.com/in/vallimamod
> .
>
> > On 19 Apr 2018, at 10:31, Mickael Hubert <mickael at winlux.fr> wrote:
> >
> > Hi list,
> > I'd like to know if there's a way to tell FS to wait for RTP reception
> before sending any RTP packet ?
> > We're facing a problem regarding early media with a PBX, NATed behind a
> dumb router : actually, if FS first sends RTP to the PBX, without waiting
> for RTP coming from the PBX, there's no audio. In some cases (related to
> the provider we use behind FS to establish call to the PSTN) FS isn't the
> 'first-shooter' of the RTP flow, and everything works fine, but when FS is
> the first to send RTP, it doesn't work...
> > Because we can't change anything at our providers configs, we'd like to
> know if FS can "delay" RTP sending until it receives some RTP from the
> calling party.
> >
> > The schematic is following :
> > PBX ===> FS ===> ...voice backbone... ===> Providers
> > (FS is acting as an SBC)
> >
> > Thanks for your help,
> >
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From abaci64 at gmail.com  Fri Apr 20 12:43:49 2018
From: abaci64 at gmail.com (Abaci B)
Date: Fri, 20 Apr 2018 08:43:49 -0400
Subject: [Freeswitch-users] Question about large confeerence
In-Reply-To: <CALXCt0q2Y_vD8tBtrBicYD0ki8uBNZpypuOj3-NjMmpbS03NDw@mail.gmail.com>
References: <CAFmZFiOZmdxtbYQJU9Ye1su0aRShqDWU+T1-j0iS32yy6pKeiw@mail.gmail.com>
 <CALXCt0rS4Mfcm8ObL8nCmOnYbMkxor9FcQy0AzPkar5B+JnUWg@mail.gmail.com>
 <CAFmZFiMBqsRuPwXYwPQho9fMR080XQRDoaPNcdkm-323YjTL_w@mail.gmail.com>
 <CAFiqYukYkTkTM2vZ2e9iyZMwqn-fcQXOasVg0JwHHPp-fhiutw@mail.gmail.com>
 <CAFmZFiMp05KUdkYSoSvd-tnGCrHgU56v6Ar5_rp6V2iLALYX_A@mail.gmail.com>
 <CALXCt0q2Y_vD8tBtrBicYD0ki8uBNZpypuOj3-NjMmpbS03NDw@mail.gmail.com>
Message-ID: <CAFmZFiP+c+iiNN6mQNo3pxVwnG3pGCy39KDvoGrexw-NOxzkBQ@mail.gmail.com>

problem with shoutcast streaming is the extreme latency

On Fri, Apr 20, 2018 at 1:30 AM, Giovanni Maruzzelli <gmaruzz at gmail.com>
wrote:

> you definitely want to have only the speakers into conference, and stream
> to all others.
>
> -giovanni
>
> On 19 April 2018 at 23:37, Abaci B <abaci64 at gmail.com> wrote:
>
>> I was thinking that the muted flag (CAN_SPEAK?) is something that can
>> change mid conference which means there is a chance of loosing a single
>> frame (the last one right before muting). I was more interested in knowing
>> as far as performance if it makes sense.
>> not sure if this is safe but something like only starting
>> conference_loop_launch_input if that flag is not set.
>> I still wish there would be module that would do exactly this (something
>> like shoutcast but local and without the shoutcast latency)
>>
>> On Thu, Apr 19, 2018 at 5:19 PM, Steven Ayre <steveayre at gmail.com> wrote:
>>
>>> There's already a muted flag you could use for that.
>>>
>>> On 19 April 2018 at 20:55, Abaci B <abaci64 at gmail.com> wrote:
>>>
>>>> Thanks for the reply.
>>>> Another idea I'm thinking of is to add a LISTEN_ONLY mflag to
>>>> mod_conference and only run conference_loop_input thread on channels that
>>>> don't have this flag, that way I don't need to set up a shoutcast server
>>>> and have everything contained within freeswitch, does that make sense?
>>>>
>>>> On Thu, Apr 19, 2018 at 3:20 PM, Giovanni Maruzzelli <gmaruzz at gmail.com
>>>> > wrote:
>>>>
>>>>> definitely conference with two participant, and 598 on localstream
>>>>>
>>>>> -giovanni
>>>>>
>>>>> On Thu, Apr 19, 2018, 18:38 Abaci B <abaci64 at gmail.com> wrote:
>>>>>
>>>>>> Hi,
>>>>>> I'm trying to set up a large audio conference of over 1000 users
>>>>>> where only 1 or 2 users are talking and the rest are just listening, but
>>>>>> after 600-700 users the CPU usage gets too high and the audio gets choppy,
>>>>>> I understand that mod_conference is not optimized for my use case but was
>>>>>> wondering if someone has any suggestions on what can be done to optimize
>>>>>> for my use case.
>>>>>> I was also thinking that maybe there is a  better alternative for
>>>>>> what I look for, such as having just the talkers on the conference and the
>>>>>> listeners should listen to it via mod_local_stream or maybe use something
>>>>>> like eavesdrop for the listeners but not sure if and how much performance I
>>>>>> would gain.
>>>>>> Thanks for any help or input
>>>>>>
>>>>>> ____________________________________________________________
>>>>>> _____________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://confluence.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free
>>>>>> switch-users
>>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>> ____________________________________________________________
>>>>> _____________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free
>>>>> switch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>> ____________________________________________________________
>>>> _____________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free
>>>> switch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> ____________________________________________________________
>>> _____________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
>
> Sincerely,
>
> Giovanni Maruzzelli
> OpenTelecom.IT
> cell: +39 347 266 56 18
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From tculjaga at gmail.com  Fri Apr 20 17:15:31 2018
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Fri, 20 Apr 2018 19:15:31 +0200
Subject: [Freeswitch-users] Question about large confeerence
In-Reply-To: <CAFmZFiP+c+iiNN6mQNo3pxVwnG3pGCy39KDvoGrexw-NOxzkBQ@mail.gmail.com>
References: <CAFmZFiOZmdxtbYQJU9Ye1su0aRShqDWU+T1-j0iS32yy6pKeiw@mail.gmail.com>
 <CALXCt0rS4Mfcm8ObL8nCmOnYbMkxor9FcQy0AzPkar5B+JnUWg@mail.gmail.com>
 <CAFmZFiMBqsRuPwXYwPQho9fMR080XQRDoaPNcdkm-323YjTL_w@mail.gmail.com>
 <CAFiqYukYkTkTM2vZ2e9iyZMwqn-fcQXOasVg0JwHHPp-fhiutw@mail.gmail.com>
 <CAFmZFiMp05KUdkYSoSvd-tnGCrHgU56v6Ar5_rp6V2iLALYX_A@mail.gmail.com>
 <CALXCt0q2Y_vD8tBtrBicYD0ki8uBNZpypuOj3-NjMmpbS03NDw@mail.gmail.com>
 <CAFmZFiP+c+iiNN6mQNo3pxVwnG3pGCy39KDvoGrexw-NOxzkBQ@mail.gmail.com>
Message-ID: <CAJoKKdVhz7h4Y6ksNz36=8W96BWjKFzOOmK22md3v6f4s9FRCg@mail.gmail.com>

a conference with 1000 members demands huge resources... and this is not
efficient if only 2-5% of members will actually talk.
you need to keep the actual conference to the minimum number of "speakers"
and move listeners to a streaming "service".

so, focus on how to get your entire conference streamed somewhere.

another thing to consider.... do the "listeners" really need to be in time
sync with the conference ? They will just hear the conversation from the
conference a bit later... thats all.

T.






On 20 April 2018 at 14:43, Abaci B <abaci64 at gmail.com> wrote:

> problem with shoutcast streaming is the extreme latency
>
> On Fri, Apr 20, 2018 at 1:30 AM, Giovanni Maruzzelli <gmaruzz at gmail.com>
> wrote:
>
>> you definitely want to have only the speakers into conference, and stream
>> to all others.
>>
>> -giovanni
>>
>> On 19 April 2018 at 23:37, Abaci B <abaci64 at gmail.com> wrote:
>>
>>> I was thinking that the muted flag (CAN_SPEAK?) is something that can
>>> change mid conference which means there is a chance of loosing a single
>>> frame (the last one right before muting). I was more interested in knowing
>>> as far as performance if it makes sense.
>>> not sure if this is safe but something like only starting
>>> conference_loop_launch_input if that flag is not set.
>>> I still wish there would be module that would do exactly this (something
>>> like shoutcast but local and without the shoutcast latency)
>>>
>>> On Thu, Apr 19, 2018 at 5:19 PM, Steven Ayre <steveayre at gmail.com>
>>> wrote:
>>>
>>>> There's already a muted flag you could use for that.
>>>>
>>>> On 19 April 2018 at 20:55, Abaci B <abaci64 at gmail.com> wrote:
>>>>
>>>>> Thanks for the reply.
>>>>> Another idea I'm thinking of is to add a LISTEN_ONLY mflag to
>>>>> mod_conference and only run conference_loop_input thread on channels that
>>>>> don't have this flag, that way I don't need to set up a shoutcast server
>>>>> and have everything contained within freeswitch, does that make sense?
>>>>>
>>>>> On Thu, Apr 19, 2018 at 3:20 PM, Giovanni Maruzzelli <
>>>>> gmaruzz at gmail.com> wrote:
>>>>>
>>>>>> definitely conference with two participant, and 598 on localstream
>>>>>>
>>>>>> -giovanni
>>>>>>
>>>>>> On Thu, Apr 19, 2018, 18:38 Abaci B <abaci64 at gmail.com> wrote:
>>>>>>
>>>>>>> Hi,
>>>>>>> I'm trying to set up a large audio conference of over 1000 users
>>>>>>> where only 1 or 2 users are talking and the rest are just listening, but
>>>>>>> after 600-700 users the CPU usage gets too high and the audio gets choppy,
>>>>>>> I understand that mod_conference is not optimized for my use case but was
>>>>>>> wondering if someone has any suggestions on what can be done to optimize
>>>>>>> for my use case.
>>>>>>> I was also thinking that maybe there is a  better alternative for
>>>>>>> what I look for, such as having just the talkers on the conference and the
>>>>>>> listeners should listen to it via mod_local_stream or maybe use something
>>>>>>> like eavesdrop for the listeners but not sure if and how much performance I
>>>>>>> would gain.
>>>>>>> Thanks for any help or input
>>>>>>>
>>>>>>> ____________________________________________________________
>>>>>>> _____________
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>
>>>>>>> Official FreeSWITCH Sites
>>>>>>> http://www.freeswitch.org
>>>>>>> http://confluence.freeswitch.org
>>>>>>> http://www.cluecon.com
>>>>>>>
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free
>>>>>>> switch-users
>>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>> ____________________________________________________________
>>>>>> _____________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://confluence.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free
>>>>>> switch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>> ____________________________________________________________
>>>>> _____________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free
>>>>> switch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>> ____________________________________________________________
>>>> _____________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free
>>>> switch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> ____________________________________________________________
>>> _____________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>>
>> Sincerely,
>>
>> Giovanni Maruzzelli
>> OpenTelecom.IT
>> cell: +39 347 266 56 18
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From abaci64 at gmail.com  Fri Apr 20 17:29:20 2018
From: abaci64 at gmail.com (Abaci B)
Date: Fri, 20 Apr 2018 13:29:20 -0400
Subject: [Freeswitch-users] Question about large confeerence
In-Reply-To: <CAJoKKdVhz7h4Y6ksNz36=8W96BWjKFzOOmK22md3v6f4s9FRCg@mail.gmail.com>
References: <CAFmZFiOZmdxtbYQJU9Ye1su0aRShqDWU+T1-j0iS32yy6pKeiw@mail.gmail.com>
 <CALXCt0rS4Mfcm8ObL8nCmOnYbMkxor9FcQy0AzPkar5B+JnUWg@mail.gmail.com>
 <CAFmZFiMBqsRuPwXYwPQho9fMR080XQRDoaPNcdkm-323YjTL_w@mail.gmail.com>
 <CAFiqYukYkTkTM2vZ2e9iyZMwqn-fcQXOasVg0JwHHPp-fhiutw@mail.gmail.com>
 <CAFmZFiMp05KUdkYSoSvd-tnGCrHgU56v6Ar5_rp6V2iLALYX_A@mail.gmail.com>
 <CALXCt0q2Y_vD8tBtrBicYD0ki8uBNZpypuOj3-NjMmpbS03NDw@mail.gmail.com>
 <CAFmZFiP+c+iiNN6mQNo3pxVwnG3pGCy39KDvoGrexw-NOxzkBQ@mail.gmail.com>
 <CAJoKKdVhz7h4Y6ksNz36=8W96BWjKFzOOmK22md3v6f4s9FRCg@mail.gmail.com>
Message-ID: <CAFmZFiOxjqTKQs0uKHXZ4AUHR6qyKCVoE_H93xw8A9fpX+ciLg@mail.gmail.com>

I do understand that the way mod_conference work it is not efficient for my
use case, my question is about the best alternative.
Streaming would be ideal if it wouldn't be for the latency, since users
need to be able to switch to talk mode there is no way a latency of 5-10
seconds (or possibly more) is acceptable.
Another option is to use eavesdrop which I plan on testing next week to see
if it scales, if someone has any input on it I would appreciate.
The next option would be to optimize mod_conference for this use case, I
started looking at the code and identified some places that performance can
be optimized but may need to hire someone with better C skills to do the
actual coding.
another possibility may be, to use an RTP proxy that would just distribute
the RTP to all callers, this would give the best performance but would
limit features.
Last option would be to develop a new module optimized for low latency
streaming, but leave that as a last resort.

On Fri, Apr 20, 2018 at 1:15 PM, Tihomir Culjaga <tculjaga at gmail.com> wrote:

> a conference with 1000 members demands huge resources... and this is not
> efficient if only 2-5% of members will actually talk.
> you need to keep the actual conference to the minimum number of "speakers"
> and move listeners to a streaming "service".
>
> so, focus on how to get your entire conference streamed somewhere.
>
> another thing to consider.... do the "listeners" really need to be in time
> sync with the conference ? They will just hear the conversation from the
> conference a bit later... thats all.
>
> T.
>
>
>
>
>
>
> On 20 April 2018 at 14:43, Abaci B <abaci64 at gmail.com> wrote:
>
>> problem with shoutcast streaming is the extreme latency
>>
>> On Fri, Apr 20, 2018 at 1:30 AM, Giovanni Maruzzelli <gmaruzz at gmail.com>
>> wrote:
>>
>>> you definitely want to have only the speakers into conference, and
>>> stream to all others.
>>>
>>> -giovanni
>>>
>>> On 19 April 2018 at 23:37, Abaci B <abaci64 at gmail.com> wrote:
>>>
>>>> I was thinking that the muted flag (CAN_SPEAK?) is something that can
>>>> change mid conference which means there is a chance of loosing a single
>>>> frame (the last one right before muting). I was more interested in knowing
>>>> as far as performance if it makes sense.
>>>> not sure if this is safe but something like only starting
>>>> conference_loop_launch_input if that flag is not set.
>>>> I still wish there would be module that would do exactly this
>>>> (something like shoutcast but local and without the shoutcast latency)
>>>>
>>>> On Thu, Apr 19, 2018 at 5:19 PM, Steven Ayre <steveayre at gmail.com>
>>>> wrote:
>>>>
>>>>> There's already a muted flag you could use for that.
>>>>>
>>>>> On 19 April 2018 at 20:55, Abaci B <abaci64 at gmail.com> wrote:
>>>>>
>>>>>> Thanks for the reply.
>>>>>> Another idea I'm thinking of is to add a LISTEN_ONLY mflag to
>>>>>> mod_conference and only run conference_loop_input thread on channels that
>>>>>> don't have this flag, that way I don't need to set up a shoutcast server
>>>>>> and have everything contained within freeswitch, does that make sense?
>>>>>>
>>>>>> On Thu, Apr 19, 2018 at 3:20 PM, Giovanni Maruzzelli <
>>>>>> gmaruzz at gmail.com> wrote:
>>>>>>
>>>>>>> definitely conference with two participant, and 598 on localstream
>>>>>>>
>>>>>>> -giovanni
>>>>>>>
>>>>>>> On Thu, Apr 19, 2018, 18:38 Abaci B <abaci64 at gmail.com> wrote:
>>>>>>>
>>>>>>>> Hi,
>>>>>>>> I'm trying to set up a large audio conference of over 1000 users
>>>>>>>> where only 1 or 2 users are talking and the rest are just listening, but
>>>>>>>> after 600-700 users the CPU usage gets too high and the audio gets choppy,
>>>>>>>> I understand that mod_conference is not optimized for my use case but was
>>>>>>>> wondering if someone has any suggestions on what can be done to optimize
>>>>>>>> for my use case.
>>>>>>>> I was also thinking that maybe there is a  better alternative for
>>>>>>>> what I look for, such as having just the talkers on the conference and the
>>>>>>>> listeners should listen to it via mod_local_stream or maybe use something
>>>>>>>> like eavesdrop for the listeners but not sure if and how much performance I
>>>>>>>> would gain.
>>>>>>>> Thanks for any help or input
>>>>>>>>
>>>>>>>> ____________________________________________________________
>>>>>>>> _____________
>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>> consulting at freeswitch.org
>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>
>>>>>>>> Official FreeSWITCH Sites
>>>>>>>> http://www.freeswitch.org
>>>>>>>> http://confluence.freeswitch.org
>>>>>>>> http://www.cluecon.com
>>>>>>>>
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free
>>>>>>>> switch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>> ____________________________________________________________
>>>>>>> _____________
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>
>>>>>>> Official FreeSWITCH Sites
>>>>>>> http://www.freeswitch.org
>>>>>>> http://confluence.freeswitch.org
>>>>>>> http://www.cluecon.com
>>>>>>>
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free
>>>>>>> switch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>
>>>>>>
>>>>>> ____________________________________________________________
>>>>>> _____________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://confluence.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free
>>>>>> switch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>> ____________________________________________________________
>>>>> _____________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free
>>>>> switch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>> ____________________________________________________________
>>>> _____________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free
>>>> switch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> --
>>>
>>> Sincerely,
>>>
>>> Giovanni Maruzzelli
>>> OpenTelecom.IT
>>> cell: +39 347 266 56 18
>>>
>>> ____________________________________________________________
>>> _____________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/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 hartnett.tom at gmail.com  Fri Apr 20 17:42:13 2018
From: hartnett.tom at gmail.com (Tom Hartnett)
Date: Fri, 20 Apr 2018 13:42:13 -0400
Subject: [Freeswitch-users] Question about large confeerence
In-Reply-To: <CAFmZFiOxjqTKQs0uKHXZ4AUHR6qyKCVoE_H93xw8A9fpX+ciLg@mail.gmail.com>
References: <CAFmZFiOZmdxtbYQJU9Ye1su0aRShqDWU+T1-j0iS32yy6pKeiw@mail.gmail.com>
 <CALXCt0rS4Mfcm8ObL8nCmOnYbMkxor9FcQy0AzPkar5B+JnUWg@mail.gmail.com>
 <CAFmZFiMBqsRuPwXYwPQho9fMR080XQRDoaPNcdkm-323YjTL_w@mail.gmail.com>
 <CAFiqYukYkTkTM2vZ2e9iyZMwqn-fcQXOasVg0JwHHPp-fhiutw@mail.gmail.com>
 <CAFmZFiMp05KUdkYSoSvd-tnGCrHgU56v6Ar5_rp6V2iLALYX_A@mail.gmail.com>
 <CALXCt0q2Y_vD8tBtrBicYD0ki8uBNZpypuOj3-NjMmpbS03NDw@mail.gmail.com>
 <CAFmZFiP+c+iiNN6mQNo3pxVwnG3pGCy39KDvoGrexw-NOxzkBQ@mail.gmail.com>
 <CAJoKKdVhz7h4Y6ksNz36=8W96BWjKFzOOmK22md3v6f4s9FRCg@mail.gmail.com>
 <CAFmZFiOxjqTKQs0uKHXZ4AUHR6qyKCVoE_H93xw8A9fpX+ciLg@mail.gmail.com>
Message-ID: <CAPWmf4u8a6x8VEELAM0WRjnMKE6gapWbhvwfSmR7JtNp+LdH0Q@mail.gmail.com>

This won't help much but here's our experience.
Had a similar requirement and posted here last year about it. Luckily my
needs were for fewer listeners (30-50) and running them all in
mod_conference (with listeners muted) was feasible. But it's not elegant
(if it's indeed the case) that all listeners have a jitter buffer and audio
decoder that's being allocated but not being used when muted.
The upside is we could implement a "barge in" function to allow listeners
to join the conference via DTMF PIN. For our application it's not a risk
and it's a neat feature.

On Fri, Apr 20, 2018 at 1:29 PM, Abaci B <abaci64 at gmail.com> wrote:

> I do understand that the way mod_conference work it is not efficient for
> my use case, my question is about the best alternative.
> Streaming would be ideal if it wouldn't be for the latency, since users
> need to be able to switch to talk mode there is no way a latency of 5-10
> seconds (or possibly more) is acceptable.
> Another option is to use eavesdrop which I plan on testing next week to
> see if it scales, if someone has any input on it I would appreciate.
> The next option would be to optimize mod_conference for this use case, I
> started looking at the code and identified some places that performance can
> be optimized but may need to hire someone with better C skills to do the
> actual coding.
> another possibility may be, to use an RTP proxy that would just distribute
> the RTP to all callers, this would give the best performance but would
> limit features.
> Last option would be to develop a new module optimized for low latency
> streaming, but leave that as a last resort.
>
> On Fri, Apr 20, 2018 at 1:15 PM, Tihomir Culjaga <tculjaga at gmail.com>
> wrote:
>
>> a conference with 1000 members demands huge resources... and this is not
>> efficient if only 2-5% of members will actually talk.
>> you need to keep the actual conference to the minimum number of
>> "speakers" and move listeners to a streaming "service".
>>
>> so, focus on how to get your entire conference streamed somewhere.
>>
>> another thing to consider.... do the "listeners" really need to be in
>> time sync with the conference ? They will just hear the conversation from
>> the conference a bit later... thats all.
>>
>> T.
>>
>>
>>
>>
>>
>>
>> On 20 April 2018 at 14:43, Abaci B <abaci64 at gmail.com> wrote:
>>
>>> problem with shoutcast streaming is the extreme latency
>>>
>>> On Fri, Apr 20, 2018 at 1:30 AM, Giovanni Maruzzelli <gmaruzz at gmail.com>
>>> wrote:
>>>
>>>> you definitely want to have only the speakers into conference, and
>>>> stream to all others.
>>>>
>>>> -giovanni
>>>>
>>>> On 19 April 2018 at 23:37, Abaci B <abaci64 at gmail.com> wrote:
>>>>
>>>>> I was thinking that the muted flag (CAN_SPEAK?) is something that can
>>>>> change mid conference which means there is a chance of loosing a single
>>>>> frame (the last one right before muting). I was more interested in knowing
>>>>> as far as performance if it makes sense.
>>>>> not sure if this is safe but something like only starting
>>>>> conference_loop_launch_input if that flag is not set.
>>>>> I still wish there would be module that would do exactly this
>>>>> (something like shoutcast but local and without the shoutcast latency)
>>>>>
>>>>> On Thu, Apr 19, 2018 at 5:19 PM, Steven Ayre <steveayre at gmail.com>
>>>>> wrote:
>>>>>
>>>>>> There's already a muted flag you could use for that.
>>>>>>
>>>>>> On 19 April 2018 at 20:55, Abaci B <abaci64 at gmail.com> wrote:
>>>>>>
>>>>>>> Thanks for the reply.
>>>>>>> Another idea I'm thinking of is to add a LISTEN_ONLY mflag to
>>>>>>> mod_conference and only run conference_loop_input thread on channels that
>>>>>>> don't have this flag, that way I don't need to set up a shoutcast server
>>>>>>> and have everything contained within freeswitch, does that make sense?
>>>>>>>
>>>>>>> On Thu, Apr 19, 2018 at 3:20 PM, Giovanni Maruzzelli <
>>>>>>> gmaruzz at gmail.com> wrote:
>>>>>>>
>>>>>>>> definitely conference with two participant, and 598 on localstream
>>>>>>>>
>>>>>>>> -giovanni
>>>>>>>>
>>>>>>>> On Thu, Apr 19, 2018, 18:38 Abaci B <abaci64 at gmail.com> wrote:
>>>>>>>>
>>>>>>>>> Hi,
>>>>>>>>> I'm trying to set up a large audio conference of over 1000 users
>>>>>>>>> where only 1 or 2 users are talking and the rest are just listening, but
>>>>>>>>> after 600-700 users the CPU usage gets too high and the audio gets choppy,
>>>>>>>>> I understand that mod_conference is not optimized for my use case but was
>>>>>>>>> wondering if someone has any suggestions on what can be done to optimize
>>>>>>>>> for my use case.
>>>>>>>>> I was also thinking that maybe there is a  better alternative for
>>>>>>>>> what I look for, such as having just the talkers on the conference and the
>>>>>>>>> listeners should listen to it via mod_local_stream or maybe use something
>>>>>>>>> like eavesdrop for the listeners but not sure if and how much performance I
>>>>>>>>> would gain.
>>>>>>>>> Thanks for any help or input
>>>>>>>>>
>>>>>>>>> ____________________________________________________________
>>>>>>>>> _____________
>>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>>> consulting at freeswitch.org
>>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>>
>>>>>>>>> Official FreeSWITCH Sites
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>> http://confluence.freeswitch.org
>>>>>>>>> http://www.cluecon.com
>>>>>>>>>
>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free
>>>>>>>>> switch-users
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>>
>>>>>>>> ____________________________________________________________
>>>>>>>> _____________
>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>> consulting at freeswitch.org
>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>
>>>>>>>> Official FreeSWITCH Sites
>>>>>>>> http://www.freeswitch.org
>>>>>>>> http://confluence.freeswitch.org
>>>>>>>> http://www.cluecon.com
>>>>>>>>
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free
>>>>>>>> switch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> ____________________________________________________________
>>>>>>> _____________
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>
>>>>>>> Official FreeSWITCH Sites
>>>>>>> http://www.freeswitch.org
>>>>>>> http://confluence.freeswitch.org
>>>>>>> http://www.cluecon.com
>>>>>>>
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free
>>>>>>> switch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>
>>>>>>
>>>>>> ____________________________________________________________
>>>>>> _____________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://confluence.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free
>>>>>> switch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>> ____________________________________________________________
>>>>> _____________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free
>>>>> switch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>>
>>>> Sincerely,
>>>>
>>>> Giovanni Maruzzelli
>>>> OpenTelecom.IT
>>>> cell: +39 347 266 56 18
>>>>
>>>> ____________________________________________________________
>>>> _____________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free
>>>> switch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> ____________________________________________________________
>>> _____________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 infos at madovsky.org  Fri Apr 20 19:45:38 2018
From: infos at madovsky.org (Madovsky)
Date: Fri, 20 Apr 2018 12:45:38 -0700
Subject: [Freeswitch-users] Question about large confeerence
In-Reply-To: <CAFmZFiOxjqTKQs0uKHXZ4AUHR6qyKCVoE_H93xw8A9fpX+ciLg@mail.gmail.com>
References: <CAFmZFiOZmdxtbYQJU9Ye1su0aRShqDWU+T1-j0iS32yy6pKeiw@mail.gmail.com>
 <CALXCt0rS4Mfcm8ObL8nCmOnYbMkxor9FcQy0AzPkar5B+JnUWg@mail.gmail.com>
 <CAFmZFiMBqsRuPwXYwPQho9fMR080XQRDoaPNcdkm-323YjTL_w@mail.gmail.com>
 <CAFiqYukYkTkTM2vZ2e9iyZMwqn-fcQXOasVg0JwHHPp-fhiutw@mail.gmail.com>
 <CAFmZFiMp05KUdkYSoSvd-tnGCrHgU56v6Ar5_rp6V2iLALYX_A@mail.gmail.com>
 <CALXCt0q2Y_vD8tBtrBicYD0ki8uBNZpypuOj3-NjMmpbS03NDw@mail.gmail.com>
 <CAFmZFiP+c+iiNN6mQNo3pxVwnG3pGCy39KDvoGrexw-NOxzkBQ@mail.gmail.com>
 <CAJoKKdVhz7h4Y6ksNz36=8W96BWjKFzOOmK22md3v6f4s9FRCg@mail.gmail.com>
 <CAFmZFiOxjqTKQs0uKHXZ4AUHR6qyKCVoE_H93xw8A9fpX+ciLg@mail.gmail.com>
Message-ID: <5849f54a-8603-7818-2aa6-4dce06cd7743@madovsky.org>

Latency is often caused by buffer and high quality video.

if you really need zero latency so find a way to set the buffer stream 
to zero and use

a very fast video encoding.

On 4/20/2018 10:29 AM, Abaci B wrote:
> I do understand that the way mod_conference work it is not efficient 
> for my use case, my question is about the best alternative.
> Streaming would be ideal if it wouldn't be for the latency, since 
> users need to be able to switch to talk mode there is no way a latency 
> of 5-10 seconds (or possibly more) is acceptable.
> Another option is to use eavesdrop which I plan on testing next week 
> to see if it scales, if someone has any input on it I would appreciate.
> The next option would be to optimize mod_conference for this use case, 
> I started looking at the code and identified some places that 
> performance can be optimized but may need to hire someone with better 
> C skills to do the actual coding.
> another possibility may be, to use an RTP proxy that would just 
> distribute the RTP to all callers, this would give the best 
> performance but would limit features.
> Last option would be to develop a new module optimized for low latency 
> streaming, but leave that as a last resort.
>
> On Fri, Apr 20, 2018 at 1:15 PM, Tihomir Culjaga <tculjaga at gmail.com 
> <mailto:tculjaga at gmail.com>> wrote:
>
>     a conference with 1000 members demands huge resources... and this
>     is not efficient if only 2-5% of members will actually talk.
>     you need to keep the actual conference to the minimum number of
>     "speakers" and move listeners to a streaming "service".
>
>     so, focus on how to get your entire conference streamed somewhere.
>
>     another thing to consider.... do the "listeners" really need to be
>     in time sync with the conference ? They will just hear the
>     conversation from the conference a bit later... thats all.
>
>     T.
>
>
>
>
>
>
>     On 20 April 2018 at 14:43, Abaci B <abaci64 at gmail.com
>     <mailto:abaci64 at gmail.com>> wrote:
>
>         problem with shoutcast streaming is the extreme latency
>
>         On Fri, Apr 20, 2018 at 1:30 AM, Giovanni Maruzzelli
>         <gmaruzz at gmail.com <mailto:gmaruzz at gmail.com>> wrote:
>
>             you definitely want to have only the speakers into
>             conference, and stream to all others.
>
>             -giovanni
>
>             On 19 April 2018 at 23:37, Abaci B <abaci64 at gmail.com
>             <mailto:abaci64 at gmail.com>> wrote:
>
>                 I was thinking that the muted flag (CAN_SPEAK?) is
>                 something that can change mid conference which means
>                 there is a chance of loosing a single frame (the last
>                 one right before muting). I was more interested in
>                 knowing as far as performance if it makes sense.
>                 not sure if this is safe but something like only
>                 starting conference_loop_launch_input if that flag is
>                 not set.
>                 I still wish there would be module that would do
>                 exactly this (something like shoutcast but local and
>                 without the shoutcast latency)
>
>                 On Thu, Apr 19, 2018 at 5:19 PM, Steven Ayre
>                 <steveayre at gmail.com <mailto:steveayre at gmail.com>> wrote:
>
>                     There's already a muted flag you could use for that.
>
>                     On 19 April 2018 at 20:55, Abaci B
>                     <abaci64 at gmail.com <mailto:abaci64 at gmail.com>> wrote:
>
>                         Thanks for the reply.
>                         Another idea I'm thinking of is to add a
>                         LISTEN_ONLY mflag to mod_conference and only
>                         run conference_loop_input thread on channels
>                         that don't have this flag, that way I don't
>                         need to set up a shoutcast server and have
>                         everything contained within freeswitch, does
>                         that make sense?
>
>                         On Thu, Apr 19, 2018 at 3:20 PM, Giovanni
>                         Maruzzelli <gmaruzz at gmail.com
>                         <mailto:gmaruzz at gmail.com>> wrote:
>
>                             definitely conference with two
>                             participant, and 598 on localstream
>
>                             -giovanni
>
>                             On Thu, Apr 19, 2018, 18:38 Abaci B
>                             <abaci64 at gmail.com
>                             <mailto:abaci64 at gmail.com>> wrote:
>
>                                 Hi,
>                                 I'm trying to set up a large audio
>                                 conference of over 1000 users where
>                                 only 1 or 2 users are talking and the
>                                 rest are just listening, but after
>                                 600-700 users the CPU usage gets too
>                                 high and the audio gets choppy, I
>                                 understand that mod_conference is not
>                                 optimized for my use case but was
>                                 wondering if someone has any
>                                 suggestions on what can be done to
>                                 optimize for my use case.
>                                 I was also thinking that maybe there
>                                 is a  better alternative for what I
>                                 look for, such as having just the
>                                 talkers on the conference and the
>                                 listeners should listen to it via
>                                 mod_local_stream or maybe use
>                                 something like eavesdrop for the
>                                 listeners but not sure if and how much
>                                 performance I would gain.
>                                 Thanks for any help or input
>
>                                 _________________________________________________________________________
>                                 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
>
>
>
>                     _________________________________________________________________________
>                     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
>
>
>
>
>             -- 
>
>             Sincerely,
>
>             Giovanni Maruzzelli
>             OpenTelecom.IT
>             cell: +39 347 266 56 18
>
>             _________________________________________________________________________
>             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
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From gmaruzz at gmail.com  Sat Apr 21 08:43:51 2018
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Sat, 21 Apr 2018 10:43:51 +0200
Subject: [Freeswitch-users] Question about large confeerence
In-Reply-To: <5849f54a-8603-7818-2aa6-4dce06cd7743@madovsky.org>
References: <CAFmZFiOZmdxtbYQJU9Ye1su0aRShqDWU+T1-j0iS32yy6pKeiw@mail.gmail.com>
 <CALXCt0rS4Mfcm8ObL8nCmOnYbMkxor9FcQy0AzPkar5B+JnUWg@mail.gmail.com>
 <CAFmZFiMBqsRuPwXYwPQho9fMR080XQRDoaPNcdkm-323YjTL_w@mail.gmail.com>
 <CAFiqYukYkTkTM2vZ2e9iyZMwqn-fcQXOasVg0JwHHPp-fhiutw@mail.gmail.com>
 <CAFmZFiMp05KUdkYSoSvd-tnGCrHgU56v6Ar5_rp6V2iLALYX_A@mail.gmail.com>
 <CALXCt0q2Y_vD8tBtrBicYD0ki8uBNZpypuOj3-NjMmpbS03NDw@mail.gmail.com>
 <CAFmZFiP+c+iiNN6mQNo3pxVwnG3pGCy39KDvoGrexw-NOxzkBQ@mail.gmail.com>
 <CAJoKKdVhz7h4Y6ksNz36=8W96BWjKFzOOmK22md3v6f4s9FRCg@mail.gmail.com>
 <CAFmZFiOxjqTKQs0uKHXZ4AUHR6qyKCVoE_H93xw8A9fpX+ciLg@mail.gmail.com>
 <5849f54a-8603-7818-2aa6-4dce06cd7743@madovsky.org>
Message-ID: <CALXCt0ptwz3fRchQzBsjVQ+Lp2a+9AqkYBE4JXCz_E2tkFugrw@mail.gmail.com>

is an audio conference.

what I would look into is something on these lines: have a conference with
only the speakers. Put the listeners on music on hold, and use the
conference output as music on hold stream. Give the listeners a bind meta
key for being connected to the conference if they need to speak.

-giovanni


On 20 April 2018 at 21:45, Madovsky <infos at madovsky.org> wrote:

> Latency is often caused by buffer and high quality video.
>
> if you really need zero latency so find a way to set the buffer stream to
> zero and use
>
> a very fast video encoding.
> On 4/20/2018 10:29 AM, Abaci B wrote:
>
> I do understand that the way mod_conference work it is not efficient for
> my use case, my question is about the best alternative.
> Streaming would be ideal if it wouldn't be for the latency, since users
> need to be able to switch to talk mode there is no way a latency of 5-10
> seconds (or possibly more) is acceptable.
> Another option is to use eavesdrop which I plan on testing next week to
> see if it scales, if someone has any input on it I would appreciate.
> The next option would be to optimize mod_conference for this use case, I
> started looking at the code and identified some places that performance can
> be optimized but may need to hire someone with better C skills to do the
> actual coding.
> another possibility may be, to use an RTP proxy that would just distribute
> the RTP to all callers, this would give the best performance but would
> limit features.
> Last option would be to develop a new module optimized for low latency
> streaming, but leave that as a last resort.
>
> On Fri, Apr 20, 2018 at 1:15 PM, Tihomir Culjaga <tculjaga at gmail.com>
> wrote:
>
>> a conference with 1000 members demands huge resources... and this is not
>> efficient if only 2-5% of members will actually talk.
>> you need to keep the actual conference to the minimum number of
>> "speakers" and move listeners to a streaming "service".
>>
>> so, focus on how to get your entire conference streamed somewhere.
>>
>> another thing to consider.... do the "listeners" really need to be in
>> time sync with the conference ? They will just hear the conversation from
>> the conference a bit later... thats all.
>>
>> T.
>>
>>
>>
>>
>>
>>
>> On 20 April 2018 at 14:43, Abaci B <abaci64 at gmail.com> wrote:
>>
>>> problem with shoutcast streaming is the extreme latency
>>>
>>> On Fri, Apr 20, 2018 at 1:30 AM, Giovanni Maruzzelli <gmaruzz at gmail.com>
>>> wrote:
>>>
>>>> you definitely want to have only the speakers into conference, and
>>>> stream to all others.
>>>>
>>>> -giovanni
>>>>
>>>> On 19 April 2018 at 23:37, Abaci B <abaci64 at gmail.com> wrote:
>>>>
>>>>> I was thinking that the muted flag (CAN_SPEAK?) is something that can
>>>>> change mid conference which means there is a chance of loosing a single
>>>>> frame (the last one right before muting). I was more interested in knowing
>>>>> as far as performance if it makes sense.
>>>>> not sure if this is safe but something like only starting
>>>>> conference_loop_launch_input if that flag is not set.
>>>>> I still wish there would be module that would do exactly this
>>>>> (something like shoutcast but local and without the shoutcast latency)
>>>>>
>>>>> On Thu, Apr 19, 2018 at 5:19 PM, Steven Ayre <steveayre at gmail.com>
>>>>> wrote:
>>>>>
>>>>>> There's already a muted flag you could use for that.
>>>>>>
>>>>>> On 19 April 2018 at 20:55, Abaci B <abaci64 at gmail.com> wrote:
>>>>>>
>>>>>>> Thanks for the reply.
>>>>>>> Another idea I'm thinking of is to add a LISTEN_ONLY mflag to
>>>>>>> mod_conference and only run conference_loop_input thread on channels that
>>>>>>> don't have this flag, that way I don't need to set up a shoutcast server
>>>>>>> and have everything contained within freeswitch, does that make sense?
>>>>>>>
>>>>>>> On Thu, Apr 19, 2018 at 3:20 PM, Giovanni Maruzzelli <
>>>>>>> gmaruzz at gmail.com> wrote:
>>>>>>>
>>>>>>>> definitely conference with two participant, and 598 on localstream
>>>>>>>>
>>>>>>>> -giovanni
>>>>>>>>
>>>>>>>> On Thu, Apr 19, 2018, 18:38 Abaci B <abaci64 at gmail.com> wrote:
>>>>>>>>
>>>>>>>>> Hi,
>>>>>>>>> I'm trying to set up a large audio conference of over 1000 users
>>>>>>>>> where only 1 or 2 users are talking and the rest are just listening, but
>>>>>>>>> after 600-700 users the CPU usage gets too high and the audio gets choppy,
>>>>>>>>> I understand that mod_conference is not optimized for my use case but was
>>>>>>>>> wondering if someone has any suggestions on what can be done to optimize
>>>>>>>>> for my use case.
>>>>>>>>> I was also thinking that maybe there is a  better alternative for
>>>>>>>>> what I look for, such as having just the talkers on the conference and the
>>>>>>>>> listeners should listen to it via mod_local_stream or maybe use something
>>>>>>>>> like eavesdrop for the listeners but not sure if and how much performance I
>>>>>>>>> would gain.
>>>>>>>>> Thanks for any help or input
>>>>>>>>>
>>>>>>>>> ____________________________________________________________
>>>>>>>>> _____________
>>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>>> consulting at freeswitch.org
>>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>>
>>>>>>>>> Official FreeSWITCH Sites
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>> http://confluence.freeswitch.org
>>>>>>>>> http://www.cluecon.com
>>>>>>>>>
>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free
>>>>>>>>> switch-users
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>>
>>>>>>>> ____________________________________________________________
>>>>>>>> _____________
>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>> consulting at freeswitch.org
>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>
>>>>>>>> Official FreeSWITCH Sites
>>>>>>>> http://www.freeswitch.org
>>>>>>>> http://confluence.freeswitch.org
>>>>>>>> http://www.cluecon.com
>>>>>>>>
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free
>>>>>>>> switch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> ____________________________________________________________
>>>>>>> _____________
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>
>>>>>>> Official FreeSWITCH Sites
>>>>>>> http://www.freeswitch.org
>>>>>>> http://confluence.freeswitch.org
>>>>>>> http://www.cluecon.com
>>>>>>>
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free
>>>>>>> switch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>
>>>>>>
>>>>>> ____________________________________________________________
>>>>>> _____________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://confluence.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free
>>>>>> switch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>> ____________________________________________________________
>>>>> _____________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free
>>>>> switch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>>
>>>> Sincerely,
>>>>
>>>> Giovanni Maruzzelli
>>>> OpenTelecom.IT
>>>> cell: +39 347 266 56 18
>>>>
>>>> ____________________________________________________________
>>>> _____________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free
>>>> switch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> ____________________________________________________________
>>> _____________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>
> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon.com
>
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

Sincerely,

Giovanni Maruzzelli
OpenTelecom.IT
cell: +39 347 266 56 18
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From infos at madovsky.org  Sat Apr 21 08:54:49 2018
From: infos at madovsky.org (Madovsky)
Date: Sat, 21 Apr 2018 01:54:49 -0700
Subject: [Freeswitch-users] Question about large confeerence
In-Reply-To: <CALXCt0ptwz3fRchQzBsjVQ+Lp2a+9AqkYBE4JXCz_E2tkFugrw@mail.gmail.com>
References: <CAFmZFiOZmdxtbYQJU9Ye1su0aRShqDWU+T1-j0iS32yy6pKeiw@mail.gmail.com>
 <CALXCt0rS4Mfcm8ObL8nCmOnYbMkxor9FcQy0AzPkar5B+JnUWg@mail.gmail.com>
 <CAFmZFiMBqsRuPwXYwPQho9fMR080XQRDoaPNcdkm-323YjTL_w@mail.gmail.com>
 <CAFiqYukYkTkTM2vZ2e9iyZMwqn-fcQXOasVg0JwHHPp-fhiutw@mail.gmail.com>
 <CAFmZFiMp05KUdkYSoSvd-tnGCrHgU56v6Ar5_rp6V2iLALYX_A@mail.gmail.com>
 <CALXCt0q2Y_vD8tBtrBicYD0ki8uBNZpypuOj3-NjMmpbS03NDw@mail.gmail.com>
 <CAFmZFiP+c+iiNN6mQNo3pxVwnG3pGCy39KDvoGrexw-NOxzkBQ@mail.gmail.com>
 <CAJoKKdVhz7h4Y6ksNz36=8W96BWjKFzOOmK22md3v6f4s9FRCg@mail.gmail.com>
 <CAFmZFiOxjqTKQs0uKHXZ4AUHR6qyKCVoE_H93xw8A9fpX+ciLg@mail.gmail.com>
 <5849f54a-8603-7818-2aa6-4dce06cd7743@madovsky.org>
 <CALXCt0ptwz3fRchQzBsjVQ+Lp2a+9AqkYBE4JXCz_E2tkFugrw@mail.gmail.com>
Message-ID: <01faf88a-aa72-f991-fdb9-7a3ecfd51d00@madovsky.org>

my mistake...

but buffer to close to zero for audio is also a good start....

why not to grab the conferencer stream to a live point

the if someone wants to talk so a 1 click call to the conference....

On 4/21/2018 1:43 AM, Giovanni Maruzzelli wrote:
> is an audio conference.
>
> what I would look into is something on these lines: have a conference 
> with only the speakers. Put the listeners on music on hold, and use 
> the conference output as music on hold stream. Give the listeners a 
> bind meta key for being connected to the conference if they need to speak.
>
> -giovanni
>
>
> On 20 April 2018 at 21:45, Madovsky <infos at madovsky.org 
> <mailto:infos at madovsky.org>> wrote:
>
>     Latency is often caused by buffer and high quality video.
>
>     if you really need zero latency so find a way to set the buffer
>     stream to zero and use
>
>     a very fast video encoding.
>
>     On 4/20/2018 10:29 AM, Abaci B wrote:
>>     I do understand that the way mod_conference work it is not
>>     efficient for my use case, my question is about the best alternative.
>>     Streaming would be ideal if it wouldn't be for the latency, since
>>     users need to be able to switch to talk mode there is no way a
>>     latency of 5-10 seconds (or possibly more) is acceptable.
>>     Another option is to use eavesdrop which I plan on testing next
>>     week to see if it scales, if someone has any input on it I would
>>     appreciate.
>>     The next option would be to optimize mod_conference for this use
>>     case, I started looking at the code and identified some places
>>     that performance can be optimized but may need to hire someone
>>     with better C skills to do the actual coding.
>>     another possibility may be, to use an RTP proxy that would just
>>     distribute the RTP to all callers, this would give the best
>>     performance but would limit features.
>>     Last option would be to develop a new module optimized for low
>>     latency streaming, but leave that as a last resort.
>>
>>     On Fri, Apr 20, 2018 at 1:15 PM, Tihomir Culjaga
>>     <tculjaga at gmail.com <mailto:tculjaga at gmail.com>> wrote:
>>
>>         a conference with 1000 members demands huge resources... and
>>         this is not efficient if only 2-5% of members will actually
>>         talk.
>>         you need to keep the actual conference to the minimum number
>>         of "speakers" and move listeners to a streaming "service".
>>
>>         so, focus on how to get your entire conference streamed
>>         somewhere.
>>
>>         another thing to consider.... do the "listeners" really need
>>         to be in time sync with the conference ? They will just hear
>>         the conversation from the conference a bit later... thats all.
>>
>>         T.
>>
>>
>>
>>
>>
>>
>>         On 20 April 2018 at 14:43, Abaci B <abaci64 at gmail.com
>>         <mailto:abaci64 at gmail.com>> wrote:
>>
>>             problem with shoutcast streaming is the extreme latency
>>
>>             On Fri, Apr 20, 2018 at 1:30 AM, Giovanni Maruzzelli
>>             <gmaruzz at gmail.com <mailto:gmaruzz at gmail.com>> wrote:
>>
>>                 you definitely want to have only the speakers into
>>                 conference, and stream to all others.
>>
>>                 -giovanni
>>
>>                 On 19 April 2018 at 23:37, Abaci B <abaci64 at gmail.com
>>                 <mailto:abaci64 at gmail.com>> wrote:
>>
>>                     I was thinking that the muted flag (CAN_SPEAK?)
>>                     is something that can change mid conference which
>>                     means there is a chance of loosing a single frame
>>                     (the last one right before muting). I was more
>>                     interested in knowing as far as performance if it
>>                     makes sense.
>>                     not sure if this is safe but something like only
>>                     starting conference_loop_launch_input if that
>>                     flag is not set.
>>                     I still wish there would be module that would do
>>                     exactly this (something like shoutcast but local
>>                     and without the shoutcast latency)
>>
>>                     On Thu, Apr 19, 2018 at 5:19 PM, Steven Ayre
>>                     <steveayre at gmail.com
>>                     <mailto:steveayre at gmail.com>> wrote:
>>
>>                         There's already a muted flag you could use
>>                         for that.
>>
>>                         On 19 April 2018 at 20:55, Abaci B
>>                         <abaci64 at gmail.com
>>                         <mailto:abaci64 at gmail.com>> wrote:
>>
>>                             Thanks for the reply.
>>                             Another idea I'm thinking of is to add a
>>                             LISTEN_ONLY mflag to mod_conference and
>>                             only run conference_loop_input thread on
>>                             channels that don't have this flag, that
>>                             way I don't need to set up a shoutcast
>>                             server and have everything contained
>>                             within freeswitch, does that make sense?
>>
>>                             On Thu, Apr 19, 2018 at 3:20 PM, Giovanni
>>                             Maruzzelli <gmaruzz at gmail.com
>>                             <mailto:gmaruzz at gmail.com>> wrote:
>>
>>                                 definitely conference with two
>>                                 participant, and 598 on localstream
>>
>>                                 -giovanni
>>
>>                                 On Thu, Apr 19, 2018, 18:38 Abaci B
>>                                 <abaci64 at gmail.com
>>                                 <mailto:abaci64 at gmail.com>> wrote:
>>
>>                                     Hi,
>>                                     I'm trying to set up a large
>>                                     audio conference of over 1000
>>                                     users where only 1 or 2 users are
>>                                     talking and the rest are just
>>                                     listening, but after 600-700
>>                                     users the CPU usage gets too high
>>                                     and the audio gets choppy, I
>>                                     understand that mod_conference is
>>                                     not optimized for my use case but
>>                                     was wondering if someone has any
>>                                     suggestions on what can be done
>>                                     to optimize for my use case.
>>                                     I was also thinking that maybe
>>                                     there is a  better alternative
>>                                     for what I look for, such as
>>                                     having just the talkers on the
>>                                     conference and the listeners
>>                                     should listen to it via
>>                                     mod_local_stream or maybe use
>>                                     something like eavesdrop for the
>>                                     listeners but not sure if and how
>>                                     much performance I would gain.
>>                                     Thanks for any help or input
>>
>>                                     _________________________________________________________________________
>>                                     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
>>
>>
>>
>>                         _________________________________________________________________________
>>                         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
>>
>>
>>
>>
>>                 -- 
>>
>>                 Sincerely,
>>
>>                 Giovanni Maruzzelli
>>                 OpenTelecom.IT
>>                 cell: +39 347 266 56 18
>>
>>                 _________________________________________________________________________
>>                 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
>>
>>
>>
>>
>>     _________________________________________________________________________
>>     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
>
>
>
>
> -- 
>
> Sincerely,
>
> Giovanni Maruzzelli
> OpenTelecom.IT
> cell: +39 347 266 56 18
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From babak.freeswitch at gmail.com  Sun Apr 22 05:05:05 2018
From: babak.freeswitch at gmail.com (Babak Yakhchali)
Date: Sun, 22 Apr 2018 09:35:05 +0430
Subject: [Freeswitch-users] High jitter on outgoing rtp
In-Reply-To: <CAEVfcaZ64F-L+JK97MK9UrqFePLwekyLAHt1bkcsGnmDQneOcw@mail.gmail.com>
References: <CAEVfcaZ64F-L+JK97MK9UrqFePLwekyLAHt1bkcsGnmDQneOcw@mail.gmail.com>
Message-ID: <CAEVfcaa89Wrkz60-5VL_cMQ63=P7ZhZ1922RKKMHdsk4Ro+qeg@mail.gmail.com>

setting rtp-rewrite-timestamps=true in internal and external profile solved
my problem

On Sat, Apr 21, 2018 at 3:05 PM, Babak Yakhchali <babak.freeswitch at gmail.com
> wrote:

> Hi
> I'm experiencing intermittent audio cut outs on IP PHones. When I capture
> packets using tcpdum it seems that rtp streams going out from freeswitch
> have high jitter. My network is like this:
> IP Phone(192.168.1.15) <=> freeswitch (192.168.1.191) <=> Gateway
> (192.168.1.11)
> all of these are on the same lan and freeswitch is latest stable running
> on debian 8.10 and server is hp g7
> is it related to network drivers or any configuration on kernel level?
> thanks
>
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From gmaruzz at gmail.com  Sun Apr 22 07:37:55 2018
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Sun, 22 Apr 2018 09:37:55 +0200
Subject: [Freeswitch-users] High jitter on outgoing rtp
In-Reply-To: <CAEVfcaa89Wrkz60-5VL_cMQ63=P7ZhZ1922RKKMHdsk4Ro+qeg@mail.gmail.com>
References: <CAEVfcaZ64F-L+JK97MK9UrqFePLwekyLAHt1bkcsGnmDQneOcw@mail.gmail.com>
 <CAEVfcaa89Wrkz60-5VL_cMQ63=P7ZhZ1922RKKMHdsk4Ro+qeg@mail.gmail.com>
Message-ID: <CALXCt0oHKrBCxfOFwv2Ljsn_fardUF48Ph8epysvh2PA_rGatw@mail.gmail.com>

On 22 April 2018 at 07:05, Babak Yakhchali <babak.freeswitch at gmail.com>
wrote:

> setting rtp-rewrite-timestamps=true in internal and external profile
> solved my problem
>
>
Thanks for sharing this one, that's how community grows!

-giovanni
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From rajil.s at gmail.com  Sun Apr 22 17:17:33 2018
From: rajil.s at gmail.com (Rajil Saraswat)
Date: Sun, 22 Apr 2018 12:17:33 -0500
Subject: [Freeswitch-users] IPv6 endpoints on LAN getting Error Cause: 19
	[NO_ANSWER]
Message-ID: <aed068ba-9f1e-2bad-921f-00ba56d7cdd3@gmail.com>

Hello,

I have two Linphone clients connected to a PBX server on my LAN. Both
the sip clients are registered to the internal-ipv6 profile.
Unfortunately, i am not able to call between them. I get an error

 [INFO] mod_dptools.c:3436 Originate Failed.  Cause: NO_ANSWER

Any hints on why this may be happening?

Thanks




From abaci64 at gmail.com  Sun Apr 22 21:38:33 2018
From: abaci64 at gmail.com (Abaci B)
Date: Sun, 22 Apr 2018 17:38:33 -0400
Subject: [Freeswitch-users] Question about large confeerence
In-Reply-To: <CALXCt0ptwz3fRchQzBsjVQ+Lp2a+9AqkYBE4JXCz_E2tkFugrw@mail.gmail.com>
References: <CAFmZFiOZmdxtbYQJU9Ye1su0aRShqDWU+T1-j0iS32yy6pKeiw@mail.gmail.com>
 <CALXCt0rS4Mfcm8ObL8nCmOnYbMkxor9FcQy0AzPkar5B+JnUWg@mail.gmail.com>
 <CAFmZFiMBqsRuPwXYwPQho9fMR080XQRDoaPNcdkm-323YjTL_w@mail.gmail.com>
 <CAFiqYukYkTkTM2vZ2e9iyZMwqn-fcQXOasVg0JwHHPp-fhiutw@mail.gmail.com>
 <CAFmZFiMp05KUdkYSoSvd-tnGCrHgU56v6Ar5_rp6V2iLALYX_A@mail.gmail.com>
 <CALXCt0q2Y_vD8tBtrBicYD0ki8uBNZpypuOj3-NjMmpbS03NDw@mail.gmail.com>
 <CAFmZFiP+c+iiNN6mQNo3pxVwnG3pGCy39KDvoGrexw-NOxzkBQ@mail.gmail.com>
 <CAJoKKdVhz7h4Y6ksNz36=8W96BWjKFzOOmK22md3v6f4s9FRCg@mail.gmail.com>
 <CAFmZFiOxjqTKQs0uKHXZ4AUHR6qyKCVoE_H93xw8A9fpX+ciLg@mail.gmail.com>
 <5849f54a-8603-7818-2aa6-4dce06cd7743@madovsky.org>
 <CALXCt0ptwz3fRchQzBsjVQ+Lp2a+9AqkYBE4JXCz_E2tkFugrw@mail.gmail.com>
Message-ID: <CAFmZFiPv6OCajSAD=xYW=s4u2CJ4-GEFzSFVPPxyga75nfgGug@mail.gmail.com>

You mean creating a new endpoint that would stream the conference, same
idea as mod_lcoal_stream can stream a shoutcast stream? that's what I was
thinking of but after looking at the code involved I see that is beyond my
C skills, and I have a budget of only around $500 for this so don't think I
will find someone else to do it.

On Sat, Apr 21, 2018 at 4:43 AM, Giovanni Maruzzelli <gmaruzz at gmail.com>
wrote:

> is an audio conference.
>
> what I would look into is something on these lines: have a conference with
> only the speakers. Put the listeners on music on hold, and use the
> conference output as music on hold stream. Give the listeners a bind meta
> key for being connected to the conference if they need to speak.
>
> -giovanni
>
>
>
> On 20 April 2018 at 21:45, Madovsky <infos at madovsky.org> wrote:
>
>> Latency is often caused by buffer and high quality video.
>>
>> if you really need zero latency so find a way to set the buffer stream to
>> zero and use
>>
>> a very fast video encoding.
>> On 4/20/2018 10:29 AM, Abaci B wrote:
>>
>> I do understand that the way mod_conference work it is not efficient for
>> my use case, my question is about the best alternative.
>> Streaming would be ideal if it wouldn't be for the latency, since users
>> need to be able to switch to talk mode there is no way a latency of 5-10
>> seconds (or possibly more) is acceptable.
>> Another option is to use eavesdrop which I plan on testing next week to
>> see if it scales, if someone has any input on it I would appreciate.
>> The next option would be to optimize mod_conference for this use case, I
>> started looking at the code and identified some places that performance can
>> be optimized but may need to hire someone with better C skills to do the
>> actual coding.
>> another possibility may be, to use an RTP proxy that would just
>> distribute the RTP to all callers, this would give the best performance but
>> would limit features.
>> Last option would be to develop a new module optimized for low latency
>> streaming, but leave that as a last resort.
>>
>> On Fri, Apr 20, 2018 at 1:15 PM, Tihomir Culjaga <tculjaga at gmail.com>
>> wrote:
>>
>>> a conference with 1000 members demands huge resources... and this is not
>>> efficient if only 2-5% of members will actually talk.
>>> you need to keep the actual conference to the minimum number of
>>> "speakers" and move listeners to a streaming "service".
>>>
>>> so, focus on how to get your entire conference streamed somewhere.
>>>
>>> another thing to consider.... do the "listeners" really need to be in
>>> time sync with the conference ? They will just hear the conversation from
>>> the conference a bit later... thats all.
>>>
>>> T.
>>>
>>>
>>>
>>>
>>>
>>>
>>> On 20 April 2018 at 14:43, Abaci B <abaci64 at gmail.com> wrote:
>>>
>>>> problem with shoutcast streaming is the extreme latency
>>>>
>>>> On Fri, Apr 20, 2018 at 1:30 AM, Giovanni Maruzzelli <gmaruzz at gmail.com
>>>> > wrote:
>>>>
>>>>> you definitely want to have only the speakers into conference, and
>>>>> stream to all others.
>>>>>
>>>>> -giovanni
>>>>>
>>>>> On 19 April 2018 at 23:37, Abaci B <abaci64 at gmail.com> wrote:
>>>>>
>>>>>> I was thinking that the muted flag (CAN_SPEAK?) is something that can
>>>>>> change mid conference which means there is a chance of loosing a single
>>>>>> frame (the last one right before muting). I was more interested in knowing
>>>>>> as far as performance if it makes sense.
>>>>>> not sure if this is safe but something like only starting
>>>>>> conference_loop_launch_input if that flag is not set.
>>>>>> I still wish there would be module that would do exactly this
>>>>>> (something like shoutcast but local and without the shoutcast latency)
>>>>>>
>>>>>> On Thu, Apr 19, 2018 at 5:19 PM, Steven Ayre <steveayre at gmail.com>
>>>>>> wrote:
>>>>>>
>>>>>>> There's already a muted flag you could use for that.
>>>>>>>
>>>>>>> On 19 April 2018 at 20:55, Abaci B <abaci64 at gmail.com> wrote:
>>>>>>>
>>>>>>>> Thanks for the reply.
>>>>>>>> Another idea I'm thinking of is to add a LISTEN_ONLY mflag to
>>>>>>>> mod_conference and only run conference_loop_input thread on channels that
>>>>>>>> don't have this flag, that way I don't need to set up a shoutcast server
>>>>>>>> and have everything contained within freeswitch, does that make sense?
>>>>>>>>
>>>>>>>> On Thu, Apr 19, 2018 at 3:20 PM, Giovanni Maruzzelli <
>>>>>>>> gmaruzz at gmail.com> wrote:
>>>>>>>>
>>>>>>>>> definitely conference with two participant, and 598 on localstream
>>>>>>>>>
>>>>>>>>> -giovanni
>>>>>>>>>
>>>>>>>>> On Thu, Apr 19, 2018, 18:38 Abaci B <abaci64 at gmail.com> wrote:
>>>>>>>>>
>>>>>>>>>> Hi,
>>>>>>>>>> I'm trying to set up a large audio conference of over 1000 users
>>>>>>>>>> where only 1 or 2 users are talking and the rest are just listening, but
>>>>>>>>>> after 600-700 users the CPU usage gets too high and the audio gets choppy,
>>>>>>>>>> I understand that mod_conference is not optimized for my use case but was
>>>>>>>>>> wondering if someone has any suggestions on what can be done to optimize
>>>>>>>>>> for my use case.
>>>>>>>>>> I was also thinking that maybe there is a  better alternative for
>>>>>>>>>> what I look for, such as having just the talkers on the conference and the
>>>>>>>>>> listeners should listen to it via mod_local_stream or maybe use something
>>>>>>>>>> like eavesdrop for the listeners but not sure if and how much performance I
>>>>>>>>>> would gain.
>>>>>>>>>> Thanks for any help or input
>>>>>>>>>>
>>>>>>>>>> ____________________________________________________________
>>>>>>>>>> _____________
>>>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>>>> consulting at freeswitch.org
>>>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>>>
>>>>>>>>>> Official FreeSWITCH Sites
>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>> http://confluence.freeswitch.org
>>>>>>>>>> http://www.cluecon.com
>>>>>>>>>>
>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free
>>>>>>>>>> switch-users
>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> ____________________________________________________________
>>>>>>>>> _____________
>>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>>> consulting at freeswitch.org
>>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>>
>>>>>>>>> Official FreeSWITCH Sites
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>> http://confluence.freeswitch.org
>>>>>>>>> http://www.cluecon.com
>>>>>>>>>
>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free
>>>>>>>>> switch-users
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> ____________________________________________________________
>>>>>>>> _____________
>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>> consulting at freeswitch.org
>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>
>>>>>>>> Official FreeSWITCH Sites
>>>>>>>> http://www.freeswitch.org
>>>>>>>> http://confluence.freeswitch.org
>>>>>>>> http://www.cluecon.com
>>>>>>>>
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free
>>>>>>>> switch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> ____________________________________________________________
>>>>>>> _____________
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>
>>>>>>> Official FreeSWITCH Sites
>>>>>>> http://www.freeswitch.org
>>>>>>> http://confluence.freeswitch.org
>>>>>>> http://www.cluecon.com
>>>>>>>
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free
>>>>>>> switch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>
>>>>>>
>>>>>> ____________________________________________________________
>>>>>> _____________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://confluence.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free
>>>>>> switch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>>
>>>>> --
>>>>>
>>>>> Sincerely,
>>>>>
>>>>> Giovanni Maruzzelli
>>>>> OpenTelecom.IT
>>>>> cell: +39 347 266 56 18
>>>>>
>>>>> ____________________________________________________________
>>>>> _____________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free
>>>>> switch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>> ____________________________________________________________
>>>> _____________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free
>>>> switch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> ____________________________________________________________
>>> _____________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon.com
>>
>> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
>
> Sincerely,
>
> Giovanni Maruzzelli
> OpenTelecom.IT
> cell: +39 347 266 56 18
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From gmaruzz at gmail.com  Mon Apr 23 04:56:57 2018
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Mon, 23 Apr 2018 06:56:57 +0200
Subject: [Freeswitch-users] Question about large confeerence
In-Reply-To: <CAFmZFiPv6OCajSAD=xYW=s4u2CJ4-GEFzSFVPPxyga75nfgGug@mail.gmail.com>
References: <CAFmZFiOZmdxtbYQJU9Ye1su0aRShqDWU+T1-j0iS32yy6pKeiw@mail.gmail.com>
 <CALXCt0rS4Mfcm8ObL8nCmOnYbMkxor9FcQy0AzPkar5B+JnUWg@mail.gmail.com>
 <CAFmZFiMBqsRuPwXYwPQho9fMR080XQRDoaPNcdkm-323YjTL_w@mail.gmail.com>
 <CAFiqYukYkTkTM2vZ2e9iyZMwqn-fcQXOasVg0JwHHPp-fhiutw@mail.gmail.com>
 <CAFmZFiMp05KUdkYSoSvd-tnGCrHgU56v6Ar5_rp6V2iLALYX_A@mail.gmail.com>
 <CALXCt0q2Y_vD8tBtrBicYD0ki8uBNZpypuOj3-NjMmpbS03NDw@mail.gmail.com>
 <CAFmZFiP+c+iiNN6mQNo3pxVwnG3pGCy39KDvoGrexw-NOxzkBQ@mail.gmail.com>
 <CAJoKKdVhz7h4Y6ksNz36=8W96BWjKFzOOmK22md3v6f4s9FRCg@mail.gmail.com>
 <CAFmZFiOxjqTKQs0uKHXZ4AUHR6qyKCVoE_H93xw8A9fpX+ciLg@mail.gmail.com>
 <5849f54a-8603-7818-2aa6-4dce06cd7743@madovsky.org>
 <CALXCt0ptwz3fRchQzBsjVQ+Lp2a+9AqkYBE4JXCz_E2tkFugrw@mail.gmail.com>
 <CAFmZFiPv6OCajSAD=xYW=s4u2CJ4-GEFzSFVPPxyga75nfgGug@mail.gmail.com>
Message-ID: <CALXCt0rpmHu4o5ptn5CgdHPbGdZH=Pun74eF8ROvHsJaeqtagA@mail.gmail.com>

On 22 April 2018 at 23:38, Abaci B <abaci64 at gmail.com> wrote:

> You mean creating a new endpoint that would stream the conference, same
> idea as mod_lcoal_stream can stream a shoutcast stream? that's what I was
> thinking of but after looking at the code involved I see that is beyond my
> C skills, and I have a budget of only around $500 for this so don't think I
> will find someone else to do it.
>
>
I do not see $500 to fit the bill :)

with a little bit of ingenuity, you can cobble together a solution:

1) have a conf with speakers only with a meta key that transfer them to
extension abc if they don't need to speak anymore
2) have the conf outcall extension xyz
3) extension xyz answers and then shoutcast to shoutcast server
4) shoutcast server is used as source for music on hold
5) all listeners calls extension abc
6) extenmsion abc answers then put the caller on music on hold and give
them a meta key that will transfer them to conf if they need to speak

CPU=0
Profit!
:D

or, you can make a variation that uses mod_shell_stream that pipes the
outcall to a temp mp3 file that is then streamed by music on hold to
listeners.

be creative!

good luck,

-giovanni


-- 

Sincerely,

Giovanni Maruzzelli
OpenTelecom.IT
cell: +39 347 266 56 18
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From ksrigo at gmail.com  Mon Apr 23 12:28:58 2018
From: ksrigo at gmail.com (KSrigo)
Date: Mon, 23 Apr 2018 22:28:58 +1000
Subject: [Freeswitch-users] How to add headers to an outgoing REGISTER
In-Reply-To: <003601d3d870$f5342b50$df9c81f0$@novatec.de>
References: <000801d3d7af$f81068f0$e8313ad0$@novatec.de>
 <CAAiAxuz6VfiJR5rro946ruQPjhGCYSXQ0MMW8vLNXb=X-DV4tA@mail.gmail.com>
 <003601d3d870$f5342b50$df9c81f0$@novatec.de>
Message-ID: <9B4DBB6B-19F2-4F9F-B618-A28A5B0577CB@gmail.com>

Hi Lars,

Thats what I understood. I’m not sure you can add a header to a SIP register send by FS. Is there any reason why you want to add these headers?

srigo


> On Apr 20, 2018, at 4:29 PM, Lars Paulsen <lapa at novatec.de> wrote:
> 
> Hello Srigo,
>  
> yes, a gateway sending out a REGISTER. And I want to add headers to it before it’s send out.
> I DO NOT mean adding headers to an REGISTER response send to a local extension.
>  
> To be precise I wanted to add “Require” and “Proxy-Require” headers.
> But I failed on adding completely new headers.
>  
> Regards,
> Lars
>  
> Von: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Srigo Kanapathipillai
> Gesendet: Freitag, 20. April 2018 03:40
> An: FreeSWITCH Users Help
> Betreff: Re: [Freeswitch-users] How to add headers to an outgoing REGISTER
>  
> Hi,
>  
> What you mean by outgoing register?a Gateway registering? 
>  
> If yes, I'm not sure you can do it.
>  
> Srigo
>  
> On Fri, 20 Apr 2018, 01:33 Lars Paulsen, <lapa at novatec.de <mailto:lapa at novatec.de>> wrote:
>> Hello,
>>  
>> I have read the documentation under https://wiki.freeswitch.org/wiki/Sofia-SIP#Adding_Request_Headers <https://wiki.freeswitch.org/wiki/Sofia-SIP#Adding_Request_Headers> and as I understood it is possible to add
>> Headers to SIP requests simply by changing the configuration.
>>  
>> But the documentation refers to the dialing plan.
>> I tried to add headers to an outgoing REGISTER but failed.
>>  
>> Because the documentation refers to the dialplan and changing headers for existing calls, I am not sure if adding headers to a REGISTER request is possible or not.
>> Can anyone please tell me if it is possible or not and if yes, how to formulate the correct “<action application=”set” …>” and “<condition>” sections in the configuration.
>>  
>> Thanks in advance for all answers.
>>  
>> Best Regards,
>> Lars
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org <http://www.freeswitch.org/>
>> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
>> http://www.cluecon.com <http://www.cluecon.com/>
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
>> http://www.freeswitch.org <http://www.freeswitch.org/>_________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> http://www.cluecon.com <http://www.cluecon.com/>
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> http://www.freeswitch.org <http://www.freeswitch.org/>
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From lapa at novatec.de  Mon Apr 23 12:38:31 2018
From: lapa at novatec.de (Lars Paulsen)
Date: Mon, 23 Apr 2018 14:38:31 +0200
Subject: [Freeswitch-users] How to add headers to an outgoing REGISTER
In-Reply-To: <9B4DBB6B-19F2-4F9F-B618-A28A5B0577CB@gmail.com>
References: <000801d3d7af$f81068f0$e8313ad0$@novatec.de>
 <CAAiAxuz6VfiJR5rro946ruQPjhGCYSXQ0MMW8vLNXb=X-DV4tA@mail.gmail.com>
 <003601d3d870$f5342b50$df9c81f0$@novatec.de>
 <9B4DBB6B-19F2-4F9F-B618-A28A5B0577CB@gmail.com>
Message-ID: <006f01d3daff$fdf67720$f9e36560$@novatec.de>

Hello Srigo,

 

yes, there is.

 

I am trying to achieve a RFC6140 compliant registration (client side).
This is e.g. a part of the SIPConnect specification from the SIPForum.



As per RFC6140 this requires the following for a REGISTER request on the client side:

-          include URI parameter "bnc" in Contact-Header-URI (this was simply solved using the configuration)

-          include "gin" option tag in "Require" and "Proxy-Require" headers (that’s where I actually fail)

 

Thanks for looking into this.

 

Regards,
Lars

 

Von: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von KSrigo
Gesendet: Montag, 23. April 2018 14:29
An: FreeSWITCH Users Help
Betreff: Re: [Freeswitch-users] How to add headers to an outgoing REGISTER

 

Hi Lars,

 

Thats what I understood. I’m not sure you can add a header to a SIP register send by FS. Is there any reason why you want to add these headers?

 

srigo

 

 

On Apr 20, 2018, at 4:29 PM, Lars Paulsen <lapa at novatec.de> wrote:

 

Hello Srigo,

 

yes, a gateway sending out a REGISTER. And I want to add headers to it before it’s send out.
I DO NOT mean adding headers to an REGISTER response send to a local extension.

 

To be precise I wanted to add “Require” and “Proxy-Require” headers.

But I failed on adding completely new headers.

 

Regards,
Lars

 

Von: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Srigo Kanapathipillai
Gesendet: Freitag, 20. April 2018 03:40
An: FreeSWITCH Users Help
Betreff: Re: [Freeswitch-users] How to add headers to an outgoing REGISTER

 

Hi,

 

What you mean by outgoing register?a Gateway registering? 

 

If yes, I'm not sure you can do it.

 

Srigo

 

On Fri, 20 Apr 2018, 01:33 Lars Paulsen, < <mailto:lapa at novatec.de> lapa at novatec.de> wrote:

Hello,

 

I have read the documentation under  <https://wiki.freeswitch.org/wiki/Sofia-SIP#Adding_Request_Headers> https://wiki.freeswitch.org/wiki/Sofia-SIP#Adding_Request_Headers and as I understood it is possible to add

Headers to SIP requests simply by changing the configuration.

 

But the documentation refers to the dialing plan.

I tried to add headers to an outgoing REGISTER but failed.

 

Because the documentation refers to the dialplan and changing headers for existing calls, I am not sure if adding headers to a REGISTER request is possible or not.

Can anyone please tell me if it is possible or not and if yes, how to formulate the correct “<action application=”set” …>” and “<condition>” sections in the configuration.

 

Thanks in advance for all answers.

 

Best Regards,
Lars

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

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

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

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

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

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

 

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From vma at vallimamod.org  Mon Apr 23 14:13:04 2018
From: vma at vallimamod.org (Vallimamod Abdullah)
Date: Mon, 23 Apr 2018 16:13:04 +0200
Subject: [Freeswitch-users] RTP reception before sending any RTP packet
In-Reply-To: <CA+fRVSXm0gWgmTQu5KNQ-76iT3aau4=0WZFGc5OXCv96E4V51g@mail.gmail.com>
References: <CA+fRVSWHki+k28uVTyFVHK1Z93YDGaB05LoZtStzcSVXBnraDg@mail.gmail.com>
 <36896921-538C-4353-8920-C80FA763FC09@vallimamod.org>
 <CA+fRVSXm0gWgmTQu5KNQ-76iT3aau4=0WZFGc5OXCv96E4V51g@mail.gmail.com>
Message-ID: <CDA59718-F658-408F-B856-66CDE9F85D58@vallimamod.org>

Hi Mickael,

Are you doing SDP mangling on the opensips box? If so, freeswitch may not detect that the endpoint is nated as the RTP ip is public.

Can you try to disable it and let freeswitch auto-detect the public RTP ip/port?

If the auto-adjust is activated, you will get something like "Auto Changing audio port from <local ip:port> to <public ip:port>" in the debug logs.

Hope this helps!

Best Regards,
-- 
Vallimamod Abdullah
SIP Solutions
linkedin.com/in/vallimamod
.

> On 20 Apr 2018, at 10:47, Mickael Hubert <mickael at winlux.fr> wrote:
> 
> Thanks for your answer Vallimamod,
> 
> You can see our dialplan bellow:
> We tried these commands "rtp auto adjust, etc ...", but the issue continues.
> 
> is there a way to show if FS knows when the call is natted or not ? FYI we have an opensips of front for SIP, but RTP is send directly to FS.
> 
> SIP call --> Opensips (nat detection) --> FS
> RTP flow ------------------------------------------> FS
> 
> <condition field="${destinationAuthorized}" expression="^yes$" break="never">
>          <action application="export" data="codec_string=${ep_codec_string}"/>
>          <action application="export" data="sip_h_X-cid=${uuid}"/>
>          <action application="set" data="hangup_after_bridge=true"/>
>           <action application="set" data="ignore_early_media=false"/>
>           <action application="set" data="ignore_display_updates=true"/>
>           <action application="set" data="inherit_codec=true"/>
>           <action application="set" data="fax_enable_t38=false"/>
>           <action application="export" data="suppress_cng=true"/>
>           <action application="set" data="disable_rtp_auto_adjust=false"/>
>           <action application="set" data="rtp_auto_adjust_threshold=1"/>
>           <action application="set" data="rtp_manual_rtp_bugs=accept_any_packets"/>
>           <action application="unset" data="sip_h_X-Username"/>
>           <action application="bridge" data="
>           {
>               sip_invite_params=user=phone,
>               sip_cid_type=none,
>               ignore_display_updates=true,
>                sip_contact_user=${caller_id_number}
>           }
>           sofia/internal/${destination_number}@${distributor(LISTINTERNAL ${sofia(profile internal gwlist down)})}"/>
> 
> Thanks in advance
> 
> 
> 2018-04-19 11:35 GMT+02:00 Vallimamod Abdullah <vma at vallimamod.org>:
> Hi,
> 
> To my knowledge, it is the default beahviour for freeswitch to wait for incoming rtp from nated endpoints before starting to transmit.
> 
> You can explicitely force it with the profile param disable-rtp-auto-adjust=false.
> You can also play with the channel variable rtp_auto_adjust_threshold which defines the number of rtp packets to wait for initially (default is 10 iirc)
> 
> Best Regards,
> -- 
> Vallimamod Abdullah
> SIP Solutions
> linkedin.com/in/vallimamod
> .
> 
> > On 19 Apr 2018, at 10:31, Mickael Hubert <mickael at winlux.fr> wrote:
> > 
> > Hi list, 
> > I'd like to know if there's a way to tell FS to wait for RTP reception before sending any RTP packet ? 
> > We're facing a problem regarding early media with a PBX, NATed behind a dumb router : actually, if FS first sends RTP to the PBX, without waiting for RTP coming from the PBX, there's no audio. In some cases (related to the provider we use behind FS to establish call to the PSTN) FS isn't the 'first-shooter' of the RTP flow, and everything works fine, but when FS is the first to send RTP, it doesn't work... 
> > Because we can't change anything at our providers configs, we'd like to know if FS can "delay" RTP sending until it receives some RTP from the calling party.
> >  
> > The schematic is following :
> > PBX ===> FS ===> ...voice backbone... ===> Providers
> > (FS is acting as an SBC)
> >  
> > Thanks for your help, 
> > 


From adrottenberg at gmail.com  Mon Apr 23 14:28:09 2018
From: adrottenberg at gmail.com (Duvid Rottenberg)
Date: Mon, 23 Apr 2018 10:28:09 -0400
Subject: [Freeswitch-users] Question about large confeerence
In-Reply-To: <CAFmZFiOZmdxtbYQJU9Ye1su0aRShqDWU+T1-j0iS32yy6pKeiw@mail.gmail.com>
References: <CAFmZFiOZmdxtbYQJU9Ye1su0aRShqDWU+T1-j0iS32yy6pKeiw@mail.gmail.com>
Message-ID: <CAC4JBEAami6epx2hL4OjzguwsOdvnXSFDgnUOPHF9mEp1PRsbg@mail.gmail.com>

I have a similar scenario, and my solution was to create several
conferences. I have one master conference, that serves to connect all
sub-conferences. So when a user calls in to the conference I add him to a
sub conference, then I do an outdial to an extension that joins the sub
conference to the master conference (this connection should not be muted).
Once the sub conference has 250 participants, I create a new sub
conference. I control all of this via ESL, this also requires some
additional logic to handle conference commands, as I need to make sure it
propogates to all sub conferences.

The advantage of this approach is that instead of having one loop running
on 1 CPU core looping through all listeners we now have multiple conference
loops running on different cores. The issue with large conferences is that
the conference loop runs every 20 ms by default and the main conference
loop is not (and probably can't be) multi-threaded.

Since on any large conference, there will most likely be only a handful of
speakers, the best solution would probably be to optimize mod_conference to
have a separate list of unmuted participants and only loop through those,
but I was afraid to undertake such a change myself.



On Thu, Apr 19, 2018 at 12:09 PM, Abaci B <abaci64 at gmail.com> wrote:

> Hi,
> I'm trying to set up a large audio conference of over 1000 users where
> only 1 or 2 users are talking and the rest are just listening, but after
> 600-700 users the CPU usage gets too high and the audio gets choppy, I
> understand that mod_conference is not optimized for my use case but was
> wondering if someone has any suggestions on what can be done to optimize
> for my use case.
> I was also thinking that maybe there is a  better alternative for what I
> look for, such as having just the talkers on the conference and the
> listeners should listen to it via mod_local_stream or maybe use something
> like eavesdrop for the listeners but not sure if and how much performance I
> would gain.
> Thanks for any help or input
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 vma at vallimamod.org  Mon Apr 23 14:38:58 2018
From: vma at vallimamod.org (Vallimamod Abdullah)
Date: Mon, 23 Apr 2018 16:38:58 +0200
Subject: [Freeswitch-users] How to add headers to an outgoing REGISTER
In-Reply-To: <006f01d3daff$fdf67720$f9e36560$@novatec.de>
References: <000801d3d7af$f81068f0$e8313ad0$@novatec.de>
 <CAAiAxuz6VfiJR5rro946ruQPjhGCYSXQ0MMW8vLNXb=X-DV4tA@mail.gmail.com>
 <003601d3d870$f5342b50$df9c81f0$@novatec.de>
 <9B4DBB6B-19F2-4F9F-B618-A28A5B0577CB@gmail.com>
 <006f01d3daff$fdf67720$f9e36560$@novatec.de>
Message-ID: <D86BD690-476E-480D-A2F8-0963EEAC539B@vallimamod.org>

Hi,

I have made a quick check on the source and have not seen any call to the SIPTAG_PROXY_REQUIRE_STR() or SIPTAG_REQUIRE_STR() sofia macros. 

So, if I am not mistaken, a patch would be required to add them in the gateway registration functions. 


Best Regards,
-- 
Vallimamod Abdullah
SIP Solutions
linkedin.com/in/vallimamod
.

> On 23 Apr 2018, at 14:38, Lars Paulsen <lapa at novatec.de> wrote:
> 
> Hello Srigo,
>  
> yes, there is.
>  
> I am trying to achieve a RFC6140 compliant registration (client side).
> This is e.g. a part of the SIPConnect specification from the SIPForum.
> 
> As per RFC6140 this requires the following for a REGISTER request on the client side:
> -          include URI parameter "bnc" in Contact-Header-URI (this was simply solved using the configuration)
> -          include "gin" option tag in "Require" and "Proxy-Require" headers (that’s where I actually fail)
>  
> Thanks for looking into this.
>  
> Regards,
> Lars
>  
> Von: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von KSrigo
> Gesendet: Montag, 23. April 2018 14:29
> An: FreeSWITCH Users Help
> Betreff: Re: [Freeswitch-users] How to add headers to an outgoing REGISTER
>  
> Hi Lars,
>  
> Thats what I understood. I’m not sure you can add a header to a SIP register send by FS. Is there any reason why you want to add these headers?
>  
> srigo
>  
>  
>> On Apr 20, 2018, at 4:29 PM, Lars Paulsen <lapa at novatec.de> wrote:
>>  
>> Hello Srigo,
>>  
>> yes, a gateway sending out a REGISTER. And I want to add headers to it before it’s send out.
>> I DO NOT mean adding headers to an REGISTER response send to a local extension.
>>  
>> To be precise I wanted to add “Require” and “Proxy-Require” headers.
>> But I failed on adding completely new headers.
>>  
>> Regards,
>> Lars
>>  
>> Von: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Srigo Kanapathipillai
>> Gesendet: Freitag, 20. April 2018 03:40
>> An: FreeSWITCH Users Help
>> Betreff: Re: [Freeswitch-users] How to add headers to an outgoing REGISTER
>>  
>> Hi,
>>  
>> What you mean by outgoing register?a Gateway registering? 
>>  
>> If yes, I'm not sure you can do it.
>>  
>> Srigo
>>  
>> On Fri, 20 Apr 2018, 01:33 Lars Paulsen, <lapa at novatec.de> wrote:
>>> Hello,
>>>  
>>> I have read the documentation under https://wiki.freeswitch.org/wiki/Sofia-SIP#Adding_Request_Headers and as I understood it is possible to add
>>> Headers to SIP requests simply by changing the configuration.
>>>  
>>> But the documentation refers to the dialing plan.
>>> I tried to add headers to an outgoing REGISTER but failed.
>>>  
>>> Because the documentation refers to the dialplan and changing headers for existing calls, I am not sure if adding headers to a REGISTER request is possible or not.
>>> Can anyone please tell me if it is possible or not and if yes, how to formulate the correct “<action application=”set” …>” and “<condition>” sections in the configuration.
>>>  
>>> Thanks in advance for all answers.
>>>  
>>> Best Regards,
>>> Lars
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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



From abaci64 at gmail.com  Mon Apr 23 14:46:42 2018
From: abaci64 at gmail.com (Abaci B)
Date: Mon, 23 Apr 2018 10:46:42 -0400
Subject: [Freeswitch-users] Question about large confeerence
In-Reply-To: <CAC4JBEAami6epx2hL4OjzguwsOdvnXSFDgnUOPHF9mEp1PRsbg@mail.gmail.com>
References: <CAFmZFiOZmdxtbYQJU9Ye1su0aRShqDWU+T1-j0iS32yy6pKeiw@mail.gmail.com>
 <CAC4JBEAami6epx2hL4OjzguwsOdvnXSFDgnUOPHF9mEp1PRsbg@mail.gmail.com>
Message-ID: <CAFmZFiME_ckW_vD-AY9+dry+eaT2s1rLW0pn_O8Ax7Nmd+Afkg@mail.gmail.com>

Agree that mod_conference should probably be optimized to only loop unmuted
members, but that only applies to the input thread the thread handling the
output still needs to exist for everybody (except maybe for deaf members)
but it would still be a savings since right now each conference member uses
at least 3 threads if i understand correctly (1 for the channel, 1 for the
input and another one for the output).
the most optimal solution would probably be to either use shoutcast to
stream for conferences where the latency isn't critical, and develop a
streaming option for conferences, maybe something like
conference_stream://conf_name that would be like local_stream and not like
additional conference members with high overhead. I hope to 1 day either
develop or sponsor development of such a feature, if there are others
interested in the same functionality we maybe able to sponsor it together,
I would offer $500 as a starter.

On Mon, Apr 23, 2018 at 10:28 AM, Duvid Rottenberg <adrottenberg at gmail.com>
wrote:

> I have a similar scenario, and my solution was to create several
> conferences. I have one master conference, that serves to connect all
> sub-conferences. So when a user calls in to the conference I add him to a
> sub conference, then I do an outdial to an extension that joins the sub
> conference to the master conference (this connection should not be muted).
> Once the sub conference has 250 participants, I create a new sub
> conference. I control all of this via ESL, this also requires some
> additional logic to handle conference commands, as I need to make sure it
> propogates to all sub conferences.
>
> The advantage of this approach is that instead of having one loop running
> on 1 CPU core looping through all listeners we now have multiple conference
> loops running on different cores. The issue with large conferences is that
> the conference loop runs every 20 ms by default and the main conference
> loop is not (and probably can't be) multi-threaded.
>
> Since on any large conference, there will most likely be only a handful of
> speakers, the best solution would probably be to optimize mod_conference to
> have a separate list of unmuted participants and only loop through those,
> but I was afraid to undertake such a change myself.
>
>
>
> On Thu, Apr 19, 2018 at 12:09 PM, Abaci B <abaci64 at gmail.com> wrote:
>
>> Hi,
>> I'm trying to set up a large audio conference of over 1000 users where
>> only 1 or 2 users are talking and the rest are just listening, but after
>> 600-700 users the CPU usage gets too high and the audio gets choppy, I
>> understand that mod_conference is not optimized for my use case but was
>> wondering if someone has any suggestions on what can be done to optimize
>> for my use case.
>> I was also thinking that maybe there is a  better alternative for what I
>> look for, such as having just the talkers on the conference and the
>> listeners should listen to it via mod_local_stream or maybe use something
>> like eavesdrop for the listeners but not sure if and how much performance I
>> would gain.
>> Thanks for any help or input
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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 adrottenberg at gmail.com  Mon Apr 23 15:10:33 2018
From: adrottenberg at gmail.com (Duvid Rottenberg)
Date: Mon, 23 Apr 2018 11:10:33 -0400
Subject: [Freeswitch-users] Question about large confeerence
In-Reply-To: <CAFmZFiME_ckW_vD-AY9+dry+eaT2s1rLW0pn_O8Ax7Nmd+Afkg@mail.gmail.com>
References: <CAFmZFiOZmdxtbYQJU9Ye1su0aRShqDWU+T1-j0iS32yy6pKeiw@mail.gmail.com>
 <CAC4JBEAami6epx2hL4OjzguwsOdvnXSFDgnUOPHF9mEp1PRsbg@mail.gmail.com>
 <CAFmZFiME_ckW_vD-AY9+dry+eaT2s1rLW0pn_O8Ax7Nmd+Afkg@mail.gmail.com>
Message-ID: <CAC4JBEBWm1Rc2kGDM44d9_RpdSuc=Jdr3qfusUUNUD-nn=xWtw@mail.gmail.com>

I believe the main bottleneck of mod_conference is on the input thread.

Regardless of which approach you take, you still need CPU capacity to push
out the audio to all listeners. I don't think that mod_conference is any
less efficient than any other module is at doing that.

On Mon, Apr 23, 2018 at 10:46 AM, Abaci B <abaci64 at gmail.com> wrote:

> Agree that mod_conference should probably be optimized to only loop
> unmuted members, but that only applies to the input thread the thread
> handling the output still needs to exist for everybody (except maybe for
> deaf members) but it would still be a savings since right now each
> conference member uses at least 3 threads if i understand correctly (1 for
> the channel, 1 for the input and another one for the output).
> the most optimal solution would probably be to either use shoutcast to
> stream for conferences where the latency isn't critical, and develop a
> streaming option for conferences, maybe something like
> conference_stream://conf_name that would be like local_stream and not like
> additional conference members with high overhead. I hope to 1 day either
> develop or sponsor development of such a feature, if there are others
> interested in the same functionality we maybe able to sponsor it together,
> I would offer $500 as a starter.
>
> On Mon, Apr 23, 2018 at 10:28 AM, Duvid Rottenberg <adrottenberg at gmail.com
> > wrote:
>
>> I have a similar scenario, and my solution was to create several
>> conferences. I have one master conference, that serves to connect all
>> sub-conferences. So when a user calls in to the conference I add him to a
>> sub conference, then I do an outdial to an extension that joins the sub
>> conference to the master conference (this connection should not be muted).
>> Once the sub conference has 250 participants, I create a new sub
>> conference. I control all of this via ESL, this also requires some
>> additional logic to handle conference commands, as I need to make sure it
>> propogates to all sub conferences.
>>
>> The advantage of this approach is that instead of having one loop running
>> on 1 CPU core looping through all listeners we now have multiple conference
>> loops running on different cores. The issue with large conferences is that
>> the conference loop runs every 20 ms by default and the main conference
>> loop is not (and probably can't be) multi-threaded.
>>
>> Since on any large conference, there will most likely be only a handful
>> of speakers, the best solution would probably be to optimize mod_conference
>> to have a separate list of unmuted participants and only loop through
>> those, but I was afraid to undertake such a change myself.
>>
>>
>>
>> On Thu, Apr 19, 2018 at 12:09 PM, Abaci B <abaci64 at gmail.com> wrote:
>>
>>> Hi,
>>> I'm trying to set up a large audio conference of over 1000 users where
>>> only 1 or 2 users are talking and the rest are just listening, but after
>>> 600-700 users the CPU usage gets too high and the audio gets choppy, I
>>> understand that mod_conference is not optimized for my use case but was
>>> wondering if someone has any suggestions on what can be done to optimize
>>> for my use case.
>>> I was also thinking that maybe there is a  better alternative for what I
>>> look for, such as having just the talkers on the conference and the
>>> listeners should listen to it via mod_local_stream or maybe use something
>>> like eavesdrop for the listeners but not sure if and how much performance I
>>> would gain.
>>> Thanks for any help or input
>>>
>>>
>>> ____________________________________________________________
>>> _____________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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 prestonh at gmail.com  Thu Apr 19 18:23:06 2018
From: prestonh at gmail.com (Preston Hagar)
Date: Thu, 19 Apr 2018 13:23:06 -0500
Subject: [Freeswitch-users] valet_park and caller Id number
Message-ID: <CAK6zN=0X06q=Xca9CgR4VxMowsxbRHe_M2uX998wJAzDNtejfA@mail.gmail.com>

I've setup mod_valet_park to have a set of "parking slots" so that our
receptionist can park the calls in, page for someone to pickup, and then
that person can dial in and pick up.  That is all working well.

I'm using Voice Operator Panel (VOP) for the receptionist software and want
them to be able to monitor presence on the parking slots so that they can
see which ones have calls parked that haven't been picked up.  I'd also
like them to be able to see the caller id of the call parked in each slot.

VOP is monitoring presence fine, but for callerId it just shows "park" as
the number.  I traced the presence packets in the console and the problem
is that in the remote->identity display section of the notify packet, it
has "park" as the value to display, instead of the caller id number parked
there.

(see example XML below).

I have a couple of questions:

1.  Is it even possible to get mod_valet_park/freeswitch to report the
parked caller id number in presence requests in the current
mod_valet_park?  If so, how is it done?

2.  If it isn't in the current code, can anyone point me to the code block
that generates that remote XML section of the presence notification?  I
poked around in the valet_send_presence function of mod_valet_parking.c,
but can't seem to figure out how that block is built.  I will admit that
I'm still getting up to speed on how Presence works in FreeSWITCH.

Thanks for your help!

Here is a sample notify block:

<?xml version="1.0"?>
      <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="0"
state="full" entity="sip:park+8001 at pbx.myfreeswitchdomain.com">
        <dialog id="8001" direction="recipient">
         <state>confirmed</state>
          <local>
             <identity display="park">sip:8001 at pbx.myfreeswitchdomain.com
;proto=park</identity>
              <target uri="sip:8001 at pbx.myfreeswitchdomain.com;proto=park">
                <param pname="+sip.rendering" pvalue="no"/>
               </target>
            </local>
            <remote>
              <identity display="park">sip:8001</identity>
              <target uri="sip:park+8001 at pbx.myfreeswitchdomain.com"/>
            </remote>
           </dialog>
        </dialog-info>
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From andrew at convergedgroup.net  Sun Apr 22 21:10:03 2018
From: andrew at convergedgroup.net (Andrew Colin)
Date: Sun, 22 Apr 2018 23:10:03 +0200 (SAST)
Subject: [Freeswitch-users] Mod_callcentre
Message-ID: <1331122245.464888.1524431402997.JavaMail.zimbra@convergedgroup.net>

Hi Guys 

Myself and lorenzo from queuemetrics are busy starting with integration of queuemetrics into freeswitch. 

quick question. 

how does the module post its data, does it use a json api or how does it work? 

thanks 



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From mike at jerris.com  Mon Apr 23 19:28:48 2018
From: mike at jerris.com (Michael Jerris)
Date: Mon, 23 Apr 2018 15:28:48 -0400
Subject: [Freeswitch-users] How to add headers to an outgoing REGISTER
In-Reply-To: <006f01d3daff$fdf67720$f9e36560$@novatec.de>
References: <000801d3d7af$f81068f0$e8313ad0$@novatec.de>
 <CAAiAxuz6VfiJR5rro946ruQPjhGCYSXQ0MMW8vLNXb=X-DV4tA@mail.gmail.com>
 <003601d3d870$f5342b50$df9c81f0$@novatec.de>
 <9B4DBB6B-19F2-4F9F-B618-A28A5B0577CB@gmail.com>
 <006f01d3daff$fdf67720$f9e36560$@novatec.de>
Message-ID: <22FE0100-7A54-4AF6-A587-766FEAF371BC@jerris.com>

This is going to require changes to c code in mod_sofia to do properly.

> On Apr 23, 2018, at 8:38 AM, Lars Paulsen <lapa at novatec.de> wrote:
> 
> Hello Srigo,
>  
> yes, there is.
>  
> I am trying to achieve a RFC6140 compliant registration (client side).
> This is e.g. a part of the SIPConnect specification from the SIPForum.
> 
> As per RFC6140 this requires the following for a REGISTER request on the client side:
> -          include URI parameter "bnc" in Contact-Header-URI (this was simply solved using the configuration)
> -          include "gin" option tag in "Require" and "Proxy-Require" headers (that’s where I actually fail)
>  
> Thanks for looking into this.
>  
> Regards,
> Lars
>  
> Von: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von KSrigo
> Gesendet: Montag, 23. April 2018 14:29
> An: FreeSWITCH Users Help
> Betreff: Re: [Freeswitch-users] How to add headers to an outgoing REGISTER
>  
> Hi Lars,
>  
> Thats what I understood. I’m not sure you can add a header to a SIP register send by FS. Is there any reason why you want to add these headers?
>  
> srigo
>  
>  
>> On Apr 20, 2018, at 4:29 PM, Lars Paulsen <lapa at novatec.de <mailto:lapa at novatec.de>> wrote:
>>  
>> Hello Srigo,
>>  
>> yes, a gateway sending out a REGISTER. And I want to add headers to it before it’s send out.
>> I DO NOT mean adding headers to an REGISTER response send to a local extension.
>>  
>> To be precise I wanted to add “Require” and “Proxy-Require” headers.
>> But I failed on adding completely new headers.
>>  
>> Regards,
>> Lars
>>  
>> Von: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org>] Im Auftrag von Srigo Kanapathipillai
>> Gesendet: Freitag, 20. April 2018 03:40
>> An: FreeSWITCH Users Help
>> Betreff: Re: [Freeswitch-users] How to add headers to an outgoing REGISTER
>>  
>> Hi,
>>  
>> What you mean by outgoing register?a Gateway registering? 
>>  
>> If yes, I'm not sure you can do it.
>>  
>> Srigo
>>  
>> On Fri, 20 Apr 2018, 01:33 Lars Paulsen, <lapa at novatec.de <mailto:lapa at novatec.de>> wrote:
>>> Hello,
>>>  
>>> I have read the documentation under https://wiki.freeswitch.org/wiki/Sofia-SIP#Adding_Request_Headers <https://wiki.freeswitch.org/wiki/Sofia-SIP#Adding_Request_Headers> and as I understood it is possible to add
>>> Headers to SIP requests simply by changing the configuration.
>>>  
>>> But the documentation refers to the dialing plan.
>>> I tried to add headers to an outgoing REGISTER but failed.
>>>  
>>> Because the documentation refers to the dialplan and changing headers for existing calls, I am not sure if adding headers to a REGISTER request is possible or not.
>>> Can anyone please tell me if it is possible or not and if yes, how to formulate the correct “<action application=”set” …>” and “<condition>” sections in the configuration.
>>>  
>>> Thanks in advance for all answers.
>>>  
>>> Best Regards,
>>> Lars
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>>> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org <http://www.freeswitch.org/>
>>> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
>>> http://www.cluecon.com <http://www.cluecon.com/>
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>> http://www.freeswitch.org <http://www.freeswitch.org/>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org <http://www.freeswitch.org/>
>> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
>> http://www.cluecon.com <http://www.cluecon.com/>
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
>> http://www.freeswitch.org <http://www.freeswitch.org/>
>  
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From lapa at novatec.de  Tue Apr 24 06:26:44 2018
From: lapa at novatec.de (Lars Paulsen)
Date: Tue, 24 Apr 2018 08:26:44 +0200
Subject: [Freeswitch-users] How to add headers to an outgoing REGISTER
In-Reply-To: <22FE0100-7A54-4AF6-A587-766FEAF371BC@jerris.com>
References: <000801d3d7af$f81068f0$e8313ad0$@novatec.de>
 <CAAiAxuz6VfiJR5rro946ruQPjhGCYSXQ0MMW8vLNXb=X-DV4tA@mail.gmail.com>
 <003601d3d870$f5342b50$df9c81f0$@novatec.de>
 <9B4DBB6B-19F2-4F9F-B618-A28A5B0577CB@gmail.com>
 <006f01d3daff$fdf67720$f9e36560$@novatec.de>
 <22FE0100-7A54-4AF6-A587-766FEAF371BC@jerris.com>
Message-ID: <000001d3db95$37b22c30$a7168490$@novatec.de>

Ok, thanks a lot for all answers.

 

Best Regards,
Lars

 

Von: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Michael Jerris
Gesendet: Montag, 23. April 2018 21:29
An: FreeSWITCH Users Help
Betreff: Re: [Freeswitch-users] How to add headers to an outgoing REGISTER

 

This is going to require changes to c code in mod_sofia to do properly.

 

On Apr 23, 2018, at 8:38 AM, Lars Paulsen <lapa at novatec.de> wrote:

 

Hello Srigo,

 

yes, there is.

 

I am trying to achieve a RFC6140 compliant registration (client side).
This is e.g. a part of the SIPConnect specification from the SIPForum.




As per RFC6140 this requires the following for a REGISTER request on the client side:

-          include URI parameter "bnc" in Contact-Header-URI (this was simply solved using the configuration)

-          include "gin" option tag in "Require" and "Proxy-Require" headers (that’s where I actually fail)

 

Thanks for looking into this.

 

Regards,
Lars

 

Von: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von KSrigo
Gesendet: Montag, 23. April 2018 14:29
An: FreeSWITCH Users Help
Betreff: Re: [Freeswitch-users] How to add headers to an outgoing REGISTER

 

Hi Lars,

 

Thats what I understood. I’m not sure you can add a header to a SIP register send by FS. Is there any reason why you want to add these headers?

 

srigo

 

 

On Apr 20, 2018, at 4:29 PM, Lars Paulsen < <mailto:lapa at novatec.de> lapa at novatec.de> wrote:

 

Hello Srigo,

 

yes, a gateway sending out a REGISTER. And I want to add headers to it before it’s send out.
I DO NOT mean adding headers to an REGISTER response send to a local extension.

 

To be precise I wanted to add “Require” and “Proxy-Require” headers.

But I failed on adding completely new headers.

 

Regards,
Lars

 

Von: FreeSWITCH-users [ <mailto:freeswitch-users-bounces at lists.freeswitch.org> mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Srigo Kanapathipillai
Gesendet: Freitag, 20. April 2018 03:40
An: FreeSWITCH Users Help
Betreff: Re: [Freeswitch-users] How to add headers to an outgoing REGISTER

 

Hi,

 

What you mean by outgoing register?a Gateway registering? 

 

If yes, I'm not sure you can do it.

 

Srigo

 

On Fri, 20 Apr 2018, 01:33 Lars Paulsen, < <mailto:lapa at novatec.de> lapa at novatec.de> wrote:

Hello,

 

I have read the documentation under  <https://wiki.freeswitch.org/wiki/Sofia-SIP#Adding_Request_Headers> https://wiki.freeswitch.org/wiki/Sofia-SIP#Adding_Request_Headers and as I understood it is possible to add

Headers to SIP requests simply by changing the configuration.

 

But the documentation refers to the dialing plan.

I tried to add headers to an outgoing REGISTER but failed.

 

Because the documentation refers to the dialplan and changing headers for existing calls, I am not sure if adding headers to a REGISTER request is possible or not.

Can anyone please tell me if it is possible or not and if yes, how to formulate the correct “<action application=”set” …>” and “<condition>” sections in the configuration.

 

Thanks in advance for all answers.

 

Best Regards,
Lars

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

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

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

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

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

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

 

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

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

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

 

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From nhall at unixlan.com.ar  Tue Apr 24 07:37:54 2018
From: nhall at unixlan.com.ar (Normando Hall)
Date: Tue, 24 Apr 2018 04:37:54 -0300
Subject: [Freeswitch-users] Question about large confeerence
In-Reply-To: <CAC4JBEAami6epx2hL4OjzguwsOdvnXSFDgnUOPHF9mEp1PRsbg@mail.gmail.com>
References: <CAFmZFiOZmdxtbYQJU9Ye1su0aRShqDWU+T1-j0iS32yy6pKeiw@mail.gmail.com>
 <CAC4JBEAami6epx2hL4OjzguwsOdvnXSFDgnUOPHF9mEp1PRsbg@mail.gmail.com>
Message-ID: <fc899a94-4d00-5026-699c-ad45092d7c8a@unixlan.com.ar>

Hello Duvid. Sorry for reply to you and not the list

I have similar issue, but want to emulate 2 way radio through
mod_conference, that means one people speak and broadcast to all
listeners or target to only one listeners. The same but for listeners,
when speak, is listen only for the main operator (agent). Can you share
with us the core of your ESL script to handle your mod_conference behaviour?

Thank you

Regars

Normando

El 23/04/2018 a las 11:28 a.m., Duvid Rottenberg escribió:
> I have a similar scenario, and my solution was to create several
> conferences. I have one master conference, that serves to connect all
> sub-conferences. So when a user calls in to the conference I add him
> to a sub conference, then I do an outdial to an extension that joins
> the sub conference to the master conference (this connection should
> not be muted).  Once the sub conference has 250 participants, I create
> a new sub conference. I control all of this via ESL, this also
> requires some additional logic to handle conference commands, as I
> need to make sure it propogates to all sub conferences.
>
> The advantage of this approach is that instead of having one loop
> running on 1 CPU core looping through all listeners we now have
> multiple conference loops running on different cores. The issue with
> large conferences is that the conference loop runs every 20 ms by
> default and the main conference loop is not (and probably can't be)
> multi-threaded. 
>
> Since on any large conference, there will most likely be only a
> handful of speakers, the best solution would probably be to optimize
> mod_conference to have a separate list of unmuted participants and
> only loop through those, but I was afraid to undertake such a change
> myself.
>
>
>
> On Thu, Apr 19, 2018 at 12:09 PM, Abaci B <abaci64 at gmail.com
> <mailto:abaci64 at gmail.com>> wrote:
>
>     Hi,
>     I'm trying to set up a large audio conference of over 1000 users
>     where only 1 or 2 users are talking and the rest are just
>     listening, but after 600-700 users the CPU usage gets too high and
>     the audio gets choppy, I understand that mod_conference is not
>     optimized for my use case but was wondering if someone has any
>     suggestions on what can be done to optimize for my use case.
>     I was also thinking that maybe there is a  better alternative for
>     what I look for, such as having just the talkers on the conference
>     and the listeners should listen to it via mod_local_stream or
>     maybe use something like eavesdrop for the listeners but not sure
>     if and how much performance I would gain.
>     Thanks for any help or input
>
>
>     _________________________________________________________________________
>     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
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 ssinyagin at gmail.com  Tue Apr 24 19:28:18 2018
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Tue, 24 Apr 2018 21:28:18 +0200
Subject: [Freeswitch-users] SIMCOM LTE modems with USB Audio
Message-ID: <CACOYK=hNTVPScbo_OGo-aO=ryuS-YbAaDb7o_nhbUi4YNKGdFw@mail.gmail.com>

hi,

I wrote about a year ago about SIMCOM SIM7100 series LTE modems and
that they support USB audio. Now the vendor offers also SIM7500 and
SIM7600 modems, also with USB audio, and in addition these newer
models support 16-bit audio streams.

Unfortunately I didn't find the free time to make a FreeSWITCH module
for them, but if someone wants to sponsor such a development, or
actually develop it, I'll be glad to work together.

I've got a SIM7100E modem, and SIM7500E is ordered and is on the way.

The SIM7500 series is cheaper, but it only supports 10Mbps downstream
for mobile data. The SIM7600 supports up to 150Mbps. But for voice
applications, data transfer rate should be irrelevant in any way.


cheers,
stanislav


From hartnett.tom at gmail.com  Tue Apr 24 20:00:51 2018
From: hartnett.tom at gmail.com (Tom Hartnett)
Date: Tue, 24 Apr 2018 16:00:51 -0400
Subject: [Freeswitch-users] SIMCOM LTE modems with USB Audio
In-Reply-To: <CACOYK=hNTVPScbo_OGo-aO=ryuS-YbAaDb7o_nhbUi4YNKGdFw@mail.gmail.com>
References: <CACOYK=hNTVPScbo_OGo-aO=ryuS-YbAaDb7o_nhbUi4YNKGdFw@mail.gmail.com>
Message-ID: <CAPWmf4tEJvaoqpsEVbueRH_QcOb9+nV-bCmm0H+CV7dhCYyE+Q@mail.gmail.com>

Funny you mention the Simcom as I'm testing the 7600AH now (unrelated to
Freeswitch). The modems are odd in that they don't create a USB audio
endpoint in the traditional sense. You have to move the PCM voice over
their UART interface. That requires a little extra work as I haven't really
found a good driver to manage that.

I'm really interested in these because I want to support VoLTE on AT&T. We
haven't got that running yet, and my AT&T contact says there was an issue
with provisioning this model in their database. Hopefully that will be
fixed later this week.

On Tue, Apr 24, 2018 at 3:28 PM, Stanislav Sinyagin <ssinyagin at gmail.com>
wrote:

> hi,
>
> I wrote about a year ago about SIMCOM SIM7100 series LTE modems and
> that they support USB audio. Now the vendor offers also SIM7500 and
> SIM7600 modems, also with USB audio, and in addition these newer
> models support 16-bit audio streams.
>
> Unfortunately I didn't find the free time to make a FreeSWITCH module
> for them, but if someone wants to sponsor such a development, or
> actually develop it, I'll be glad to work together.
>
> I've got a SIM7100E modem, and SIM7500E is ordered and is on the way.
>
> The SIM7500 series is cheaper, but it only supports 10Mbps downstream
> for mobile data. The SIM7600 supports up to 150Mbps. But for voice
> applications, data transfer rate should be irrelevant in any way.
>
>
> cheers,
> stanislav
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 abaci64 at gmail.com  Tue Apr 24 20:06:55 2018
From: abaci64 at gmail.com (Abaci B)
Date: Tue, 24 Apr 2018 16:06:55 -0400
Subject: [Freeswitch-users] Fwd:  Voicemail via web interface
In-Reply-To: <191c3a031001281442m2b74ebb5xffa19113f54f0948@mail.gmail.com>
References: <4B61ACB3.50903@acsol.net>
 <090DDB5F675E4FC48512CA3FC52FA096@greyhawk.tonecommander.com>
 <4B61EAE4.2070607@acsol.net>
 <191c3a031001281442m2b74ebb5xffa19113f54f0948@mail.gmail.com>
Message-ID: <CAFmZFiOquf9SEs9PXv41HiCVRaPSLRnp1hAFU3Tzsu38dHW7wQ@mail.gmail.com>

is there any setting in freeswitch to make mod_vociemail record mp3 in 11khz

---------- Forwarded message ----------
From: Anthony Minessale <anthony.minessale at gmail.com>
Date: Thu, Jan 28, 2010 at 5:42 PM
Subject: Re: [Freeswitch-users] Voicemail via web interface
To: freeswitch-users at lists.freeswitch.org


yes sadly mp3 up sampled to 11khz is the only thing that works with that
flash player.



On Thu, Jan 28, 2010 at 1:52 PM, John <john at acsol.net> wrote:

> Thanks Robert. I believe the issue is probably because our files are in
> WAV format and not MP3.
>
> On 1/28/2010 12:28 PM, Robert Hadley wrote:
> > Using Firefox I was asked to install the latest Flash plugin and then I
> > could play the messages from the webpage directly.  IE8 never asked to
> add
> > the plugin that I noticed.
> > -RobertH
> >
> >
> > -----Original Message-----
> > From: John [mailto:john at acsol.net]
> > Sent: Thursday, January 28, 2010 7:27 AM
> > To: freeswitch-users at lists.freeswitch.org
> > Subject: [Freeswitch-users] Voicemail via web interface
> >
> > Hello,
> > Can you point me to any additional information about the voice mail via
> > web interface? I have it up and running; however if you click the play
> > button there is no playback, if you click download it will play in MS
> > media player. Thanks John
> >
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900

_______________________________________________
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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From jprangi at gmail.com  Tue Apr 24 21:14:37 2018
From: jprangi at gmail.com (Jai Rangi)
Date: Tue, 24 Apr 2018 14:14:37 -0700
Subject: [Freeswitch-users] Linphone Softphone crashing freeswitch
Message-ID: <CAPfjz3gVu9qY7F8MqGQz=-R6YLTu-=efGTjQ_GmRu2PmcDY4tg@mail.gmail.com>

I have noticed Linphone client app can easily crash free-switch. Noticed
few restarts right after. These logs


d239a46b-2c2b-4ca9-9efc-1984b41e333f 2018-04-24 13:13:18.128573 [WARNING]
switch_rtp.c:1092 missed 2
6c1e0ae1-dbe2-4630-8d8d-3cd20ba7e639 2018-04-24 13:13:18.348574 [ERR]
switch_rtp.c:930 Invalid STUN/ICE packet received 28 bytes
6c1e0ae1-dbe2-4630-8d8d-3cd20ba7e639 2018-04-24 13:13:18.368567 [ERR]
switch_rtp.c:930 Invalid STUN/ICE packet received 28 bytes
d239a46b-2c2b-4ca9-9efc-1984b41e333f 2018-04-24 13:13:18.548572 [WARNING]
switch_rtp.c:1092 missed 3
d239a46b-2c2b-4ca9-9efc-1984b41e333f 2018-04-24 13:13:18.668567 [WARNING]
switch_rtp.c:1092 missed 1
de1df2ac-d136-4c24-9a4c-6a16f55a9422 2018-04-24 13:13:18.688574 [ALERT]
switch_core_media.c:445 Looking for zrtp-hash
de1df2ac-d136-4c24-9a4c-6a16f55a9422 2018-04-24 13:13:18.688574 [ALERT]
switch_core_media.c:400 Deciding whether to pass zrtp-hash between legs
de1df2ac-d136-4c24-9a4c-6a16f55a9422 2018-04-24 13:13:18.688574 [ALERT]
switch_core_media.c:408 Found peer channel; propagating zrtp-hash if set
de1df2ac-d136-4c24-9a4c-6a16f55a9422 2018-04-24 13:13:18.688574 [ALERT]
switch_core_media.c:323 Deciding whether to pass zrtp-hash between a-leg
and b-leg
de1df2ac-d136-4c24-9a4c-6a16f55a9422 2018-04-24 13:13:18.688574 [ALERT]
switch_core_media.c:323 Deciding whether to pass zrtp-hash between a-leg
and b-leg

And freeswitch restarts,

de1df2ac-d136-4c24-9a4c-6a16f55a9422 2018-04-24 13:13:18.968579 [DEBUG]
switch_rtp.c:7229 Correct audio ip/port confirmed.
2018-04-24 13:13:19.576345 [CONSOLE] switch_loadable_module.c:1540
Successfully Loaded [mod_logfile]
2018-04-24 13:13:19.576605 [CONSOLE] switch_loadable_module.c:1540
Successfully Loaded [mod_blacklist]
2018-04-24 13:13:19.576617 [NOTICE] switch_loadable_module.c:338 Adding API
Function 'blacklist'
2018-04-24 13:13:19.577958 [CONSOLE] switch_loadable_module.c:1540
Successfully Loaded [mod_enum]
2018-04-24 13:13:19.577969 [NOTICE] switch_loadable_module.c:250 Adding
Dialplan 'enum'
2018-04-24 13:13:19.577987 [NOTICE] switch_loadable_module.c:292 Adding
Application 'enum'
2018-04-24 13:13:19.578009 [NOTICE] switch_loadable_module.c:338 Adding API
Function 'enum'
2018-04-24 13:13:19.578029 [NOTICE] switch_loadable_module.c:338 Adding API
Function 'enum_auto'


Is there any setting to ignore ICE and STUN packets.
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From jprangi at gmail.com  Tue Apr 24 00:32:23 2018
From: jprangi at gmail.com (Jai Rangi)
Date: Mon, 23 Apr 2018 17:32:23 -0700
Subject: [Freeswitch-users] Memory Leak in 1.6.20
Message-ID: <CAPfjz3gA1AgxXymCcASWghkuWwLfY7zKiF9McDWqf1_cd0p4oA@mail.gmail.com>

Not 100 % sure, but I noticed memory leak in version 1.6.20, anyone else
noticed same? Just want to check out before I create a bug report.

My quick test was on aws compute t2.medium with 4 GB RAM. Downgrading back
to 1.6.16 seems to be working fine.

Thank you,
-Jai
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From michael at mailworks.org  Wed Apr 25 00:10:29 2018
From: michael at mailworks.org (Michael Avers)
Date: Tue, 24 Apr 2018 17:10:29 -0700
Subject: [Freeswitch-users] Implementing customer callback with
	mod_callcenter
Message-ID: <1524615029.611966.1349675520.0DE35BAB@webmail.messagingengine.com>

Hello,

Looking for some best practices.. I'm trying to let users exit a mod_callcenter queue to be placed in a callback queue that will (ideally) place them first in line to be connected to an agent when one becomes available.

One way I had in mind is to disconnect the caller once we saved the fact that they requested a callback (application logic, doesn't matter here). Then, originate a call that plays in a loop "press 1 to connect" that is added to the relevant mod_callcenter queue with a base score that places it first in line to be picked up as the first priority. When the call is picked up the an agent, they hear the "press 1" playback, and once they actually press 1, we execute a lua script that calls the user that requested the callback and bridge agent and user.

In theory this makes perfect sense . Any thoughts? Is there a simpler solution I am overlooking?

Thanks
Mike


From gmina at connectfirst.com  Wed Apr 25 00:30:10 2018
From: gmina at connectfirst.com (Geoff Mina)
Date: Wed, 25 Apr 2018 00:30:10 +0000
Subject: [Freeswitch-users] Trouble Shooting Hung Sofia Profile
Message-ID: <CAKzROs29Ti4F=wMeFy1mGTZYwnd03oApmhJhthfOFYRLhrnJVg@mail.gmail.com>

Anyone have any pointers on the best way to troubleshoot hung Sofia
profiles?

We are on 1.6.19 installed via Yum on CentOS 7 and have had consistent
system hangs every few days. You can see the SIP message come in on the
siptrace logs and then there is never a response.

Process restart required to correct. CLI is functional. Nothing interesting
in logs.

Any pointers greatly appreciated.

Thanks.
-- 
GEOFF MINA
Chief Executive Officer
Connect First / Contact Center Solutions, Built Better.

2545 Central Ave #200, Boulder, CO 80301
720.335.5924
Connect First / Contact Center Solutions, Built Better
www.connectfirst.com

This email and any files transmitted with it are confidential and are
intended solely for the use of the individual or entity to whom they are
addressed. If you have received this email in error, please notify the
system manager.
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From tony at intelecenter.com  Wed Apr 25 02:40:20 2018
From: tony at intelecenter.com (Tony Bourdeaux)
Date: Tue, 24 Apr 2018 19:40:20 -0700
Subject: [Freeswitch-users] Trouble Shooting Hung Sofia Profile
In-Reply-To: <CAKzROs29Ti4F=wMeFy1mGTZYwnd03oApmhJhthfOFYRLhrnJVg@mail.gmail.com>
References: <CAKzROs29Ti4F=wMeFy1mGTZYwnd03oApmhJhthfOFYRLhrnJVg@mail.gmail.com>
Message-ID: <CAANhKXLsO0gXDOz8Nikmy02GP4HyF2xg2J-rV0oEWFuQNeLj3w@mail.gmail.com>

​Geoff-

saw this behavior when using  mod_xml_curl for Directory and had many
authentication requests to Directory service ​that was very slow to
respond.  Sofia stopped responding to any new requests. Fixed with caching
the Directory requests.

might help.

Thanks.

On Tue, Apr 24, 2018 at 5:30 PM, Geoff Mina <gmina at connectfirst.com> wrote:

> Anyone have any pointers on the best way to troubleshoot hung Sofia
> profiles?
>
> We are on 1.6.19 installed via Yum on CentOS 7 and have had consistent
> system hangs every few days. You can see the SIP message come in on the
> siptrace logs and then there is never a response.
>
> Process restart required to correct. CLI is functional. Nothing
> interesting in logs.
>
> Any pointers greatly appreciated.
>
> Thanks.
> --
> GEOFF MINA
> Chief Executive Officer
> Connect First / Contact Center Solutions, Built Better.
>
> 2545 Central Ave #200, Boulder, CO 80301
> 720.335.5924
> Connect First / Contact Center Solutions, Built Better
> www.connectfirst.com
>
> This email and any files transmitted with it are confidential and are
> intended solely for the use of the individual or entity to whom they are
> addressed. If you have received this email in error, please notify the
> system manager.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 


  *T**ony Bourdeaux*



*   Intelecenter, LLC*

   ph: 805-703-8277

   Skype: tony.bourdeaux



"This message and any attachments are solely for the intended recipient and
may contain confidential or privileged information. If you are not the
intended recipient, any disclosure, copying, use, or distribution of the
information included in this message and any attachments is prohibited. If
you have received this communication in error, please notify me by reply
e-mail and immediately and permanently delete this message and any
attachments."
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From gmina at connectfirst.com  Wed Apr 25 03:05:36 2018
From: gmina at connectfirst.com (Geoff Mina)
Date: Tue, 24 Apr 2018 21:05:36 -0600
Subject: [Freeswitch-users] Trouble Shooting Hung Sofia Profile
In-Reply-To: <CAANhKXLsO0gXDOz8Nikmy02GP4HyF2xg2J-rV0oEWFuQNeLj3w@mail.gmail.com>
References: <CAKzROs29Ti4F=wMeFy1mGTZYwnd03oApmhJhthfOFYRLhrnJVg@mail.gmail.com>
 <CAANhKXLsO0gXDOz8Nikmy02GP4HyF2xg2J-rV0oEWFuQNeLj3w@mail.gmail.com>
Message-ID: <81FBEEE6-EF7B-4C06-B08D-A0279264B52E@connectfirst.com>

First, thanks a lot for the response we use mod_xml_curl very heavily. While I don’t think our local host Apache server is lagging, it’s possible this related.  Any tips on caching directory requests? 

I don’t see anything directly related to mod_xml_curl that enables cache. 

Thanks,
Geoff 




> On Apr 24, 2018, at 8:40 PM, Tony Bourdeaux <tony at intelecenter.com> wrote:
> 
> ​Geoff-
> 
> saw this behavior when using  mod_xml_curl for Directory and had many authentication requests to Directory service ​that was very slow to respond.  Sofia stopped responding to any new requests. Fixed with caching the Directory requests.
> 
> might help.
> 
> Thanks.
> 
>> On Tue, Apr 24, 2018 at 5:30 PM, Geoff Mina <gmina at connectfirst.com> wrote:
>> Anyone have any pointers on the best way to troubleshoot hung Sofia profiles?
>> 
>> We are on 1.6.19 installed via Yum on CentOS 7 and have had consistent system hangs every few days. You can see the SIP message come in on the siptrace logs and then there is never a response. 
>> 
>> Process restart required to correct. CLI is functional. Nothing interesting in logs. 
>> 
>> Any pointers greatly appreciated. 
>> 
>> Thanks. 
>> -- 
>> GEOFF MINA
>> Chief Executive Officer
>> Connect First / Contact Center Solutions, Built Better.
>> 
>> 
>> 2545 Central Ave #200, Boulder, CO 80301
>> 720.335.5924
>> Connect First / Contact Center Solutions, Built Better
>> www.connectfirst.com
>> 
>> 
>> This email and any files transmitted with it are confidential and are intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error, please notify the system manager. 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> 
> -- 
> 
>   Tony Bourdeaux
> 
> 
>    Intelecenter, LLC
>    ph: 805-703-8277
>    Skype: tony.bourdeaux
>  
> "This message and any attachments are solely for the intended recipient and may contain confidential or privileged information. If you are not the intended recipient, any disclosure, copying, use, or distribution of the information included in this message and any attachments is prohibited. If you have received this communication in error, please notify me by reply e-mail and immediately and permanently delete this message and any attachments."
>  
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 tony at intelecenter.com  Wed Apr 25 03:48:27 2018
From: tony at intelecenter.com (Tony Bourdeaux)
Date: Tue, 24 Apr 2018 20:48:27 -0700
Subject: [Freeswitch-users] Trouble Shooting Hung Sofia Profile
In-Reply-To: <81FBEEE6-EF7B-4C06-B08D-A0279264B52E@connectfirst.com>
References: <CAKzROs29Ti4F=wMeFy1mGTZYwnd03oApmhJhthfOFYRLhrnJVg@mail.gmail.com>
 <CAANhKXLsO0gXDOz8Nikmy02GP4HyF2xg2J-rV0oEWFuQNeLj3w@mail.gmail.com>
 <81FBEEE6-EF7B-4C06-B08D-A0279264B52E@connectfirst.com>
Message-ID: <CAANhKXLxNBLuh8PqVCiGTVoEHgy-dP3qb7Dn+Hb-SM6bSbNW1w@mail.gmail.com>

Geoff-

in Directory for a user can set cacheable=true like this:

<user id='1000' cacheable='true'>
     <params>
         <param name='password' value='password123'/>
     </params>
     <variables>
      <variable name='toll_allow' value='domestic,international,local'/>
      <variable name='accountcode' value='1000'/>
      <variable name='user_context' value='defaukt'/>
      <variable name='effective_caller_id_name' value='Extension 1000'/>
      <variable name='effective_caller_id_number' value='1000'/>
      <variable name='outbound_caller_id_name'
value='$${outbound_caller_name}'/>
      <variable name='outbound_caller_id_number'
value='$${outbound_caller_id}'/>
     </variables>
    </user>

See this link:
https://freeswitch.org/confluence/display/FREESWITCH/mod_xml_curl#Caching_objects

And then the user is stored in memory so re-register authentication comes
from memory rather than another lookup.

If the user is deleted/disabled or changes password then the cache should
be cleared for that user. Like this:  xml_flush_cache id 1000 domainname

See below for reference:

If you then make a change to the directory you should run the
xml_flush_cache command to clear, some examples:

# This clears items for user 1001
xml_flush_cache id 1002 domain-name

# This clears all items
xml_flush_cache

You may also wish to enable an expiry on the cache entries by setting the
cacheable attribute to a numeric value which would be the number of
milliseconds, see http://jira.freeswitch.org/browse/FS-4871

      <user id="1002" cacheable="60000">

will cache the result for 60 seconds (60000 milliseconds)


*ALSO-*

Preloading in the config files also reduces requests:

In freeswitch.xml replace the section for dialplan with a pre-process wget
statement to load the dialplan:


*Something like this:*

<section name="dialplan" description="Regex/XML Dialplan">

 <X-PRE-PROCESS cmd="exec" data="wget --post-data
'SomePOSTVariable=SomePOSTValue' -q --output-document=-
https://someURLtheReturnsDialplan/api/ICSwitch/ICDialplan" />


 <!-- (Important: Note the ' - '.  This instructs wget to output the
retrieved document to stdout.) -->
 <!-- (to test: "wget -post-data 'SomePOSTVariable=SomePOSTValue'
https://someURLtheReturnsDialplan/api/ICSwitch/ICDialplan") -->

     <!--  <##X-PRE-PROCESS cmd="include" data="dialplan/*.xml"/>  -->

  </section>


Hope this helps.

Thanks,

Tony

On Tue, Apr 24, 2018 at 8:05 PM, Geoff Mina <gmina at connectfirst.com> wrote:

> First, thanks a lot for the response we use mod_xml_curl very heavily.
> While I don’t think our local host Apache server is lagging, it’s possible
> this related.  Any tips on caching directory requests?
>
> I don’t see anything directly related to mod_xml_curl that enables cache.
>
> Thanks,
> Geoff
>
>
>
>
> On Apr 24, 2018, at 8:40 PM, Tony Bourdeaux <tony at intelecenter.com> wrote:
>
> ​Geoff-
>
> saw this behavior when using  mod_xml_curl for Directory and had many
> authentication requests to Directory service ​that was very slow to
> respond.  Sofia stopped responding to any new requests. Fixed with caching
> the Directory requests.
>
> might help.
>
> Thanks.
>
> On Tue, Apr 24, 2018 at 5:30 PM, Geoff Mina <gmina at connectfirst.com>
> wrote:
>
>> Anyone have any pointers on the best way to troubleshoot hung Sofia
>> profiles?
>>
>> We are on 1.6.19 installed via Yum on CentOS 7 and have had consistent
>> system hangs every few days. You can see the SIP message come in on the
>> siptrace logs and then there is never a response.
>>
>> Process restart required to correct. CLI is functional. Nothing
>> interesting in logs.
>>
>> Any pointers greatly appreciated.
>>
>> Thanks.
>> --
>> GEOFF MINA
>> Chief Executive Officer
>> Connect First / Contact Center Solutions, Built Better.
>>
>> 2545 Central Ave #200, Boulder, CO 80301
>> 720.335.5924
>> Connect First / Contact Center Solutions, Built Better
>> www.connectfirst.com
>>
>> This email and any files transmitted with it are confidential and are
>> intended solely for the use of the individual or entity to whom they are
>> addressed. If you have received this email in error, please notify the
>> system manager.
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
>
>
>   *T**ony Bourdeaux*
>
>
>
> *   Intelecenter, LLC*
>
>    ph: 805-703-8277
>
>    Skype: tony.bourdeaux
>
>
>
> "This message and any attachments are solely for the intended recipient
> and may contain confidential or privileged information. If you are not the
> intended recipient, any disclosure, copying, use, or distribution of the
> information included in this message and any attachments is prohibited. If
> you have received this communication in error, please notify me by reply
> e-mail and immediately and permanently delete this message and any
> attachments."
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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
>



-- 


  *T**ony Bourdeaux*



*   Intelecenter, LLC*

   ph: 805-703-8277

   Skype: tony.bourdeaux



"This message and any attachments are solely for the intended recipient and
may contain confidential or privileged information. If you are not the
intended recipient, any disclosure, copying, use, or distribution of the
information included in this message and any attachments is prohibited. If
you have received this communication in error, please notify me by reply
e-mail and immediately and permanently delete this message and any
attachments."
-------------- next part --------------
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From gmaruzz at gmail.com  Wed Apr 25 04:27:59 2018
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Wed, 25 Apr 2018 06:27:59 +0200
Subject: [Freeswitch-users] Trouble Shooting Hung Sofia Profile
In-Reply-To: <CAANhKXLxNBLuh8PqVCiGTVoEHgy-dP3qb7Dn+Hb-SM6bSbNW1w@mail.gmail.com>
References: <CAKzROs29Ti4F=wMeFy1mGTZYwnd03oApmhJhthfOFYRLhrnJVg@mail.gmail.com>
 <CAANhKXLsO0gXDOz8Nikmy02GP4HyF2xg2J-rV0oEWFuQNeLj3w@mail.gmail.com>
 <81FBEEE6-EF7B-4C06-B08D-A0279264B52E@connectfirst.com>
 <CAANhKXLxNBLuh8PqVCiGTVoEHgy-dP3qb7Dn+Hb-SM6bSbNW1w@mail.gmail.com>
Message-ID: <CALXCt0rbLPiWjGLYgY4OVhYt4OJ+kW7KnqZjs5yu8PWF991o-Q@mail.gmail.com>

Tony, Geoff,

registrations are served by one sofia thread, by default, and if the
backend keep the registration from succeeding, that can badly affect things.

You may want to look into this:
inbound-reg-in-new-thread

For each inbound register, launch a new thread to process it, e.g. for when
using heavier backends

<param name="inbound-reg-in-new-thread" value="true"/>


https://freeswitch.org/confluence/display/FREESWITCH/
Sofia+Configuration+Files

-giovanni


On 25 April 2018 at 05:48, Tony Bourdeaux <tony at intelecenter.com> wrote:

> Geoff-
>
> in Directory for a user can set cacheable=true like this:
>
> <user id='1000' cacheable='true'>
>      <params>
>          <param name='password' value='password123'/>
>      </params>
>      <variables>
>       <variable name='toll_allow' value='domestic,international,local'/>
>       <variable name='accountcode' value='1000'/>
>       <variable name='user_context' value='defaukt'/>
>       <variable name='effective_caller_id_name' value='Extension 1000'/>
>       <variable name='effective_caller_id_number' value='1000'/>
>       <variable name='outbound_caller_id_name'
> value='$${outbound_caller_name}'/>
>       <variable name='outbound_caller_id_number'
> value='$${outbound_caller_id}'/>
>      </variables>
>     </user>
>
> See this link:  https://freeswitch.org/confluence/display/FREESWITCH/mod_
> xml_curl#Caching_objects
>
> And then the user is stored in memory so re-register authentication comes
> from memory rather than another lookup.
>
> If the user is deleted/disabled or changes password then the cache should
> be cleared for that user. Like this:  xml_flush_cache id 1000 domainname
>
> See below for reference:
>
> If you then make a change to the directory you should run the
> xml_flush_cache command to clear, some examples:
>
> # This clears items for user 1001
> xml_flush_cache id 1002 domain-name
>
> # This clears all items
> xml_flush_cache
>
> You may also wish to enable an expiry on the cache entries by setting the
> cacheable attribute to a numeric value which would be the number of
> milliseconds, see http://jira.freeswitch.org/browse/FS-4871
>
>       <user id="1002" cacheable="60000">
>
> will cache the result for 60 seconds (60000 milliseconds)
>
>
> *ALSO-*
>
> Preloading in the config files also reduces requests:
>
> In freeswitch.xml replace the section for dialplan with a pre-process wget
> statement to load the dialplan:
>
>
> *Something like this:*
>
> <section name="dialplan" description="Regex/XML Dialplan">
>
>  <X-PRE-PROCESS cmd="exec" data="wget --post-data
> 'SomePOSTVariable=SomePOSTValue' -q --output-document=-
> https://someURLtheReturnsDialplan/api/ICSwitch/ICDialplan" />
>
>
>  <!-- (Important: Note the ' - '.  This instructs wget to output the
> retrieved document to stdout.) -->
>  <!-- (to test: "wget -post-data 'SomePOSTVariable=SomePOSTValue'
> https://someURLtheReturnsDialplan/api/ICSwitch/ICDialplan") -->
>
>      <!--  <##X-PRE-PROCESS cmd="include" data="dialplan/*.xml"/>  -->
>
>   </section>
>
>
> Hope this helps.
>
> Thanks,
>
> Tony
>
> On Tue, Apr 24, 2018 at 8:05 PM, Geoff Mina <gmina at connectfirst.com>
> wrote:
>
>> First, thanks a lot for the response we use mod_xml_curl very heavily.
>> While I don’t think our local host Apache server is lagging, it’s possible
>> this related.  Any tips on caching directory requests?
>>
>> I don’t see anything directly related to mod_xml_curl that enables cache.
>>
>> Thanks,
>> Geoff
>>
>>
>>
>>
>> On Apr 24, 2018, at 8:40 PM, Tony Bourdeaux <tony at intelecenter.com>
>> wrote:
>>
>> ​Geoff-
>>
>> saw this behavior when using  mod_xml_curl for Directory and had many
>> authentication requests to Directory service ​that was very slow to
>> respond.  Sofia stopped responding to any new requests. Fixed with caching
>> the Directory requests.
>>
>> might help.
>>
>> Thanks.
>>
>> On Tue, Apr 24, 2018 at 5:30 PM, Geoff Mina <gmina at connectfirst.com>
>> wrote:
>>
>>> Anyone have any pointers on the best way to troubleshoot hung Sofia
>>> profiles?
>>>
>>> We are on 1.6.19 installed via Yum on CentOS 7 and have had consistent
>>> system hangs every few days. You can see the SIP message come in on the
>>> siptrace logs and then there is never a response.
>>>
>>> Process restart required to correct. CLI is functional. Nothing
>>> interesting in logs.
>>>
>>> Any pointers greatly appreciated.
>>>
>>> Thanks.
>>> --
>>> GEOFF MINA
>>> Chief Executive Officer
>>> Connect First / Contact Center Solutions, Built Better.
>>>
>>> 2545 Central Ave #200, Boulder, CO 80301
>>> 720.335.5924
>>> Connect First / Contact Center Solutions, Built Better
>>> www.connectfirst.com
>>>
>>> This email and any files transmitted with it are confidential and are
>>> intended solely for the use of the individual or entity to whom they are
>>> addressed. If you have received this email in error, please notify the
>>> system manager.
>>>
>>> ____________________________________________________________
>>> _____________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>>
>>
>>   *T**ony Bourdeaux*
>>
>>
>>
>> *   Intelecenter, LLC*
>>
>>    ph: 805-703-8277
>>
>>    Skype: tony.bourdeaux
>>
>>
>>
>> "This message and any attachments are solely for the intended recipient
>> and may contain confidential or privileged information. If you are not the
>> intended recipient, any disclosure, copying, use, or distribution of the
>> information included in this message and any attachments is prohibited. If
>> you have received this communication in error, please notify me by reply
>> e-mail and immediately and permanently delete this message and any
>> attachments."
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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
>>
>
>
>
> --
>
>
>   *T**ony Bourdeaux*
>
>
>
> *   Intelecenter, LLC*
>
>    ph: 805-703-8277
>
>    Skype: tony.bourdeaux
>
>
>
> "This message and any attachments are solely for the intended recipient
> and may contain confidential or privileged information. If you are not the
> intended recipient, any disclosure, copying, use, or distribution of the
> information included in this message and any attachments is prohibited. If
> you have received this communication in error, please notify me by reply
> e-mail and immediately and permanently delete this message and any
> attachments."
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

Sincerely,

Giovanni Maruzzelli
OpenTelecom.IT
cell: +39 347 266 56 18
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From asilva at wirelessmundi.com  Wed Apr 25 07:36:37 2018
From: asilva at wirelessmundi.com (antonio)
Date: Wed, 25 Apr 2018 09:36:37 +0200
Subject: [Freeswitch-users] Linphone Softphone crashing freeswitch
In-Reply-To: <CAPfjz3gVu9qY7F8MqGQz=-R6YLTu-=efGTjQ_GmRu2PmcDY4tg@mail.gmail.com>
References: <CAPfjz3gVu9qY7F8MqGQz=-R6YLTu-=efGTjQ_GmRu2PmcDY4tg@mail.gmail.com>
Message-ID: <b671e9b5-a97d-a931-f8a7-1b2f3d98da68@wirelessmundi.com>

the best place to post issues is jira, please check:

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



I think your issue might be solved: 
https://freeswitch.org/jira/browse/FS-10394


On 04/24/2018 11:14 PM, Jai Rangi wrote:
>
> I have noticed Linphone client app can easily crash free-switch. 
> Noticed few restarts right after. These logs
>
>
> d239a46b-2c2b-4ca9-9efc-1984b41e333f 2018-04-24 13:13:18.128573 
> [WARNING] switch_rtp.c:1092 missed 2
> 6c1e0ae1-dbe2-4630-8d8d-3cd20ba7e639 2018-04-24 13:13:18.348574 [ERR] 
> switch_rtp.c:930 Invalid STUN/ICE packet received 28 bytes
> 6c1e0ae1-dbe2-4630-8d8d-3cd20ba7e639 2018-04-24 13:13:18.368567 [ERR] 
> switch_rtp.c:930 Invalid STUN/ICE packet received 28 bytes
> d239a46b-2c2b-4ca9-9efc-1984b41e333f 2018-04-24 13:13:18.548572 
> [WARNING] switch_rtp.c:1092 missed 3
> d239a46b-2c2b-4ca9-9efc-1984b41e333f 2018-04-24 13:13:18.668567 
> [WARNING] switch_rtp.c:1092 missed 1
> de1df2ac-d136-4c24-9a4c-6a16f55a9422 2018-04-24 13:13:18.688574 
> [ALERT] switch_core_media.c:445 Looking for zrtp-hash
> de1df2ac-d136-4c24-9a4c-6a16f55a9422 2018-04-24 13:13:18.688574 
> [ALERT] switch_core_media.c:400 Deciding whether to pass zrtp-hash 
> between legs
> de1df2ac-d136-4c24-9a4c-6a16f55a9422 2018-04-24 13:13:18.688574 
> [ALERT] switch_core_media.c:408 Found peer channel; propagating 
> zrtp-hash if set
> de1df2ac-d136-4c24-9a4c-6a16f55a9422 2018-04-24 13:13:18.688574 
> [ALERT] switch_core_media.c:323 Deciding whether to pass zrtp-hash 
> between a-leg and b-leg
> de1df2ac-d136-4c24-9a4c-6a16f55a9422 2018-04-24 13:13:18.688574 
> [ALERT] switch_core_media.c:323 Deciding whether to pass zrtp-hash 
> between a-leg and b-leg
>
> And freeswitch restarts,
>
> de1df2ac-d136-4c24-9a4c-6a16f55a9422 2018-04-24 13:13:18.968579 
> [DEBUG] switch_rtp.c:7229 Correct audio ip/port confirmed.
> 2018-04-24 13:13:19.576345 [CONSOLE] switch_loadable_module.c:1540 
> Successfully Loaded [mod_logfile]
> 2018-04-24 13:13:19.576605 [CONSOLE] switch_loadable_module.c:1540 
> Successfully Loaded [mod_blacklist]
> 2018-04-24 13:13:19.576617 [NOTICE] switch_loadable_module.c:338 
> Adding API Function 'blacklist'
> 2018-04-24 13:13:19.577958 [CONSOLE] switch_loadable_module.c:1540 
> Successfully Loaded [mod_enum]
> 2018-04-24 13:13:19.577969 [NOTICE] switch_loadable_module.c:250 
> Adding Dialplan 'enum'
> 2018-04-24 13:13:19.577987 [NOTICE] switch_loadable_module.c:292 
> Adding Application 'enum'
> 2018-04-24 13:13:19.578009 [NOTICE] switch_loadable_module.c:338 
> Adding API Function 'enum'
> 2018-04-24 13:13:19.578029 [NOTICE] switch_loadable_module.c:338 
> Adding API Function 'enum_auto'
>
>
> Is there any setting to ignore ICE and STUN packets.
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Saludos / Regards / Cumprimentos
Anónio Silva

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From ssinyagin at gmail.com  Wed Apr 25 11:49:32 2018
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Wed, 25 Apr 2018 11:49:32 +0000
Subject: [Freeswitch-users] SIMCOM LTE modems with USB Audio
In-Reply-To: <CAPWmf4tEJvaoqpsEVbueRH_QcOb9+nV-bCmm0H+CV7dhCYyE+Q@mail.gmail.com>
References: <CACOYK=hNTVPScbo_OGo-aO=ryuS-YbAaDb7o_nhbUi4YNKGdFw@mail.gmail.com>
 <CAPWmf4tEJvaoqpsEVbueRH_QcOb9+nV-bCmm0H+CV7dhCYyE+Q@mail.gmail.com>
Message-ID: <CACOYK=gWzsD__m2bowvN0Cmf_P+43h3P5VkNnTTUn3y9AJn=Yg@mail.gmail.com>

Yes, it's clearly written in their documentation, that raw PCM is sent in
an USB UART device.

Mod_gsmopen is too imperfect, to say politely. It's easier to create a new
module for UART audio than trying to adapt gsmopen for new modems.

So, it only needs a budget for development.





On Tue, Apr 24, 2018, 22:01 Tom Hartnett <hartnett.tom at gmail.com> wrote:

> Funny you mention the Simcom as I'm testing the 7600AH now (unrelated to
> Freeswitch). The modems are odd in that they don't create a USB audio
> endpoint in the traditional sense. You have to move the PCM voice over
> their UART interface. That requires a little extra work as I haven't really
> found a good driver to manage that.
>
> I'm really interested in these because I want to support VoLTE on AT&T. We
> haven't got that running yet, and my AT&T contact says there was an issue
> with provisioning this model in their database. Hopefully that will be
> fixed later this week.
>
> On Tue, Apr 24, 2018 at 3:28 PM, Stanislav Sinyagin <ssinyagin at gmail.com>
> wrote:
>
>> hi,
>>
>> I wrote about a year ago about SIMCOM SIM7100 series LTE modems and
>> that they support USB audio. Now the vendor offers also SIM7500 and
>> SIM7600 modems, also with USB audio, and in addition these newer
>> models support 16-bit audio streams.
>>
>> Unfortunately I didn't find the free time to make a FreeSWITCH module
>> for them, but if someone wants to sponsor such a development, or
>> actually develop it, I'll be glad to work together.
>>
>> I've got a SIM7100E modem, and SIM7500E is ordered and is on the way.
>>
>> The SIM7500 series is cheaper, but it only supports 10Mbps downstream
>> for mobile data. The SIM7600 supports up to 150Mbps. But for voice
>> applications, data transfer rate should be irrelevant in any way.
>>
>>
>> cheers,
>> stanislav
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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 gmina at connectfirst.com  Wed Apr 25 13:19:08 2018
From: gmina at connectfirst.com (Geoff Mina)
Date: Wed, 25 Apr 2018 07:19:08 -0600
Subject: [Freeswitch-users] Trouble Shooting Hung Sofia Profile
In-Reply-To: <CALXCt0rbLPiWjGLYgY4OVhYt4OJ+kW7KnqZjs5yu8PWF991o-Q@mail.gmail.com>
References: <CAKzROs29Ti4F=wMeFy1mGTZYwnd03oApmhJhthfOFYRLhrnJVg@mail.gmail.com>
 <CAANhKXLsO0gXDOz8Nikmy02GP4HyF2xg2J-rV0oEWFuQNeLj3w@mail.gmail.com>
 <81FBEEE6-EF7B-4C06-B08D-A0279264B52E@connectfirst.com>
 <CAANhKXLxNBLuh8PqVCiGTVoEHgy-dP3qb7Dn+Hb-SM6bSbNW1w@mail.gmail.com>
 <CALXCt0rbLPiWjGLYgY4OVhYt4OJ+kW7KnqZjs5yu8PWF991o-Q@mail.gmail.com>
Message-ID: <CAKzROs1GGnerNz5M77pT73Nf8-YwgEzgqwB6C16kSdu1xnjRiQ@mail.gmail.com>

Thanks to both of you for the additional information.  We have enabled
caching, multi-threaded-registrations, as well as a timeout on the
mod_xml_curl configuration.

Hopefully this resolves our issues going forward.

Thanks,
Geoff



On Tue, Apr 24, 2018 at 10:27 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>
wrote:

> Tony, Geoff,
>
> registrations are served by one sofia thread, by default, and if the
> backend keep the registration from succeeding, that can badly affect things.
>
> You may want to look into this:
> inbound-reg-in-new-thread
>
> For each inbound register, launch a new thread to process it, e.g. for
> when using heavier backends
>
> <param name="inbound-reg-in-new-thread" value="true"/>
>
>
> https://freeswitch.org/confluence/display/FREESWITCH/Sofia+
> Configuration+Files
>
> -giovanni
>
>
> On 25 April 2018 at 05:48, Tony Bourdeaux <tony at intelecenter.com> wrote:
>
>> Geoff-
>>
>> in Directory for a user can set cacheable=true like this:
>>
>> <user id='1000' cacheable='true'>
>>      <params>
>>          <param name='password' value='password123'/>
>>      </params>
>>      <variables>
>>       <variable name='toll_allow' value='domestic,international,local'/>
>>       <variable name='accountcode' value='1000'/>
>>       <variable name='user_context' value='defaukt'/>
>>       <variable name='effective_caller_id_name' value='Extension 1000'/>
>>       <variable name='effective_caller_id_number' value='1000'/>
>>       <variable name='outbound_caller_id_name'
>> value='$${outbound_caller_name}'/>
>>       <variable name='outbound_caller_id_number'
>> value='$${outbound_caller_id}'/>
>>      </variables>
>>     </user>
>>
>> See this link:  https://freeswitch.org/conflu
>> ence/display/FREESWITCH/mod_xml_curl#Caching_objects
>>
>> And then the user is stored in memory so re-register authentication comes
>> from memory rather than another lookup.
>>
>> If the user is deleted/disabled or changes password then the cache should
>> be cleared for that user. Like this:  xml_flush_cache id 1000 domainname
>>
>> See below for reference:
>>
>> If you then make a change to the directory you should run the
>> xml_flush_cache command to clear, some examples:
>>
>> # This clears items for user 1001
>> xml_flush_cache id 1002 domain-name
>>
>> # This clears all items
>> xml_flush_cache
>>
>> You may also wish to enable an expiry on the cache entries by setting the
>> cacheable attribute to a numeric value which would be the number of
>> milliseconds, see http://jira.freeswitch.org/browse/FS-4871
>>
>>       <user id="1002" cacheable="60000">
>>
>> will cache the result for 60 seconds (60000 milliseconds)
>>
>>
>> *ALSO-*
>>
>> Preloading in the config files also reduces requests:
>>
>> In freeswitch.xml replace the section for dialplan with a pre-process
>> wget statement to load the dialplan:
>>
>>
>> *Something like this:*
>>
>> <section name="dialplan" description="Regex/XML Dialplan">
>>
>>  <X-PRE-PROCESS cmd="exec" data="wget --post-data
>> 'SomePOSTVariable=SomePOSTValue' -q --output-document=-
>> https://someURLtheReturnsDialplan/api/ICSwitch/ICDialplan" />
>>
>>
>>  <!-- (Important: Note the ' - '.  This instructs wget to output the
>> retrieved document to stdout.) -->
>>  <!-- (to test: "wget -post-data 'SomePOSTVariable=SomePOSTValue'
>> https://someURLtheReturnsDialplan/api/ICSwitch/ICDialplan") -->
>>
>>      <!--  <##X-PRE-PROCESS cmd="include" data="dialplan/*.xml"/>  -->
>>
>>   </section>
>>
>>
>> Hope this helps.
>>
>> Thanks,
>>
>> Tony
>>
>> On Tue, Apr 24, 2018 at 8:05 PM, Geoff Mina <gmina at connectfirst.com>
>> wrote:
>>
>>> First, thanks a lot for the response we use mod_xml_curl very heavily.
>>> While I don’t think our local host Apache server is lagging, it’s possible
>>> this related.  Any tips on caching directory requests?
>>>
>>> I don’t see anything directly related to mod_xml_curl that enables
>>> cache.
>>>
>>> Thanks,
>>> Geoff
>>>
>>>
>>>
>>>
>>> On Apr 24, 2018, at 8:40 PM, Tony Bourdeaux <tony at intelecenter.com>
>>> wrote:
>>>
>>> ​Geoff-
>>>
>>> saw this behavior when using  mod_xml_curl for Directory and had many
>>> authentication requests to Directory service ​that was very slow to
>>> respond.  Sofia stopped responding to any new requests. Fixed with caching
>>> the Directory requests.
>>>
>>> might help.
>>>
>>> Thanks.
>>>
>>> On Tue, Apr 24, 2018 at 5:30 PM, Geoff Mina <gmina at connectfirst.com>
>>> wrote:
>>>
>>>> Anyone have any pointers on the best way to troubleshoot hung Sofia
>>>> profiles?
>>>>
>>>> We are on 1.6.19 installed via Yum on CentOS 7 and have had consistent
>>>> system hangs every few days. You can see the SIP message come in on the
>>>> siptrace logs and then there is never a response.
>>>>
>>>> Process restart required to correct. CLI is functional. Nothing
>>>> interesting in logs.
>>>>
>>>> Any pointers greatly appreciated.
>>>>
>>>> Thanks.
>>>> --
>>>> GEOFF MINA
>>>> Chief Executive Officer
>>>> Connect First / Contact Center Solutions, Built Better.
>>>>
>>>> 2545 Central Ave #200, Boulder, CO 80301
>>>> 720.335.5924
>>>> Connect First / Contact Center Solutions, Built Better
>>>> www.connectfirst.com
>>>>
>>>> This email and any files transmitted with it are confidential and are
>>>> intended solely for the use of the individual or entity to whom they are
>>>> addressed. If you have received this email in error, please notify the
>>>> system manager.
>>>>
>>>> ____________________________________________________________
>>>> _____________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free
>>>> switch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> --
>>>
>>>
>>>   *T**ony Bourdeaux*
>>>
>>>
>>>
>>> *   Intelecenter, LLC*
>>>
>>>    ph: 805-703-8277
>>>
>>>    Skype: tony.bourdeaux
>>>
>>>
>>>
>>> "This message and any attachments are solely for the intended recipient
>>> and may contain confidential or privileged information. If you are not the
>>> intended recipient, any disclosure, copying, use, or distribution of the
>>> information included in this message and any attachments is prohibited. If
>>> you have received this communication in error, please notify me by reply
>>> e-mail and immediately and permanently delete this message and any
>>> attachments."
>>>
>>>
>>>
>>> ____________________________________________________________
>>> _____________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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
>>>
>>
>>
>>
>> --
>>
>>
>>   *T**ony Bourdeaux*
>>
>>
>>
>> *   Intelecenter, LLC*
>>
>>    ph: 805-703-8277
>>
>>    Skype: tony.bourdeaux
>>
>>
>>
>> "This message and any attachments are solely for the intended recipient
>> and may contain confidential or privileged information. If you are not the
>> intended recipient, any disclosure, copying, use, or distribution of the
>> information included in this message and any attachments is prohibited. If
>> you have received this communication in error, please notify me by reply
>> e-mail and immediately and permanently delete this message and any
>> attachments."
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
>
> Sincerely,
>
> Giovanni Maruzzelli
> OpenTelecom.IT
> cell: +39 347 266 56 18
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From tg-maillistings at level5.de  Wed Apr 25 13:23:03 2018
From: tg-maillistings at level5.de (=?UTF-8?Q?Thorsten_G=c3=b6llner?=)
Date: Wed, 25 Apr 2018 15:23:03 +0200
Subject: [Freeswitch-users] mod_say / switch_xml.c:2486 Invalid UTF-8
 character to ampersand, skip it
Message-ID: <8ce7a7bc-7e4f-4338-5bf7-508427757882@level5.de>

Hi,

I am using Freeswitch 1.6.19~64bit on CentOs 7 (with FusionPBX 4.2.2) -
working fine so far. But when calling a mailbox, I can hear the greeting
but can not hear the destination number. I can find this macro is used:

<macro name="voicemail_play_greeting">
  <input pattern="^(.*)$">
    <match>
      <action function="play-file" data="voicemail/vm-person.wav"/>
      <action function="say" data="$1" method="pronounced"
type="name_spelled"/>
      <action function="play-file" data="voicemail/vm-not_available.wav"/>
    </match>
  </input>
</macro>

This is leading me to track it down to the function "say".

So I configured a simple inbound destination which answers the call and
just makes "say : de number pronounced 12345". When calling this number
I get the following error:

EXECUTE sofia/internal/12 at mydomain.cloud say(de number pronounced 12345)
2018-04-25 15:07:09.732930 [DEBUG] freeswitch_lua.cpp:365 DBH handle
0x7f33b405a700 Connected.
2018-04-25 15:07:09.732930 [DEBUG] freeswitch_lua.cpp:382 DBH handle
0x7f33b405a700 released.
2018-04-25 15:07:09.732930 [WARNING] switch_xml.c:2486 Invalid UTF-8
character to ampersand, skip it
2018-04-25 15:07:09.732930 [WARNING] switch_xml.c:2486 Invalid UTF-8
character to ampersand, skip it
2018-04-25 15:07:09.732930 [WARNING] switch_xml.c:2486 Invalid UTF-8
character to ampersand, skip it
2018-04-25 15:07:09.732930 [WARNING] switch_xml.c:2486 Invalid UTF-8
character to ampersand, skip it
2018-04-25 15:07:09.732930 [ERR] switch_xml.c:3180 Can't find phrases tag.
EXECUTE sofia/internal/12 at mydomain.cloud hangup()

I checked the source code but my coding skill is not good enough. I do
not really know, where the problem is.

Do you have any hint for me?

Best regards, Thorsten



From tg-maillistings at level5.de  Wed Apr 25 14:59:38 2018
From: tg-maillistings at level5.de (=?UTF-8?Q?Thorsten_G=c3=b6llner?=)
Date: Wed, 25 Apr 2018 16:59:38 +0200
Subject: [Freeswitch-users] mod_say / switch_xml.c:2486 Invalid UTF-8
 character to ampersand, skip it
In-Reply-To: <8ce7a7bc-7e4f-4338-5bf7-508427757882@level5.de>
References: <8ce7a7bc-7e4f-4338-5bf7-508427757882@level5.de>
Message-ID: <ee2d7ae9-88c2-8455-fa81-cf7ae3ce0561@level5.de>

Hi,

I "converted" all xml-files under ${conf_dir}/lang/ to utf-8 and added
the following line to the top of each file:
<?xml version="1.0" encoding="utf-8"?>

Now the warnings about invalid utf-8-characters vanished.

But I still face the error:
[ERR] switch_xml.c:3180 Can't find phrases tag
and I will not hear the dialed extension number (mailbox number).

Because I use FusionPBX I examined the voicemail-lua-app and found the
following:

--the person at extension 101 is not available record your message at
the tone press any key or stop talking to end the recording
if (name == "person_not_available_record_message") then
        table.insert(actions,
{app="streamFile",data="voicemail/vm-person.wav"});
        --pronounce the voicemail_id
        if (voicemail_alternate_greet_id and
string.len(voicemail_alternate_greet_id) > 0) then
                table.insert(actions,
{app="say.number.iterated",data=voicemail_alternate_greet_id});
        elseif (voicemail_greet_id and string.len(voicemail_greet_id) >
0) then
                table.insert(actions,
{app="say.number.iterated",data=voicemail_greet_id});
        else
                table.insert(actions,
{app="say.number.iterated",data=voicemail_id});
        end
        table.insert(actions,
{app="streamFile",data="voicemail/vm-not_available.wav"});
end

So I thought the problem was "data=voicemail_id". I insertet a
Log-Output and got the following:
voicemail_id=13

And yes - I called extension 13.

But the question remains: Why does this lead to "[ERR] switch_xml.c:3180
Can't find phrases tag."?

Thorsten

Am 25.04.2018 um 15:23 schrieb Thorsten Göllner:
> Hi,
>
> I am using Freeswitch 1.6.19~64bit on CentOs 7 (with FusionPBX 4.2.2) -
> working fine so far. But when calling a mailbox, I can hear the greeting
> but can not hear the destination number. I can find this macro is used:
>
> <macro name="voicemail_play_greeting">
>   <input pattern="^(.*)$">
>     <match>
>       <action function="play-file" data="voicemail/vm-person.wav"/>
>       <action function="say" data="$1" method="pronounced"
> type="name_spelled"/>
>       <action function="play-file" data="voicemail/vm-not_available.wav"/>
>     </match>
>   </input>
> </macro>
>
> This is leading me to track it down to the function "say".
>
> So I configured a simple inbound destination which answers the call and
> just makes "say : de number pronounced 12345". When calling this number
> I get the following error:
>
> EXECUTE sofia/internal/12 at mydomain.cloud say(de number pronounced 12345)
> 2018-04-25 15:07:09.732930 [DEBUG] freeswitch_lua.cpp:365 DBH handle
> 0x7f33b405a700 Connected.
> 2018-04-25 15:07:09.732930 [DEBUG] freeswitch_lua.cpp:382 DBH handle
> 0x7f33b405a700 released.
> 2018-04-25 15:07:09.732930 [WARNING] switch_xml.c:2486 Invalid UTF-8
> character to ampersand, skip it
> 2018-04-25 15:07:09.732930 [WARNING] switch_xml.c:2486 Invalid UTF-8
> character to ampersand, skip it
> 2018-04-25 15:07:09.732930 [WARNING] switch_xml.c:2486 Invalid UTF-8
> character to ampersand, skip it
> 2018-04-25 15:07:09.732930 [WARNING] switch_xml.c:2486 Invalid UTF-8
> character to ampersand, skip it
> 2018-04-25 15:07:09.732930 [ERR] switch_xml.c:3180 Can't find phrases tag.
> EXECUTE sofia/internal/12 at mydomain.cloud hangup()
>
> I checked the source code but my coding skill is not good enough. I do
> not really know, where the problem is.
>
> Do you have any hint for me?
>
> Best regards, Thorsten
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From tg-maillistings at level5.de  Wed Apr 25 15:28:04 2018
From: tg-maillistings at level5.de (=?UTF-8?Q?Thorsten_G=c3=b6llner?=)
Date: Wed, 25 Apr 2018 17:28:04 +0200
Subject: [Freeswitch-users] mod_say / switch_xml.c:2486 Invalid UTF-8
 character to ampersand, skip it
In-Reply-To: <ee2d7ae9-88c2-8455-fa81-cf7ae3ce0561@level5.de>
References: <8ce7a7bc-7e4f-4338-5bf7-508427757882@level5.de>
 <ee2d7ae9-88c2-8455-fa81-cf7ae3ce0561@level5.de>
Message-ID: <f1332d7d-a23c-e8f5-d96e-fc21ceebaaa5@level5.de>

SOLVED:

The problem only occured with german language. I copied a
xml-config-file from the voice-prompt-provider to the config directory
and did not check it. There was an error within the config file. Now it
works fine.

The correct config file look like this:

<?xml version="1.0" encoding="utf-8"?>
<include>
  <language name="de" sound-path="$${sounds_dir}/de/de/marilda"
tts-engine="cepstral" tts-voice="callie">
    <phrases>
      <macros>
        <X-PRE-PROCESS cmd="include" data="demo/*.xml"/> <!-- Note: this
now grabs whole subdir, previously grabbed only demo.xml -->
        <!--voicemail_en_tts is purely implemented with tts, we have the
files based one that is the default. -->
        <X-PRE-PROCESS cmd="include" data="vm/sounds.xml"/>  <!--
vm/tts.xml if you want to use tts and have cepstral -->
        <X-PRE-PROCESS cmd="include" data="dir/sounds.xml"/>  <!--
dir/tts.xml if you want to use tts and have cepstral -->
      </macros>
    </phrases>
  </language>
</include>

Best regards,
Thorsten

Am 25.04.2018 um 16:59 schrieb Thorsten Göllner:
> Hi,
>
> I "converted" all xml-files under ${conf_dir}/lang/ to utf-8 and added
> the following line to the top of each file:
> <?xml version="1.0" encoding="utf-8"?>
>
> Now the warnings about invalid utf-8-characters vanished.
>
> But I still face the error:
> [ERR] switch_xml.c:3180 Can't find phrases tag
> and I will not hear the dialed extension number (mailbox number).
>
> Because I use FusionPBX I examined the voicemail-lua-app and found the
> following:
>
> --the person at extension 101 is not available record your message at
> the tone press any key or stop talking to end the recording
> if (name == "person_not_available_record_message") then
>         table.insert(actions,
> {app="streamFile",data="voicemail/vm-person.wav"});
>         --pronounce the voicemail_id
>         if (voicemail_alternate_greet_id and
> string.len(voicemail_alternate_greet_id) > 0) then
>                 table.insert(actions,
> {app="say.number.iterated",data=voicemail_alternate_greet_id});
>         elseif (voicemail_greet_id and string.len(voicemail_greet_id) >
> 0) then
>                 table.insert(actions,
> {app="say.number.iterated",data=voicemail_greet_id});
>         else
>                 table.insert(actions,
> {app="say.number.iterated",data=voicemail_id});
>         end
>         table.insert(actions,
> {app="streamFile",data="voicemail/vm-not_available.wav"});
> end
>
> So I thought the problem was "data=voicemail_id". I insertet a
> Log-Output and got the following:
> voicemail_id=13
>
> And yes - I called extension 13.
>
> But the question remains: Why does this lead to "[ERR] switch_xml.c:3180
> Can't find phrases tag."?
>
> Thorsten
>
> Am 25.04.2018 um 15:23 schrieb Thorsten Göllner:
>> Hi,
>>
>> I am using Freeswitch 1.6.19~64bit on CentOs 7 (with FusionPBX 4.2.2) -
>> working fine so far. But when calling a mailbox, I can hear the greeting
>> but can not hear the destination number. I can find this macro is used:
>>
>> <macro name="voicemail_play_greeting">
>>   <input pattern="^(.*)$">
>>     <match>
>>       <action function="play-file" data="voicemail/vm-person.wav"/>
>>       <action function="say" data="$1" method="pronounced"
>> type="name_spelled"/>
>>       <action function="play-file" data="voicemail/vm-not_available.wav"/>
>>     </match>
>>   </input>
>> </macro>
>>
>> This is leading me to track it down to the function "say".
>>
>> So I configured a simple inbound destination which answers the call and
>> just makes "say : de number pronounced 12345". When calling this number
>> I get the following error:
>>
>> EXECUTE sofia/internal/12 at mydomain.cloud say(de number pronounced 12345)
>> 2018-04-25 15:07:09.732930 [DEBUG] freeswitch_lua.cpp:365 DBH handle
>> 0x7f33b405a700 Connected.

>> 2018-04-25 15:07:09.732930 [DEBUG] freeswitch_lua.cpp:382 DBH handle
>> 0x7f33b405a700 released.
>> 2018-04-25 15:07:09.732930 [WARNING] switch_xml.c:2486 Invalid UTF-8
>> character to ampersand, skip it
>> 2018-04-25 15:07:09.732930 [WARNING] switch_xml.c:2486 Invalid UTF-8
>> character to ampersand, skip it
>> 2018-04-25 15:07:09.732930 [WARNING] switch_xml.c:2486 Invalid UTF-8
>> character to ampersand, skip it
>> 2018-04-25 15:07:09.732930 [WARNING] switch_xml.c:2486 Invalid UTF-8
>> character to ampersand, skip it
>> 2018-04-25 15:07:09.732930 [ERR] switch_xml.c:3180 Can't find phrases tag.
>> EXECUTE sofia/internal/12 at mydomain.cloud hangup()
>>
>> I checked the source code but my coding skill is not good enough. I do
>> not really know, where the problem is.
>>
>> Do you have any hint for me?
>>
>> Best regards, Thorsten
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>




From tony at intelecenter.com  Wed Apr 25 15:57:48 2018
From: tony at intelecenter.com (Tony Bourdeaux)
Date: Wed, 25 Apr 2018 08:57:48 -0700
Subject: [Freeswitch-users] Trouble Shooting Hung Sofia Profile
In-Reply-To: <CALXCt0rbLPiWjGLYgY4OVhYt4OJ+kW7KnqZjs5yu8PWF991o-Q@mail.gmail.com>
References: <CAKzROs29Ti4F=wMeFy1mGTZYwnd03oApmhJhthfOFYRLhrnJVg@mail.gmail.com>
 <CAANhKXLsO0gXDOz8Nikmy02GP4HyF2xg2J-rV0oEWFuQNeLj3w@mail.gmail.com>
 <81FBEEE6-EF7B-4C06-B08D-A0279264B52E@connectfirst.com>
 <CAANhKXLxNBLuh8PqVCiGTVoEHgy-dP3qb7Dn+Hb-SM6bSbNW1w@mail.gmail.com>
 <CALXCt0rbLPiWjGLYgY4OVhYt4OJ+kW7KnqZjs5yu8PWF991o-Q@mail.gmail.com>
Message-ID: <CAANhKXKvaDhHQ_1WaSZ2otqL-tfk1D2Z8nUz5qrpoP5jjvqYcQ@mail.gmail.com>

Thanks Giovanni

On Tue, Apr 24, 2018 at 9:27 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>
wrote:

> Tony, Geoff,
>
> registrations are served by one sofia thread, by default, and if the
> backend keep the registration from succeeding, that can badly affect things.
>
> You may want to look into this:
> inbound-reg-in-new-thread
>
> For each inbound register, launch a new thread to process it, e.g. for
> when using heavier backends
>
> <param name="inbound-reg-in-new-thread" value="true"/>
>
>
> https://freeswitch.org/confluence/display/FREESWITCH/Sofia+
> Configuration+Files
>
> -giovanni
>
>
> On 25 April 2018 at 05:48, Tony Bourdeaux <tony at intelecenter.com> wrote:
>
>> Geoff-
>>
>> in Directory for a user can set cacheable=true like this:
>>
>> <user id='1000' cacheable='true'>
>>      <params>
>>          <param name='password' value='password123'/>
>>      </params>
>>      <variables>
>>       <variable name='toll_allow' value='domestic,international,local'/>
>>       <variable name='accountcode' value='1000'/>
>>       <variable name='user_context' value='defaukt'/>
>>       <variable name='effective_caller_id_name' value='Extension 1000'/>
>>       <variable name='effective_caller_id_number' value='1000'/>
>>       <variable name='outbound_caller_id_name'
>> value='$${outbound_caller_name}'/>
>>       <variable name='outbound_caller_id_number'
>> value='$${outbound_caller_id}'/>
>>      </variables>
>>     </user>
>>
>> See this link:  https://freeswitch.org/conflu
>> ence/display/FREESWITCH/mod_xml_curl#Caching_objects
>>
>> And then the user is stored in memory so re-register authentication comes
>> from memory rather than another lookup.
>>
>> If the user is deleted/disabled or changes password then the cache should
>> be cleared for that user. Like this:  xml_flush_cache id 1000 domainname
>>
>> See below for reference:
>>
>> If you then make a change to the directory you should run the
>> xml_flush_cache command to clear, some examples:
>>
>> # This clears items for user 1001
>> xml_flush_cache id 1002 domain-name
>>
>> # This clears all items
>> xml_flush_cache
>>
>> You may also wish to enable an expiry on the cache entries by setting the
>> cacheable attribute to a numeric value which would be the number of
>> milliseconds, see http://jira.freeswitch.org/browse/FS-4871
>>
>>       <user id="1002" cacheable="60000">
>>
>> will cache the result for 60 seconds (60000 milliseconds)
>>
>>
>> *ALSO-*
>>
>> Preloading in the config files also reduces requests:
>>
>> In freeswitch.xml replace the section for dialplan with a pre-process
>> wget statement to load the dialplan:
>>
>>
>> *Something like this:*
>>
>> <section name="dialplan" description="Regex/XML Dialplan">
>>
>>  <X-PRE-PROCESS cmd="exec" data="wget --post-data
>> 'SomePOSTVariable=SomePOSTValue' -q --output-document=-
>> https://someURLtheReturnsDialplan/api/ICSwitch/ICDialplan" />
>>
>>
>>  <!-- (Important: Note the ' - '.  This instructs wget to output the
>> retrieved document to stdout.) -->
>>  <!-- (to test: "wget -post-data 'SomePOSTVariable=SomePOSTValue'
>> https://someURLtheReturnsDialplan/api/ICSwitch/ICDialplan") -->
>>
>>      <!--  <##X-PRE-PROCESS cmd="include" data="dialplan/*.xml"/>  -->
>>
>>   </section>
>>
>>
>> Hope this helps.
>>
>> Thanks,
>>
>> Tony
>>
>> On Tue, Apr 24, 2018 at 8:05 PM, Geoff Mina <gmina at connectfirst.com>
>> wrote:
>>
>>> First, thanks a lot for the response we use mod_xml_curl very heavily.
>>> While I don’t think our local host Apache server is lagging, it’s possible
>>> this related.  Any tips on caching directory requests?
>>>
>>> I don’t see anything directly related to mod_xml_curl that enables
>>> cache.
>>>
>>> Thanks,
>>> Geoff
>>>
>>>
>>>
>>>
>>> On Apr 24, 2018, at 8:40 PM, Tony Bourdeaux <tony at intelecenter.com>
>>> wrote:
>>>
>>> ​Geoff-
>>>
>>> saw this behavior when using  mod_xml_curl for Directory and had many
>>> authentication requests to Directory service ​that was very slow to
>>> respond.  Sofia stopped responding to any new requests. Fixed with caching
>>> the Directory requests.
>>>
>>> might help.
>>>
>>> Thanks.
>>>
>>> On Tue, Apr 24, 2018 at 5:30 PM, Geoff Mina <gmina at connectfirst.com>
>>> wrote:
>>>
>>>> Anyone have any pointers on the best way to troubleshoot hung Sofia
>>>> profiles?
>>>>
>>>> We are on 1.6.19 installed via Yum on CentOS 7 and have had consistent
>>>> system hangs every few days. You can see the SIP message come in on the
>>>> siptrace logs and then there is never a response.
>>>>
>>>> Process restart required to correct. CLI is functional. Nothing
>>>> interesting in logs.
>>>>
>>>> Any pointers greatly appreciated.
>>>>
>>>> Thanks.
>>>> --
>>>> GEOFF MINA
>>>> Chief Executive Officer
>>>> Connect First / Contact Center Solutions, Built Better.
>>>>
>>>> 2545 Central Ave #200, Boulder, CO 80301
>>>> 720.335.5924
>>>> Connect First / Contact Center Solutions, Built Better
>>>> www.connectfirst.com
>>>>
>>>> This email and any files transmitted with it are confidential and are
>>>> intended solely for the use of the individual or entity to whom they are
>>>> addressed. If you have received this email in error, please notify the
>>>> system manager.
>>>>
>>>> ____________________________________________________________
>>>> _____________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free
>>>> switch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> --
>>>
>>>
>>>   *T**ony Bourdeaux*
>>>
>>>
>>>
>>> *   Intelecenter, LLC*
>>>
>>>    ph: 805-703-8277
>>>
>>>    Skype: tony.bourdeaux
>>>
>>>
>>>
>>> "This message and any attachments are solely for the intended recipient
>>> and may contain confidential or privileged information. If you are not the
>>> intended recipient, any disclosure, copying, use, or distribution of the
>>> information included in this message and any attachments is prohibited. If
>>> you have received this communication in error, please notify me by reply
>>> e-mail and immediately and permanently delete this message and any
>>> attachments."
>>>
>>>
>>>
>>> ____________________________________________________________
>>> _____________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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
>>>
>>
>>
>>
>> --
>>
>>
>>   *T**ony Bourdeaux*
>>
>>
>>
>> *   Intelecenter, LLC*
>>
>>    ph: 805-703-8277
>>
>>    Skype: tony.bourdeaux
>>
>>
>>
>> "This message and any attachments are solely for the intended recipient
>> and may contain confidential or privileged information. If you are not the
>> intended recipient, any disclosure, copying, use, or distribution of the
>> information included in this message and any attachments is prohibited. If
>> you have received this communication in error, please notify me by reply
>> e-mail and immediately and permanently delete this message and any
>> attachments."
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
>
> Sincerely,
>
> Giovanni Maruzzelli
> OpenTelecom.IT
> cell: +39 347 266 56 18
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 


  *T**ony Bourdeaux*



*   Intelecenter, LLC*

   ph: 805-703-8277

   Skype: tony.bourdeaux



"This message and any attachments are solely for the intended recipient and
may contain confidential or privileged information. If you are not the
intended recipient, any disclosure, copying, use, or distribution of the
information included in this message and any attachments is prohibited. If
you have received this communication in error, please notify me by reply
e-mail and immediately and permanently delete this message and any
attachments."
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From ssinyagin at gmail.com  Wed Apr 25 21:30:23 2018
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Wed, 25 Apr 2018 23:30:23 +0200
Subject: [Freeswitch-users] SIMCOM LTE modems with USB Audio
In-Reply-To: <CACOYK=gWzsD__m2bowvN0Cmf_P+43h3P5VkNnTTUn3y9AJn=Yg@mail.gmail.com>
References: <CACOYK=hNTVPScbo_OGo-aO=ryuS-YbAaDb7o_nhbUi4YNKGdFw@mail.gmail.com>
 <CAPWmf4tEJvaoqpsEVbueRH_QcOb9+nV-bCmm0H+CV7dhCYyE+Q@mail.gmail.com>
 <CACOYK=gWzsD__m2bowvN0Cmf_P+43h3P5VkNnTTUn3y9AJn=Yg@mail.gmail.com>
Message-ID: <CACOYK=gj_5Lz-d_ih6EsjmC3uATz-Z4dDnzPX0kyVyXKqihLQw@mail.gmail.com>

probably the most flexible option would be a small standalone daemon
that links the SIMCOM modem audio with a virtual PortAudio device.
Then you can have this stream in FreeSWITCH or in any other software
that is able to communicate with PA.



On Wed, Apr 25, 2018 at 1:49 PM, Stanislav Sinyagin <ssinyagin at gmail.com> wrote:
> Yes, it's clearly written in their documentation, that raw PCM is sent in an
> USB UART device.
>
> Mod_gsmopen is too imperfect, to say politely. It's easier to create a new
> module for UART audio than trying to adapt gsmopen for new modems.
>
> So, it only needs a budget for development.
>
>
>
>
>
> On Tue, Apr 24, 2018, 22:01 Tom Hartnett <hartnett.tom at gmail.com> wrote:
>>
>> Funny you mention the Simcom as I'm testing the 7600AH now (unrelated to
>> Freeswitch). The modems are odd in that they don't create a USB audio
>> endpoint in the traditional sense. You have to move the PCM voice over their
>> UART interface. That requires a little extra work as I haven't really found
>> a good driver to manage that.
>>
>> I'm really interested in these because I want to support VoLTE on AT&T. We
>> haven't got that running yet, and my AT&T contact says there was an issue
>> with provisioning this model in their database. Hopefully that will be fixed
>> later this week.
>>
>> On Tue, Apr 24, 2018 at 3:28 PM, Stanislav Sinyagin <ssinyagin at gmail.com>
>> wrote:
>>>
>>> hi,
>>>
>>> I wrote about a year ago about SIMCOM SIM7100 series LTE modems and
>>> that they support USB audio. Now the vendor offers also SIM7500 and
>>> SIM7600 modems, also with USB audio, and in addition these newer
>>> models support 16-bit audio streams.
>>>
>>> Unfortunately I didn't find the free time to make a FreeSWITCH module
>>> for them, but if someone wants to sponsor such a development, or
>>> actually develop it, I'll be glad to work together.
>>>
>>> I've got a SIM7100E modem, and SIM7500E is ordered and is on the way.
>>>
>>> The SIM7500 series is cheaper, but it only supports 10Mbps downstream
>>> for mobile data. The SIM7600 supports up to 150Mbps. But for voice
>>> applications, data transfer rate should be irrelevant in any way.
>>>
>>>
>>> cheers,
>>> stanislav
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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


From gmaruzz at gmail.com  Thu Apr 26 19:13:51 2018
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Thu, 26 Apr 2018 21:13:51 +0200
Subject: [Freeswitch-users] SIMCOM LTE modems with USB Audio
In-Reply-To: <CACOYK=gj_5Lz-d_ih6EsjmC3uATz-Z4dDnzPX0kyVyXKqihLQw@mail.gmail.com>
References: <CACOYK=hNTVPScbo_OGo-aO=ryuS-YbAaDb7o_nhbUi4YNKGdFw@mail.gmail.com>
 <CAPWmf4tEJvaoqpsEVbueRH_QcOb9+nV-bCmm0H+CV7dhCYyE+Q@mail.gmail.com>
 <CACOYK=gWzsD__m2bowvN0Cmf_P+43h3P5VkNnTTUn3y9AJn=Yg@mail.gmail.com>
 <CACOYK=gj_5Lz-d_ih6EsjmC3uATz-Z4dDnzPX0kyVyXKqihLQw@mail.gmail.com>
Message-ID: <CALXCt0pKBFBHV4oH8BxkEkpD30Jpoc7xkRckMO6fWLismbUYTQ@mail.gmail.com>

please do it, then let us know. Maybe do it for free, and donate it to the
community.

Thanks in advance



On 25 April 2018 at 23:30, Stanislav Sinyagin <ssinyagin at gmail.com> wrote:

> probably the most flexible option would be a small standalone daemon
> that links the SIMCOM modem audio with a virtual PortAudio device.
> Then you can have this stream in FreeSWITCH or in any other software
> that is able to communicate with PA.
>
>
>
> On Wed, Apr 25, 2018 at 1:49 PM, Stanislav Sinyagin <ssinyagin at gmail.com>
> wrote:
> > Yes, it's clearly written in their documentation, that raw PCM is sent
> in an
> > USB UART device.
> >
> > Mod_gsmopen is too imperfect, to say politely. It's easier to create a
> new
> > module for UART audio than trying to adapt gsmopen for new modems.
> >
> > So, it only needs a budget for development.
> >
> >
> >
> >
> >
> > On Tue, Apr 24, 2018, 22:01 Tom Hartnett <hartnett.tom at gmail.com> wrote:
> >>
> >> Funny you mention the Simcom as I'm testing the 7600AH now (unrelated to
> >> Freeswitch). The modems are odd in that they don't create a USB audio
> >> endpoint in the traditional sense. You have to move the PCM voice over
> their
> >> UART interface. That requires a little extra work as I haven't really
> found
> >> a good driver to manage that.
> >>
> >> I'm really interested in these because I want to support VoLTE on AT&T.
> We
> >> haven't got that running yet, and my AT&T contact says there was an
> issue
> >> with provisioning this model in their database. Hopefully that will be
> fixed
> >> later this week.
> >>
> >> On Tue, Apr 24, 2018 at 3:28 PM, Stanislav Sinyagin <
> ssinyagin at gmail.com>
> >> wrote:
> >>>
> >>> hi,
> >>>
> >>> I wrote about a year ago about SIMCOM SIM7100 series LTE modems and
> >>> that they support USB audio. Now the vendor offers also SIM7500 and
> >>> SIM7600 modems, also with USB audio, and in addition these newer
> >>> models support 16-bit audio streams.
> >>>
> >>> Unfortunately I didn't find the free time to make a FreeSWITCH module
> >>> for them, but if someone wants to sponsor such a development, or
> >>> actually develop it, I'll be glad to work together.
> >>>
> >>> I've got a SIM7100E modem, and SIM7500E is ordered and is on the way.
> >>>
> >>> The SIM7500 series is cheaper, but it only supports 10Mbps downstream
> >>> for mobile data. The SIM7600 supports up to 150Mbps. But for voice
> >>> applications, data transfer rate should be irrelevant in any way.
> >>>
> >>>
> >>> cheers,
> >>> stanislav
> >>>
> >>> ____________________________________________________________
> _____________
> >>> Professional FreeSWITCH Consulting Services:
> >>> consulting at freeswitch.org
> >>> http://www.freeswitchsolutions.com
> >>>
> >>> Official FreeSWITCH Sites
> >>> http://www.freeswitch.org
> >>> http://confluence.freeswitch.org
> >>> http://www.cluecon.com
> >>>
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/
> options/freeswitch-users
> >>> http://www.freeswitch.org
> >>
> >>
> >> ____________________________________________________________
> _____________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://confluence.freeswitch.org
> >> http://www.cluecon.com
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/
> options/freeswitch-users
> >> http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

Sincerely,

Giovanni Maruzzelli
OpenTelecom.IT
cell: +39 347 266 56 18
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From dominique.jeannerod at interact-iv.com  Thu Apr 26 21:20:23 2018
From: dominique.jeannerod at interact-iv.com (Dominique Jeannerod)
Date: Thu, 26 Apr 2018 23:20:23 +0200
Subject: [Freeswitch-users] Freeswitch 1.6.19 and T.38 /
	INCOMPATIBLE_DESTINATION
Message-ID: <CAEzwe14y6PX3Vzewjt27fMQT=rL1SGotzUzoEr-38C8=vDA36Q@mail.gmail.com>

Hello,

I currently use Freeswitch 1.6.8-1 in production, and have some ISDN
gateways sending SDP including T.38 for voice calls (not faxes). Calls are
handled without problems by freeswitch.

I have another platform, with freeswitch 1.6.19 and same configuration.
However, voice calls received with SDP including t.38 are rejected with
488 INCOMPATIBLE_DESTINATION :
ba17b1eb-961d-43fe-b820-cb469a8f292d 2018-04-25 23:47:39.434491 [DEBUG]
switch_core_media.c:4028 sofia/internal_1/33183817938 at gv-ics-prd-301 T38
REFUSE on request
ba17b1eb-961d-43fe-b820-cb469a8f292d 2018-04-25 23:47:39.434491 [NOTICE]
sofia.c:7566 Hangup sofia/internal_1/33183817938 at gv-ics-prd-301 [CS_NEW]
[INCOMPATIBLE_DESTINATION]

I didn't find any valid configuration to make those calls accepted, except
configuring proxy_media, which I don't want to use.

Is this behaviour a "normal" behaviour ? Could it be a bug ?
This is a very big issue for us ...


Here is the log extract :

recv 1391 bytes from udp/[10.9.30.152]:5060 at 23:47:39.443941:
   ------------------------------------------------------------------------
   INVITE sip:0170993660 at 10.9.30.188:5060 SIP/2.0
   Record-Route: <sip:10.9.30.152;lr=on>
   Record-Route: <sip:10.3.1.220;lr=on>
   Via: SIP/2.0/UDP 10.9.30.152;branch=z9hG4bK51d.d7e922c4.0
   Via: SIP/2.0/UDP 10.3.1.220;branch=z9hG4bK51d.ba6f34d1.0
   Via: SIP/2.0/UDP 10.3.1.231;branch=z9hG4bKac1339541470
   Max-Forwards: 68
   From: <sip:33183817938 at gv-ics-prd-301>;tag=1c1339531455
   To: <sip:0170993660 at 10.3.1.220;user=phone>
   Call-ID: 13395229682542018214736 at 10.3.1.231
   CSeq: 1 INVITE
   Contact: <sip:33183817938 at 10.3.1.231:5060>
   Supported: em,100rel,timer,replaces,path,resource-priority,sdp-anat
   Allow:
REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
   Remote-Party-ID: <sip:33183817938 at 10.3.1.220
>;party=calling;privacy=off;screen=yes;screen-ind=3;npi=1;ton=1
   Remote-Party-ID: <sip:0170993660 at 10.3.1.220>;party=called;npi=1;ton=2
   User-Agent: Audiocodes-Sip-Gateway-Mediant 3000/v.6.00A.056
   Content-Type: application/sdp
   Content-Length: 417

   v=0
   o=AudiocodesGW 1339507326 1339507042 IN IP4 10.3.1.231
   s=Phone-Call
   c=IN IP4 10.3.1.231
   t=0 0
   m=audio 23970 RTP/AVP 8 101
   a=rtpmap:8 PCMA/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-15
   a=ptime:20
   a=sendrecv
   m=image 23972 udptl t38
   a=T38FaxVersion:0
   a=T38MaxBitRate:33600
   a=T38FaxMaxBuffer:3000
   a=T38FaxMaxDatagram:560
   a=T38FaxRateManagement:transferredTCF
   a=T38FaxUdpEC:t38UDPRedundancy

   ------------------------------------------------------------------------
send 477 bytes to udp/[10.9.30.152]:5060 at 23:47:39.444372:
   ------------------------------------------------------------------------
   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP 10.9.30.152;branch=z9hG4bK51d.d7e922c4.0
   Via: SIP/2.0/UDP 10.3.1.220;branch=z9hG4bK51d.ba6f34d1.0
   Via: SIP/2.0/UDP 10.3.1.231;branch=z9hG4bKac1339541470
   Record-Route: <sip:10.9.30.152;lr=on>
   Record-Route: <sip:10.3.1.220;lr=on>
   From: <sip:33183817938 at gv-ics-prd-301>;tag=1c1339531455
   To: <sip:0170993660 at 10.3.1.220;user=phone>
   Call-ID: 13395229682542018214736 at 10.3.1.231
   CSeq: 1 INVITE
   User-Agent: FreeSWITCH
   Content-Length: 0

   ------------------------------------------------------------------------
ba17b1eb-961d-43fe-b820-cb469a8f292d 2018-04-25 23:47:39.434491 [NOTICE]
switch_channel.c:1104 New Channel
sofia/internal_1/33183817938 at gv-ics-prd-301
[ba17b1eb-961d-43fe-b820-cb469a8f292d]
ba17b1eb-961d-43fe-b820-cb469a8f292d 2018-04-25 23:47:39.434491 [DEBUG]
switch_core_state_machine.c:584 (sofia/internal_1/33183817938 at gv-ics-prd-301)
Running State Change CS_NEW (Cur 1 Tot 69)
ba17b1eb-961d-43fe-b820-cb469a8f292d 2018-04-25 23:47:39.434491 [DEBUG]
sofia.c:9874 sofia/internal_1/33183817938 at gv-ics-prd-301 receiving invite
from 10.9.30.152:5060 version: 1.6.19  64bit
ba17b1eb-961d-43fe-b820-cb469a8f292d 2018-04-25 23:47:39.434491 [DEBUG]
sofia.c:7085 Channel sofia/internal_1/33183817938 at gv-ics-prd-301 entering
state [received][100]
ba17b1eb-961d-43fe-b820-cb469a8f292d 2018-04-25 23:47:39.434491 [DEBUG]
sofia.c:7095 Remote SDP:
ba17b1eb-961d-43fe-b820-cb469a8f292d v=0^M
ba17b1eb-961d-43fe-b820-cb469a8f292d o=AudiocodesGW 1339507326 1339507042
IN IP4 10.3.1.231^M
ba17b1eb-961d-43fe-b820-cb469a8f292d s=Phone-Call^M
ba17b1eb-961d-43fe-b820-cb469a8f292d c=IN IP4 10.3.1.231^M
ba17b1eb-961d-43fe-b820-cb469a8f292d t=0 0^M
ba17b1eb-961d-43fe-b820-cb469a8f292d m=audio 23970 RTP/AVP 8 101^M
ba17b1eb-961d-43fe-b820-cb469a8f292d a=rtpmap:8 PCMA/8000^M
ba17b1eb-961d-43fe-b820-cb469a8f292d a=rtpmap:101 telephone-event/8000^M
ba17b1eb-961d-43fe-b820-cb469a8f292d a=fmtp:101 0-15^M
ba17b1eb-961d-43fe-b820-cb469a8f292d a=ptime:20^M
ba17b1eb-961d-43fe-b820-cb469a8f292d m=image 23972 udptl t38^M
ba17b1eb-961d-43fe-b820-cb469a8f292d a=T38FaxVersion:0^M
ba17b1eb-961d-43fe-b820-cb469a8f292d a=T38MaxBitRate:33600^M
ba17b1eb-961d-43fe-b820-cb469a8f292d a=T38FaxMaxBuffer:3000^M
ba17b1eb-961d-43fe-b820-cb469a8f292d a=T38FaxMaxDatagram:560^M
ba17b1eb-961d-43fe-b820-cb469a8f292d a=T38FaxRateManagement:transferredTCF^M
ba17b1eb-961d-43fe-b820-cb469a8f292d a=T38FaxUdpEC:t38UDPRedundancy^M
ba17b1eb-961d-43fe-b820-cb469a8f292d
ba17b1eb-961d-43fe-b820-cb469a8f292d 2018-04-25 23:47:39.434491 [DEBUG]
switch_core_media.c:4449 Audio Codec Compare
[PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
ba17b1eb-961d-43fe-b820-cb469a8f292d 2018-04-25 23:47:39.434491 [DEBUG]
switch_core_media.c:4504 Audio Codec Compare [PCMA:8:8000:20:64000:1] ++++
is saved as a match
ba17b1eb-961d-43fe-b820-cb469a8f292d 2018-04-25 23:47:39.434491 [DEBUG]
switch_core_media.c:4449 Audio Codec Compare
[PCMA:8:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]
ba17b1eb-961d-43fe-b820-cb469a8f292d 2018-04-25 23:47:39.434491 [DEBUG]
switch_core_media.c:4365 Set telephone-event payload to 101 at 8000
ba17b1eb-961d-43fe-b820-cb469a8f292d 2018-04-25 23:47:39.434491 [DEBUG]
switch_core_media.c:3061 Set Codec
sofia/internal_1/33183817938 at gv-ics-prd-301 PCMA/8000 20 ms 160 samples
64000 bits 1 channels
ba17b1eb-961d-43fe-b820-cb469a8f292d 2018-04-25 23:47:39.434491 [DEBUG]
switch_core_codec.c:111 sofia/internal_1/33183817938 at gv-ics-prd-301
Original read codec set to PCMA:8
ba17b1eb-961d-43fe-b820-cb469a8f292d 2018-04-25 23:47:39.434491 [DEBUG]
switch_core_media.c:4708 Set telephone-event payload to 101 at 8000
ba17b1eb-961d-43fe-b820-cb469a8f292d 2018-04-25 23:47:39.434491 [DEBUG]
switch_core_media.c:4767 sofia/internal_1/33183817938 at gv-ics-prd-301 Set
2833 dtmf send payload to 101 recv payload to 101
ba17b1eb-961d-43fe-b820-cb469a8f292d 2018-04-25 23:47:39.434491 [DEBUG]
switch_core_media.c:4028 sofia/internal_1/33183817938 at gv-ics-prd-301 T38
REFUSE on request
ba17b1eb-961d-43fe-b820-cb469a8f292d 2018-04-25 23:47:39.434491 [NOTICE]
sofia.c:7566 Hangup sofia/internal_1/33183817938 at gv-ics-prd-301 [CS_NEW]
[INCOMPATIBLE_DESTINATION]
ba17b1eb-961d-43fe-b820-cb469a8f292d 2018-04-25 23:47:39.434491 [DEBUG]
switch_core_state_machine.c:603 (sofia/internal_1/33183817938 at gv-ics-prd-301)
State NEW
ba17b1eb-961d-43fe-b820-cb469a8f292d 2018-04-25 23:47:39.434491 [DEBUG]
switch_core_state_machine.c:584 (sofia/internal_1/33183817938 at gv-ics-prd-301)
Running State Change CS_HANGUP (Cur 1 Tot 69)
ba17b1eb-961d-43fe-b820-cb469a8f292d 2018-04-25 23:47:39.434491 [DEBUG]
switch_core_state_machine.c:850 (sofia/internal_1/33183817938 at gv-ics-prd-301)
Callstate Change DOWN -> HANGUP
ba17b1eb-961d-43fe-b820-cb469a8f292d 2018-04-25 23:47:39.434491 [DEBUG]
switch_core_state_machine.c:852 (sofia/internal_1/33183817938 at gv-ics-prd-301)
State HANGUP
ba17b1eb-961d-43fe-b820-cb469a8f292d 2018-04-25 23:47:39.434491 [DEBUG]
mod_sofia.c:438 Channel sofia/internal_1/33183817938 at gv-ics-prd-301 hanging
up, cause: INCOMPATIBLE_DESTINATION
ba17b1eb-961d-43fe-b820-cb469a8f292d 2018-04-25 23:47:39.434491 [DEBUG]
mod_sofia.c:577 Responding to INVITE with: 488
ba17b1eb-961d-43fe-b820-cb469a8f292d 2018-04-25 23:47:39.434491 [DEBUG]
switch_core_state_machine.c:60 sofia/internal_1/33183817938 at gv-ics-prd-301
Standard HANGUP, cause: INCOMPATIBLE_DESTINATION
ba17b1eb-961d-43fe-b820-cb469a8f292d 2018-04-25 23:47:39.434491 [DEBUG]
switch_core_state_machine.c:852 (sofia/internal_1/33183817938 at gv-ics-prd-301)
State HANGUP going to sleep
ba17b1eb-961d-43fe-b820-cb469a8f292d 2018-04-25 23:47:39.434491 [DEBUG]
switch_core_state_machine.c:619 (sofia/internal_1/33183817938 at gv-ics-prd-301)
State Change CS_HANGUP -> CS_REPORTING
ba17b1eb-961d-43fe-b820-cb469a8f292d 2018-04-25 23:47:39.434491 [DEBUG]
switch_core_state_machine.c:584 (sofia/internal_1/33183817938 at gv-ics-prd-301)
Running State Change CS_REPORTING (Cur 1 Tot 69)
send 784 bytes to udp/[10.9.30.152]:5060 at 23:47:39.446918:
   ------------------------------------------------------------------------
   SIP/2.0 488 Not Acceptable Here
   Via: SIP/2.0/UDP 10.9.30.152;branch=z9hG4bK51d.d7e922c4.0
   Via: SIP/2.0/UDP 10.3.1.220;branch=z9hG4bK51d.ba6f34d1.0
   Via: SIP/2.0/UDP 10.3.1.231;branch=z9hG4bKac1339541470
   Max-Forwards: 68
   From: <sip:33183817938 at gv-ics-prd-301>;tag=1c1339531455
   To: <sip:0170993660 at 10.3.1.220;user=phone>;tag=9USNQNQDByZ6a
   Call-ID: 13395229682542018214736 at 10.3.1.231
   CSeq: 1 INVITE
   User-Agent: FreeSWITCH
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER,
NOTIFY
   Supported: timer, path, replaces
   Allow-Events: talk, hold, conference, refer
   Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION"
   Content-Length: 0
   Remote-Party-ID: "0170993660" <sip:0170993660 at 10.3.1.220
>;party=calling;privacy=off;screen=no


Thanks with anticipation

D. Jeannerod
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From mike at jerris.com  Thu Apr 26 21:25:38 2018
From: mike at jerris.com (Michael Jerris)
Date: Thu, 26 Apr 2018 17:25:38 -0400
Subject: [Freeswitch-users] Freeswitch 1.6.19 and T.38 /
 INCOMPATIBLE_DESTINATION
In-Reply-To: <CAEzwe14y6PX3Vzewjt27fMQT=rL1SGotzUzoEr-38C8=vDA36Q@mail.gmail.com>
References: <CAEzwe14y6PX3Vzewjt27fMQT=rL1SGotzUzoEr-38C8=vDA36Q@mail.gmail.com>
Message-ID: <1714B91B-C1F3-4D4E-B7BD-A0035E1599C6@jerris.com>

http://lists.freeswitch.org/pipermail/freeswitch-users/2017-January/124459.html <http://lists.freeswitch.org/pipermail/freeswitch-users/2017-January/124459.html>

> On Apr 26, 2018, at 5:20 PM, Dominique Jeannerod <dominique.jeannerod at interact-iv.com> wrote:
> 
> Hello,
> 
> I currently use Freeswitch 1.6.8-1 in production, and have some ISDN gateways sending SDP including T.38 for voice calls (not faxes). Calls are handled without problems by freeswitch.
> 
> I have another platform, with freeswitch 1.6.19 and same configuration.
> However, voice calls received with SDP including t.38 are rejected with 488 INCOMPATIBLE_DESTINATION :
> ba17b1eb-961d-43fe-b820-cb469a8f292d 2018-04-25 23:47:39.434491 [DEBUG] switch_core_media.c:4028 sofia/internal_1/33183817938 at gv-ics-prd-301 T38 REFUSE on request
> ba17b1eb-961d-43fe-b820-cb469a8f292d 2018-04-25 23:47:39.434491 [NOTICE] sofia.c:7566 Hangup sofia/internal_1/33183817938 at gv-ics-prd-301 [CS_NEW] [INCOMPATIBLE_DESTINATION]
> 
> I didn't find any valid configuration to make those calls accepted, except configuring proxy_media, which I don't want to use.
> 
> Is this behaviour a "normal" behaviour ? Could it be a bug ?
> This is a very big issue for us ...
> 

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From nerad.peter at gmail.com  Fri Apr 27 04:46:36 2018
From: nerad.peter at gmail.com (=?UTF-8?Q?Peter_Ner=C3=A1d?=)
Date: Fri, 27 Apr 2018 06:46:36 +0200
Subject: [Freeswitch-users] Implementing emergency calls ... or call
	priority ?
Message-ID: <!&!AAAAAAAAAAAYAAAAAAAAAI/RfDx9+axBlz43ajVi2RXCgAAAEAAAAEWQvGn+wuxGngnm+Jt2R/YBAAAAAA==@gmail.com>

Whats are best practicies to implement emergency calls ?

I can identify  such a call by regexp but how to handle those calls ?

Is there some build param to be set ?

How I can set priority for call ? e.g. I have limited capacity for outgoing and incoming call to 10. I need to drop some calls with small priority to allow emergency call, but I have no idea where to start.

Can somebody give me right direction for this ?

Thank you very much

PS: Google does not helped me :(



From sebastian_ml at gmx.net  Fri Apr 27 06:10:35 2018
From: sebastian_ml at gmx.net (Sebastian Kemper)
Date: Fri, 27 Apr 2018 06:10:35 +0000
Subject: [Freeswitch-users] Implementing emergency calls ... or
	call	priority ?
In-Reply-To: <!&!AAAAAAAAAAAYAAAAAAAAAI/RfDx9+axBlz43ajVi2RXCgAAAEAAAAEWQvGn+wuxGngnm+Jt2R/YBAAAAAA==@gmail.com>
References: <!&!AAAAAAAAAAAYAAAAAAAAAI/RfDx9+axBlz43ajVi2RXCgAAAEAAAAEWQvGn+wuxGngnm+Jt2R/YBAAAAAA==@gmail.com>
Message-ID: <E99DC39B-6D6C-4200-9035-1C99D49EAB3F@gmx.net>

Am April 27, 2018 4:46:36 AM UTC schrieb "Peter Nerád" <nerad.peter at gmail.com>:
>Whats are best practicies to implement emergency calls ?
>
>I can identify  such a call by regexp but how to handle those calls ?
>
>Is there some build param to be set ?
>
>How I can set priority for call ? e.g. I have limited capacity for
>outgoing and incoming call to 10. I need to drop some calls with small
>priority to allow emergency call, but I have no idea where to start.
>
>Can somebody give me right direction for this ?
>
>Thank you very much
>
>PS: Google does not helped me :(
>
>
>_________________________________________________________________________
>Professional FreeSWITCH Consulting Services:
>consulting at freeswitch.org
>http://www.freeswitchsolutions.com
>
>Official FreeSWITCH Sites
>http://www.freeswitch.org
>http://confluence.freeswitch.org
>http://www.cluecon.com
>
>FreeSWITCH-users mailing list
>FreeSWITCH-users at lists.freeswitch.org
>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>http://www.freeswitch.org

Hi Peter,

You can define all sorts of limits. Have a look here: https://freeswitch.org/confluence/display/FREESWITCH/mod_dptools%3A+Limit

Regards,
Sebastian


From dominique.jeannerod at interact-iv.com  Fri Apr 27 06:24:39 2018
From: dominique.jeannerod at interact-iv.com (Dominique Jeannerod)
Date: Fri, 27 Apr 2018 06:24:39 +0000
Subject: [Freeswitch-users] Freeswitch 1.6.19 and T.38 /
	INCOMPATIBLE_DESTINATION
In-Reply-To: <1714B91B-C1F3-4D4E-B7BD-A0035E1599C6@jerris.com>
References: <CAEzwe14y6PX3Vzewjt27fMQT=rL1SGotzUzoEr-38C8=vDA36Q@mail.gmail.com>
 <1714B91B-C1F3-4D4E-B7BD-A0035E1599C6@jerris.com>
Message-ID: <CAEzwe17g2zFjuS27aMS+yfb8f3Y-Qy_+Ht4t6nLNk4a77GBx6Q@mail.gmail.com>

Thanks a lot for this answer.
I already read this information mail, and read the changes done for t38
re-invite.
The problem i have is NOT on the reinvite but directly on the first invite
of the Call
Do you think the behaviour also changed for the first invite ?

Anyway, i will try the new configuration parameter, and test if it fixes my
problem.
My short term fallback plan would just be to downgrade to the 1.6.8 version
we already have on production.

Le jeu. 26 avr. 2018 à 23:52, Michael Jerris <mike at jerris.com> a écrit :

>
> http://lists.freeswitch.org/pipermail/freeswitch-users/2017-January/124459.html
>
> On Apr 26, 2018, at 5:20 PM, Dominique Jeannerod <
> dominique.jeannerod at interact-iv.com> wrote:
>
> Hello,
>
> I currently use Freeswitch 1.6.8-1 in production, and have some ISDN
> gateways sending SDP including T.38 for voice calls (not faxes). Calls are
> handled without problems by freeswitch.
>
> I have another platform, with freeswitch 1.6.19 and same configuration.
> However, voice calls received with SDP including t.38 are rejected with
> 488 INCOMPATIBLE_DESTINATION :
> ba17b1eb-961d-43fe-b820-cb469a8f292d 2018-04-25 23:47:39.434491 [DEBUG]
> switch_core_media.c:4028 sofia/internal_1/33183817938 at gv-ics-prd-301 T38
> REFUSE on request
> ba17b1eb-961d-43fe-b820-cb469a8f292d 2018-04-25 23:47:39.434491 [NOTICE]
> sofia.c:7566 Hangup sofia/internal_1/33183817938 at gv-ics-prd-301 [CS_NEW]
> [INCOMPATIBLE_DESTINATION]
>
> I didn't find any valid configuration to make those calls accepted, except
> configuring proxy_media, which I don't want to use.
>
> Is this behaviour a "normal" behaviour ? Could it be a bug ?
> This is a very big issue for us ...
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 shaun.stokes at itec-support.co.uk  Fri Apr 27 07:07:36 2018
From: shaun.stokes at itec-support.co.uk (Shaun Stokes)
Date: Fri, 27 Apr 2018 07:07:36 +0000
Subject: [Freeswitch-users] Implementing emergency calls ...
	or	call	priority ?
In-Reply-To: <E99DC39B-6D6C-4200-9035-1C99D49EAB3F@gmx.net>
References: <!&!AAAAAAAAAAAYAAAAAAAAAI/RfDx9+axBlz43ajVi2RXCgAAAEAAAAEWQvGn+wuxGngnm+Jt2R/YBAAAAAA==@gmail.com>,
 <E99DC39B-6D6C-4200-9035-1C99D49EAB3F@gmx.net>
Message-ID: <1524812856279.27045@itec-support.co.uk>

The best solution would probably be to build a LUA script.

Using LUA you can interact with FreeSWITCH via API 'api show channels as json' to list currently active channels into a JSON string, then convert that into an array. Process the array in a loop (i.e. for each or while) to identify which channels to drop based on your criteria.

________________________________________
From: FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> on behalf of Sebastian Kemper <sebastian_ml at gmx.net>
Sent: 27 April 2018 07:10
To: FreeSWITCH Users Help; Peter Nerád; 'FreeSWITCH Users Help'
Subject: Re: [Freeswitch-users] Implementing emergency calls ... or     call    priority ?

Am April 27, 2018 4:46:36 AM UTC schrieb "Peter Nerád" <nerad.peter at gmail.com>:
>Whats are best practicies to implement emergency calls ?
>
>I can identify  such a call by regexp but how to handle those calls ?
>
>Is there some build param to be set ?
>
>How I can set priority for call ? e.g. I have limited capacity for
>outgoing and incoming call to 10. I need to drop some calls with small
>priority to allow emergency call, but I have no idea where to start.
>
>Can somebody give me right direction for this ?
>
>Thank you very much
>
>PS: Google does not helped me :(
>
>
>_________________________________________________________________________
>Professional FreeSWITCH Consulting Services:
>consulting at freeswitch.org
>http://www.freeswitchsolutions.com
>
>Official FreeSWITCH Sites
>http://www.freeswitch.org
>http://confluence.freeswitch.org
>http://www.cluecon.com
>
>FreeSWITCH-users mailing list
>FreeSWITCH-users at lists.freeswitch.org
>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>http://www.freeswitch.org

Hi Peter,

You can define all sorts of limits. Have a look here: https://freeswitch.org/confluence/display/FREESWITCH/mod_dptools%3A+Limit

Regards,
Sebastian

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

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

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

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

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


From shaun.stokes at itec-support.co.uk  Fri Apr 27 07:25:28 2018
From: shaun.stokes at itec-support.co.uk (Shaun Stokes)
Date: Fri, 27 Apr 2018 07:25:28 +0000
Subject: [Freeswitch-users] Implementing emergency calls ...
	or	call	priority ?
In-Reply-To: <1524812856279.27045@itec-support.co.uk>
References: <!&!AAAAAAAAAAAYAAAAAAAAAI/RfDx9+axBlz43ajVi2RXCgAAAEAAAAEWQvGn+wuxGngnm+Jt2R/YBAAAAAA==@gmail.com>,
 <E99DC39B-6D6C-4200-9035-1C99D49EAB3F@gmx.net>,
 <1524812856279.27045@itec-support.co.uk>
Message-ID: <1524813928431.84646@itec-support.co.uk>

As Sebastian has pointed out you can also use limits, this will cause new calls to drop once the limit is reached but won't drop already established calls\channels. I believe the same call can be in multiple limits, if you had a limit for low priority calls you could use a LUA script to list these and drop one at random if there's a call which has higher priority such as emergency calls.

________________________________________
From: Shaun Stokes
Sent: 27 April 2018 08:07
To: FreeSWITCH Users Help; Peter Nerád; FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Implementing emergency calls ... or     call    priority ?

The best solution would probably be to build a LUA script.

Using LUA you can interact with FreeSWITCH via API 'api show channels as json' to list currently active channels into a JSON string, then convert that into an array. Process the array in a loop (i.e. for each or while) to identify which channels to drop based on your criteria.

________________________________________
From: FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> on behalf of Sebastian Kemper <sebastian_ml at gmx.net>
Sent: 27 April 2018 07:10
To: FreeSWITCH Users Help; Peter Nerád; 'FreeSWITCH Users Help'
Subject: Re: [Freeswitch-users] Implementing emergency calls ... or     call    priority ?

Am April 27, 2018 4:46:36 AM UTC schrieb "Peter Nerád" <nerad.peter at gmail.com>:
>Whats are best practicies to implement emergency calls ?
>
>I can identify  such a call by regexp but how to handle those calls ?
>
>Is there some build param to be set ?
>
>How I can set priority for call ? e.g. I have limited capacity for
>outgoing and incoming call to 10. I need to drop some calls with small
>priority to allow emergency call, but I have no idea where to start.
>
>Can somebody give me right direction for this ?
>
>Thank you very much
>
>PS: Google does not helped me :(
>
>
>_________________________________________________________________________
>Professional FreeSWITCH Consulting Services:
>consulting at freeswitch.org
>http://www.freeswitchsolutions.com
>
>Official FreeSWITCH Sites
>http://www.freeswitch.org
>http://confluence.freeswitch.org
>http://www.cluecon.com
>
>FreeSWITCH-users mailing list
>FreeSWITCH-users at lists.freeswitch.org
>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>http://www.freeswitch.org

Hi Peter,

You can define all sorts of limits. Have a look here: https://freeswitch.org/confluence/display/FREESWITCH/mod_dptools%3A+Limit

Regards,
Sebastian

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

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

FreeSWITCH-users mailing list
FreeSWITCH-users at 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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______________________________________________________________________

______________________________________________________________________
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From nerad.peter at gmail.com  Fri Apr 27 07:26:47 2018
From: nerad.peter at gmail.com (=?iso-8859-2?Q?Peter_Ner=E1d?=)
Date: Fri, 27 Apr 2018 09:26:47 +0200
Subject: [Freeswitch-users] Implementing emergency calls ...
	or	call	priority ?
In-Reply-To: <1524812856279.27045@itec-support.co.uk>
References: <!&!AAAAAAAAAAAYAAAAAAAAAI/RfDx9+axBlz43ajVi2RXCgAAAEAAAAEWQvGn+wuxGngnm+Jt2R/YBAAAAAA==@gmail.com>,
 <E99DC39B-6D6C-4200-9035-1C99D49EAB3F@gmx.net>
 <1524812856279.27045@itec-support.co.uk>
Message-ID: <!&!AAAAAAAAAAAYAAAAAAAAAI/RfDx9+axBlz43ajVi2RXCgAAAEAAAAIpGVUVPSy5Jk4UmlYU+TeUBAAAAAA==@gmail.com>

Thanks... This is a last option for me...
First i would like to search for ready solution ..

I thought that there is some standard solution to implement emergency calls
in freeswitch .. or prioritizing calls in case of  call limits is reached in
trunks for example..

-----Original Message-----
From: Shaun Stokes <shaun.stokes at itec-support.co.uk> 
Sent: Friday, April 27, 2018 9:08 AM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>; Peter
Nerád <nerad.peter at gmail.com>; FreeSWITCH Users Help
<freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Implementing emergency calls ... or call
priority ?

The best solution would probably be to build a LUA script.

Using LUA you can interact with FreeSWITCH via API 'api show channels as
json' to list currently active channels into a JSON string, then convert
that into an array. Process the array in a loop (i.e. for each or while) to
identify which channels to drop based on your criteria.

________________________________________
From: FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> on
behalf of Sebastian Kemper <sebastian_ml at gmx.net>
Sent: 27 April 2018 07:10
To: FreeSWITCH Users Help; Peter Nerád; 'FreeSWITCH Users Help'
Subject: Re: [Freeswitch-users] Implementing emergency calls ... or     call
priority ?

Am April 27, 2018 4:46:36 AM UTC schrieb "Peter Nerád"
<nerad.peter at gmail.com>:
>Whats are best practicies to implement emergency calls ?
>
>I can identify  such a call by regexp but how to handle those calls ?
>
>Is there some build param to be set ?
>
>How I can set priority for call ? e.g. I have limited capacity for 
>outgoing and incoming call to 10. I need to drop some calls with small 
>priority to allow emergency call, but I have no idea where to start.
>
>Can somebody give me right direction for this ?
>
>Thank you very much
>
>PS: Google does not helped me :(
>
>
>_______________________________________________________________________
>__ Professional FreeSWITCH Consulting Services:
>consulting at freeswitch.org
>http://www.freeswitchsolutions.com
>
>Official FreeSWITCH Sites
>http://www.freeswitch.org
>http://confluence.freeswitch.org
>http://www.cluecon.com
>
>FreeSWITCH-users mailing list
>FreeSWITCH-users at lists.freeswitch.org
>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-user
>s
>http://www.freeswitch.org

Hi Peter,

You can define all sorts of limits. Have a look here:
https://freeswitch.org/confluence/display/FREESWITCH/mod_dptools%3A+Limit

Regards,
Sebastian

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

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

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

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

______________________________________________________________________
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Scanning Service.
______________________________________________________________________



From dominique.jeannerod at interact-iv.com  Fri Apr 27 07:37:57 2018
From: dominique.jeannerod at interact-iv.com (Dominique Jeannerod)
Date: Fri, 27 Apr 2018 09:37:57 +0200
Subject: [Freeswitch-users] Freeswitch 1.6.19 and T.38 /
	INCOMPATIBLE_DESTINATION
Message-ID: <CAEzwe17TGK45wZH8j6sum04=pftXjrOw0bgMg8yZMCNqkK31Zw@mail.gmail.com>

So, I tested with "  <X-PRE-PROCESS cmd="set" data="fax_enable_t38=true"/>"
in vars.xml, which i didn't do previously.
The behaviour is quite different. However, the call is directly rejected
after the first INVITE (which is not a REINVITE), but instead of having " T38
REFUSE on request", i have :
2018-04-27 09:28:48.697285 [DEBUG] switch_core_media.c:4041
sofia/internal_2/698778763 at gv-ics-prd-301 T38 ACCEPT on request
2018-04-27 09:28:48.697285 [DEBUG] switch_core_media.c:4146
sofia/internal_2/698778763 at gv-ics-prd-301 T38 IS NOT POSSIBLE on request
2018-04-27 09:28:48.697285 [NOTICE] sofia.c:7566 Hangup
sofia/internal_2/698778763 at gv-ics-prd-301 [CS_NEW]
[INCOMPATIBLE_DESTINATION]

Is there another parameter to configure to allow T38 ? Add a codec (I only
have PCMA, and PCMU)

Full log :

   INVITE sip:0170993650 at 10.9.30.34:5060 SIP/2.0
   Record-Route: <sip:10.3.1.220;lr=on>
   Via: SIP/2.0/UDP 10.3.1.220;branch=z9hG4bK002b.b2b0e9d2.0
   Via: SIP/2.0/UDP 10.3.1.231;branch=z9hG4bKac408981189
   Max-Forwards: 69
   From: <sip:698778763 at gv-ics-prd-301>;tag=1c408970979
   To: <sip:0170993650 at 10.3.1.220;user=phone>
   Call-ID: 408970087274201872848 at 10.3.1.231
   CSeq: 1 INVITE
   Contact: <sip:698778763 at 10.3.1.231:5060>
   Supported: em,100rel,timer,replaces,path,resource-priority,sdp-anat
   Allow:
REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
   Remote-Party-ID: <sip:698778763 at 10.3.1.220
>;party=calling;privacy=off;screen=yes;screen-ind=3;npi=1;ton=2
   Remote-Party-ID: <sip:0170993650 at 10.3.1.220>;party=called;npi=1;ton=2
   User-Agent: Audiocodes-Sip-Gateway-Mediant 3000/v.6.00A.056
   Content-Type: application/sdp
   Content-Length: 413

   v=0
   o=AudiocodesGW 408954776 408954489 IN IP4 10.3.1.231
   s=Phone-Call
   c=IN IP4 10.3.1.231
   t=0 0
   m=audio 7850 RTP/AVP 8 101
   a=rtpmap:8 PCMA/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-15
   a=ptime:20
   a=sendrecv
   m=image 7852 udptl t38
   a=T38FaxVersion:0
   a=T38MaxBitRate:33600
   a=T38FaxMaxBuffer:3000
   a=T38FaxMaxDatagram:560
   a=T38FaxRateManagement:transferredTCF
   a=T38FaxUdpEC:t38UDPRedundancy
   ------------------------------------------------------------------------
send 374 bytes to udp/[10.3.1.220]:5060 at 09:28:49.974884:
   ------------------------------------------------------------------------
   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP 10.3.1.220;branch=z9hG4bK002b.b2b0e9d2.0
   Via: SIP/2.0/UDP 10.3.1.231;branch=z9hG4bKac408981189
   Record-Route: <sip:10.3.1.220;lr=on>
   From: <sip:698778763 at gv-ics-prd-301>;tag=1c408970979
   To: <sip:0170993650 at 10.3.1.220;user=phone>
   Call-ID: 408970087274201872848 at 10.3.1.231
   CSeq: 1 INVITE
   User-Agent: FreeSWITCH
   Content-Length: 0

   ------------------------------------------------------------------------
2018-04-27 09:28:49.957287 [NOTICE] switch_channel.c:1104 New Channel
sofia/internal_2/698778763 at gv-ics-prd-301
[233f8391-10e6-4877-a792-6ad2a0f651e0]
2018-04-27 09:28:49.957287 [DEBUG] switch_core_state_machine.c:584
(sofia/internal_2/698778763 at gv-ics-prd-301) Running State Change CS_NEW
(Cur 1 Tot 2)
2018-04-27 09:28:49.957287 [DEBUG] sofia.c:9874
sofia/internal_2/698778763 at gv-ics-prd-301 receiving invite from
10.3.1.220:5060 version: 1.6.19  64bit
2018-04-27 09:28:49.957287 [DEBUG] sofia.c:7085 Channel
sofia/internal_2/698778763 at gv-ics-prd-301 entering state [received][100]
2018-04-27 09:28:49.957287 [DEBUG] sofia.c:7095 Remote SDP:
v=0
o=AudiocodesGW 408954776 408954489 IN IP4 10.3.1.231
s=Phone-Call
c=IN IP4 10.3.1.231
t=0 0
m=audio 7850 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
m=image 7852 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:33600
a=T38FaxMaxBuffer:3000
a=T38FaxMaxDatagram:560
a=T38FaxRateManagement:transferredTCF
a=T38FaxUdpEC:t38UDPRedundancy

2018-04-27 09:28:49.957287 [DEBUG] switch_core_media.c:4449 Audio Codec
Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
2018-04-27 09:28:49.957287 [DEBUG] switch_core_media.c:4504 Audio Codec
Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match
2018-04-27 09:28:49.957287 [DEBUG] switch_core_media.c:4449 Audio Codec
Compare [PCMA:8:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]
2018-04-27 09:28:49.957287 [DEBUG] switch_core_media.c:4365 Set
telephone-event payload to 101 at 8000
2018-04-27 09:28:49.957287 [DEBUG] switch_core_media.c:3061 Set Codec
sofia/internal_2/698778763 at gv-ics-prd-301 PCMA/8000 20 ms 160 samples 64000
bits 1 channels
2018-04-27 09:28:49.957287 [DEBUG] switch_core_codec.c:111
sofia/internal_2/698778763 at gv-ics-prd-301 Original read codec set to PCMA:8
2018-04-27 09:28:49.957287 [DEBUG] switch_core_media.c:4708 Set
telephone-event payload to 101 at 8000
2018-04-27 09:28:49.957287 [DEBUG] switch_core_media.c:4767
sofia/internal_2/698778763 at gv-ics-prd-301 Set 2833 dtmf send payload to 101
recv payload to 101
2018-04-27 09:28:49.977279 [DEBUG] switch_core_media.c:4041
sofia/internal_2/698778763 at gv-ics-prd-301 T38 ACCEPT on request
2018-04-27 09:28:49.977279 [DEBUG] switch_core_media.c:4146
sofia/internal_2/698778763 at gv-ics-prd-301 T38 IS NOT POSSIBLE on request
2018-04-27 09:28:49.977279 [NOTICE] sofia.c:7566 Hangup
sofia/internal_2/698778763 at gv-ics-prd-301 [CS_NEW]
[INCOMPATIBLE_DESTINATION]
2018-04-27 09:28:49.977279 [DEBUG] switch_core_state_machine.c:603
(sofia/internal_2/698778763 at gv-ics-prd-301) State NEW
2018-04-27 09:28:49.977279 [DEBUG] switch_core_state_machine.c:584
(sofia/internal_2/698778763 at gv-ics-prd-301) Running State Change CS_HANGUP
(Cur 1 Tot 2)
2018-04-27 09:28:49.977279 [DEBUG] switch_core_state_machine.c:850
(sofia/internal_2/698778763 at gv-ics-prd-301) Callstate Change DOWN -> HANGUP
2018-04-27 09:28:49.977279 [DEBUG] switch_core_state_machine.c:852
(sofia/internal_2/698778763 at gv-ics-prd-301) State HANGUP
2018-04-27 09:28:49.977279 [DEBUG] mod_sofia.c:438 Channel
sofia/internal_2/698778763 at gv-ics-prd-301 hanging up, cause:
INCOMPATIBLE_DESTINATION
2018-04-27 09:28:49.977279 [DEBUG] mod_sofia.c:577 Responding to INVITE
with: 488
send 720 bytes to udp/[10.3.1.220]:5060 at 09:28:49.978310:
   ------------------------------------------------------------------------
   SIP/2.0 488 Not Acceptable Here
   Via: SIP/2.0/UDP 10.3.1.220;branch=z9hG4bK002b.b2b0e9d2.0
   Via: SIP/2.0/UDP 10.3.1.231;branch=z9hG4bKac408981189
   Max-Forwards: 69
   From: <sip:698778763 at gv-ics-prd-301>;tag=1c408970979
   To: <sip:0170993650 at 10.3.1.220;user=phone>;tag=vy16j7B2HN1Xm
   Call-ID: 408970087274201872848 at 10.3.1.231
   CSeq: 1 INVITE
   User-Agent: FreeSWITCH
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER,
NOTIFY
   Supported: timer, path, replaces
   Allow-Events: talk, hold, conference, refer
   Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION"
   Content-Length: 0
   Remote-Party-ID: "0170993650" <sip:0170993650 at 10.3.1.220
>;party=calling;privacy=off;screen=no

2018-04-27 8:24 GMT+02:00 Dominique Jeannerod <
dominique.jeannerod at interact-iv.com>:

> Thanks a lot for this answer.
> I already read this information mail, and read the changes done for t38
> re-invite.
> The problem i have is NOT on the reinvite but directly on the first invite
> of the Call
> Do you think the behaviour also changed for the first invite ?
>
> Anyway, i will try the new configuration parameter, and test if it fixes
> my problem.
> My short term fallback plan would just be to downgrade to the 1.6.8
> version we already have on production.
>
> Le jeu. 26 avr. 2018 à 23:52, Michael Jerris <mike at jerris.com> a écrit :
>
>> http://lists.freeswitch.org/pipermail/freeswitch-users/
>> 2017-January/124459.html
>>
>> On Apr 26, 2018, at 5:20 PM, Dominique Jeannerod <
>> dominique.jeannerod at interact-iv.com> wrote:
>>
>> Hello,
>>
>> I currently use Freeswitch 1.6.8-1 in production, and have some ISDN
>> gateways sending SDP including T.38 for voice calls (not faxes). Calls are
>> handled without problems by freeswitch.
>>
>> I have another platform, with freeswitch 1.6.19 and same configuration.
>> However, voice calls received with SDP including t.38 are rejected with
>> 488 INCOMPATIBLE_DESTINATION :
>> ba17b1eb-961d-43fe-b820-cb469a8f292d 2018-04-25 23:47:39.434491 [DEBUG]
>> switch_core_media.c:4028 sofia/internal_1/33183817938 at gv-ics-prd-301 T38
>> REFUSE on request
>> ba17b1eb-961d-43fe-b820-cb469a8f292d 2018-04-25 23:47:39.434491 [NOTICE]
>> sofia.c:7566 Hangup sofia/internal_1/33183817938 at gv-ics-prd-301 [CS_NEW]
>> [INCOMPATIBLE_DESTINATION]
>>
>> I didn't find any valid configuration to make those calls accepted,
>> except configuring proxy_media, which I don't want to use.
>>
>> Is this behaviour a "normal" behaviour ? Could it be a bug ?
>> This is a very big issue for us ...
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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 s.safarov at gmail.com  Fri Apr 27 07:40:12 2018
From: s.safarov at gmail.com (Sergey Safarov)
Date: Fri, 27 Apr 2018 07:40:12 +0000
Subject: [Freeswitch-users] Freeswitch 1.6.19 and T.38 /
	INCOMPATIBLE_DESTINATION
In-Reply-To: <CAEzwe17g2zFjuS27aMS+yfb8f3Y-Qy_+Ht4t6nLNk4a77GBx6Q@mail.gmail.com>
References: <CAEzwe14y6PX3Vzewjt27fMQT=rL1SGotzUzoEr-38C8=vDA36Q@mail.gmail.com>
 <1714B91B-C1F3-4D4E-B7BD-A0035E1599C6@jerris.com>
 <CAEzwe17g2zFjuS27aMS+yfb8f3Y-Qy_+Ht4t6nLNk4a77GBx6Q@mail.gmail.com>
Message-ID: <CAHtxdDdemBNrLZLKQOe31_miPCmenG-9SX1tgFzRfK2vvJ2ckA@mail.gmail.com>

Think you not applied recommendation

If you wish to retain the previous behavior you can set fax_enable_t38=true
in your vars.xml, You can also not set that and just set refuse_t38=true if
you wish to never do t.38.


пт, 27 апр. 2018 г. в 10:03, Dominique Jeannerod <
dominique.jeannerod at interact-iv.com>:

> Thanks a lot for this answer.
> I already read this information mail, and read the changes done for t38
> re-invite.
> The problem i have is NOT on the reinvite but directly on the first invite
> of the Call
> Do you think the behaviour also changed for the first invite ?
>
> Anyway, i will try the new configuration parameter, and test if it fixes
> my problem.
> My short term fallback plan would just be to downgrade to the 1.6.8
> version we already have on production.
>
>
> Le jeu. 26 avr. 2018 à 23:52, Michael Jerris <mike at jerris.com> a écrit :
>
>>
>> http://lists.freeswitch.org/pipermail/freeswitch-users/2017-January/124459.html
>>
>> On Apr 26, 2018, at 5:20 PM, Dominique Jeannerod <
>> dominique.jeannerod at interact-iv.com> wrote:
>>
>> Hello,
>>
>> I currently use Freeswitch 1.6.8-1 in production, and have some ISDN
>> gateways sending SDP including T.38 for voice calls (not faxes). Calls are
>> handled without problems by freeswitch.
>>
>> I have another platform, with freeswitch 1.6.19 and same configuration.
>> However, voice calls received with SDP including t.38 are rejected with
>> 488 INCOMPATIBLE_DESTINATION :
>> ba17b1eb-961d-43fe-b820-cb469a8f292d 2018-04-25 23:47:39.434491 [DEBUG]
>> switch_core_media.c:4028 sofia/internal_1/33183817938 at gv-ics-prd-301 T38
>> REFUSE on request
>> ba17b1eb-961d-43fe-b820-cb469a8f292d 2018-04-25 23:47:39.434491 [NOTICE]
>> sofia.c:7566 Hangup sofia/internal_1/33183817938 at gv-ics-prd-301 [CS_NEW]
>> [INCOMPATIBLE_DESTINATION]
>>
>> I didn't find any valid configuration to make those calls accepted,
>> except configuring proxy_media, which I don't want to use.
>>
>> Is this behaviour a "normal" behaviour ? Could it be a bug ?
>> This is a very big issue for us ...
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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 dominique.jeannerod at interact-iv.com  Fri Apr 27 07:42:04 2018
From: dominique.jeannerod at interact-iv.com (Dominique Jeannerod)
Date: Fri, 27 Apr 2018 09:42:04 +0200
Subject: [Freeswitch-users] Freeswitch 1.6.19 and T.38 /
	INCOMPATIBLE_DESTINATION
In-Reply-To: <CAEzwe17TGK45wZH8j6sum04=pftXjrOw0bgMg8yZMCNqkK31Zw@mail.gmail.com>
References: <CAEzwe17TGK45wZH8j6sum04=pftXjrOw0bgMg8yZMCNqkK31Zw@mail.gmail.com>
Message-ID: <CAEzwe15oPqvnMLhQ_U6hjsNGfgUP8Crp9HmcHWZhd-azySqruQ@mail.gmail.com>

Looks exactly like issue FS-10422
<https://freeswitch.org/jira/browse/FS-10422>​ and I can reproduce it easily
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From dominique.jeannerod at interact-iv.com  Fri Apr 27 08:18:06 2018
From: dominique.jeannerod at interact-iv.com (Dominique Jeannerod)
Date: Fri, 27 Apr 2018 10:18:06 +0200
Subject: [Freeswitch-users] Freeswitch 1.6.19 and T.38 /
	INCOMPATIBLE_DESTINATION
Message-ID: <CAEzwe179x63iP+9PKaNDz7xpqhzLQDU3aLO2WYrnuZhP-TWf0A@mail.gmail.com>

Yes, you're right, and i apologize for that.
I tested in the meantime (sent another mail in parallel), and it still
doesn't work.
I have the same error as in JIRA 10422 :
https://freeswitch.org/jira/browse/FS-10422

I've got " T38 IS NOT POSSIBLE on request " on the FIRST invite (not a
RE-INVITE)


2018-04-27 9:40 GMT+02:00 Sergey Safarov <s.safarov at gmail.com>:

> Think you not applied recommendation
>
> If you wish to retain the previous behavior you can set fax_enable_t38=true
> in your vars.xml, You can also not set that and just set refuse_t38=true if
> you wish to never do t.38.
>
>
> пт, 27 апр. 2018 г. в 10:03, Dominique Jeannerod <
> dominique.jeannerod at interact-iv.com>:
>
>> Thanks a lot for this answer.
>> I already read this information mail, and read the changes done for t38
>> re-invite.
>> The problem i have is NOT on the reinvite but directly on the first
>> invite of the Call
>> Do you think the behaviour also changed for the first invite ?
>>
>> Anyway, i will try the new configuration parameter, and test if it fixes
>> my problem.
>> My short term fallback plan would just be to downgrade to the 1.6.8
>> version we already have on production.
>>
>>
>> Le jeu. 26 avr. 2018 à 23:52, Michael Jerris <mike at jerris.com> a écrit :
>>
>>> http://lists.freeswitch.org/pipermail/freeswitch-users/
>>> 2017-January/124459.html
>>>
>>> On Apr 26, 2018, at 5:20 PM, Dominique Jeannerod <
>>> dominique.jeannerod at interact-iv.com> wrote:
>>>
>>> Hello,
>>>
>>> I currently use Freeswitch 1.6.8-1 in production, and have some ISDN
>>> gateways sending SDP including T.38 for voice calls (not faxes). Calls are
>>> handled without problems by freeswitch.
>>>
>>> I have another platform, with freeswitch 1.6.19 and same configuration.
>>> However, voice calls received with SDP including t.38 are rejected with
>>> 488 INCOMPATIBLE_DESTINATION :
>>> ba17b1eb-961d-43fe-b820-cb469a8f292d 2018-04-25 23:47:39.434491 [DEBUG]
>>> switch_core_media.c:4028 sofia/internal_1/33183817938 at gv-ics-prd-301
>>> T38 REFUSE on request
>>> ba17b1eb-961d-43fe-b820-cb469a8f292d 2018-04-25 23:47:39.434491
>>> [NOTICE] sofia.c:7566 Hangup sofia/internal_1/33183817938 at gv-ics-prd-301
>>> [CS_NEW] [INCOMPATIBLE_DESTINATION]
>>>
>>> I didn't find any valid configuration to make those calls accepted,
>>> except configuring proxy_media, which I don't want to use.
>>>
>>> Is this behaviour a "normal" behaviour ? Could it be a bug ?
>>> This is a very big issue for us ...
>>>
>>>
>>> ____________________________________________________________
>>> _____________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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 nerad.peter at gmail.com  Fri Apr 27 08:31:11 2018
From: nerad.peter at gmail.com (=?iso-8859-2?Q?Peter_Ner=E1d?=)
Date: Fri, 27 Apr 2018 10:31:11 +0200
Subject: [Freeswitch-users] Some changes in events messages from
	mod_callcenter ?
Message-ID: <!&!AAAAAAAAAAAYAAAAAAAAAI/RfDx9+axBlz43ajVi2RXCgAAAEAAAAOzfWp1S42lLuAHpAeO9rgIBAAAAAA==@gmail.com>

Hi,

 

I m parsing some events from callcenter in my app.

 

I m upgrade freeswitch to 1.6.20 version... i have problems to parse this
events now because some values in events messages are now in uuid format,
not text.

 

In this new version:

 

CC-Queue: 13b5a0a3-c8ec-428d-a8f6-36e32cbd3912

CC-Agent: 5b5e51b2-94b0-4b64-85d8-28fde9505894 which is not same as
CC-Agent-UUID 

 

In the past version:

CC-Queue: support at default

CC-Agent: AgentNameHere

 

Is there any option in config i can change the CC-QUEUE and CC-AGENT to the
past format or it is a bug ?

 

Im using a fusionpbx to manage mod_callcenter... som maybe there is some
error...

 

 

 

 

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From ssinyagin at gmail.com  Fri Apr 27 15:23:51 2018
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Fri, 27 Apr 2018 15:23:51 +0000
Subject: [Freeswitch-users] SIMCOM LTE modems with USB Audio
In-Reply-To: <CALXCt0pKBFBHV4oH8BxkEkpD30Jpoc7xkRckMO6fWLismbUYTQ@mail.gmail.com>
References: <CACOYK=hNTVPScbo_OGo-aO=ryuS-YbAaDb7o_nhbUi4YNKGdFw@mail.gmail.com>
 <CAPWmf4tEJvaoqpsEVbueRH_QcOb9+nV-bCmm0H+CV7dhCYyE+Q@mail.gmail.com>
 <CACOYK=gWzsD__m2bowvN0Cmf_P+43h3P5VkNnTTUn3y9AJn=Yg@mail.gmail.com>
 <CACOYK=gj_5Lz-d_ih6EsjmC3uATz-Z4dDnzPX0kyVyXKqihLQw@mail.gmail.com>
 <CALXCt0pKBFBHV4oH8BxkEkpD30Jpoc7xkRckMO6fWLismbUYTQ@mail.gmail.com>
Message-ID: <CACOYK=iZq_thkVnN10jzeeya_VWCXTVmwJMnoZTDZx2futOJ2g@mail.gmail.com>

I'd be glad to, but I don't have a budget on my own. Need to find the
funding first.




On Thu, Apr 26, 2018, 21:15 Giovanni Maruzzelli <gmaruzz at gmail.com> wrote:

> please do it, then let us know. Maybe do it for free, and donate it to the
> community.
>
> Thanks in advance
>
>
>
> On 25 April 2018 at 23:30, Stanislav Sinyagin <ssinyagin at gmail.com> wrote:
>
>> probably the most flexible option would be a small standalone daemon
>> that links the SIMCOM modem audio with a virtual PortAudio device.
>> Then you can have this stream in FreeSWITCH or in any other software
>> that is able to communicate with PA.
>>
>>
>>
>> On Wed, Apr 25, 2018 at 1:49 PM, Stanislav Sinyagin <ssinyagin at gmail.com>
>> wrote:
>> > Yes, it's clearly written in their documentation, that raw PCM is sent
>> in an
>> > USB UART device.
>> >
>> > Mod_gsmopen is too imperfect, to say politely. It's easier to create a
>> new
>> > module for UART audio than trying to adapt gsmopen for new modems.
>> >
>> > So, it only needs a budget for development.
>> >
>> >
>> >
>> >
>> >
>> > On Tue, Apr 24, 2018, 22:01 Tom Hartnett <hartnett.tom at gmail.com>
>> wrote:
>> >>
>> >> Funny you mention the Simcom as I'm testing the 7600AH now (unrelated
>> to
>> >> Freeswitch). The modems are odd in that they don't create a USB audio
>> >> endpoint in the traditional sense. You have to move the PCM voice over
>> their
>> >> UART interface. That requires a little extra work as I haven't really
>> found
>> >> a good driver to manage that.
>> >>
>> >> I'm really interested in these because I want to support VoLTE on
>> AT&T. We
>> >> haven't got that running yet, and my AT&T contact says there was an
>> issue
>> >> with provisioning this model in their database. Hopefully that will be
>> fixed
>> >> later this week.
>> >>
>> >> On Tue, Apr 24, 2018 at 3:28 PM, Stanislav Sinyagin <
>> ssinyagin at gmail.com>
>> >> wrote:
>> >>>
>> >>> hi,
>> >>>
>> >>> I wrote about a year ago about SIMCOM SIM7100 series LTE modems and
>> >>> that they support USB audio. Now the vendor offers also SIM7500 and
>> >>> SIM7600 modems, also with USB audio, and in addition these newer
>> >>> models support 16-bit audio streams.
>> >>>
>> >>> Unfortunately I didn't find the free time to make a FreeSWITCH module
>> >>> for them, but if someone wants to sponsor such a development, or
>> >>> actually develop it, I'll be glad to work together.
>> >>>
>> >>> I've got a SIM7100E modem, and SIM7500E is ordered and is on the way.
>> >>>
>> >>> The SIM7500 series is cheaper, but it only supports 10Mbps downstream
>> >>> for mobile data. The SIM7600 supports up to 150Mbps. But for voice
>> >>> applications, data transfer rate should be irrelevant in any way.
>> >>>
>> >>>
>> >>> cheers,
>> >>> stanislav
>> >>>
>> >>>
>> _________________________________________________________________________
>> >>> Professional FreeSWITCH Consulting Services:
>> >>> consulting at freeswitch.org
>> >>> http://www.freeswitchsolutions.com
>> >>>
>> >>> Official FreeSWITCH Sites
>> >>> http://www.freeswitch.org
>> >>> http://confluence.freeswitch.org
>> >>> http://www.cluecon.com
>> >>>
>> >>> FreeSWITCH-users mailing list
>> >>> FreeSWITCH-users at lists.freeswitch.org
>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> http://www.freeswitch.org
>> >>
>> >>
>> >>
>> _________________________________________________________________________
>> >> Professional FreeSWITCH Consulting Services:
>> >> consulting at freeswitch.org
>> >> http://www.freeswitchsolutions.com
>> >>
>> >> Official FreeSWITCH Sites
>> >> http://www.freeswitch.org
>> >> http://confluence.freeswitch.org
>> >> http://www.cluecon.com
>> >>
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
>
> Sincerely,
>
> Giovanni Maruzzelli
> OpenTelecom.IT
> cell: +39 347 266 56 18
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From mike at jerris.com  Fri Apr 27 17:03:19 2018
From: mike at jerris.com (Michael Jerris)
Date: Fri, 27 Apr 2018 13:03:19 -0400
Subject: [Freeswitch-users] Freeswitch 1.6.19 and T.38 /
 INCOMPATIBLE_DESTINATION
In-Reply-To: <CAEzwe179x63iP+9PKaNDz7xpqhzLQDU3aLO2WYrnuZhP-TWf0A@mail.gmail.com>
References: <CAEzwe179x63iP+9PKaNDz7xpqhzLQDU3aLO2WYrnuZhP-TWf0A@mail.gmail.com>
Message-ID: <5DA8D848-3451-471D-9F8C-92DEAC8ED82F@jerris.com>

As noted in that bug it was closed can’t reproduce.  Please follow the requests in that bug to get this looked at.

> On Apr 27, 2018, at 4:18 AM, Dominique Jeannerod <dominique.jeannerod at interact-iv.com> wrote:
> 
> Yes, you're right, and i apologize for that.
> I tested in the meantime (sent another mail in parallel), and it still doesn't work.
> I have the same error as in JIRA 10422 : https://freeswitch.org/jira/browse/FS-10422 <https://freeswitch.org/jira/browse/FS-10422>
> 
> I've got " T38 IS NOT POSSIBLE on request " on the FIRST invite (not a RE-INVITE)
> 
> 
> 2018-04-27 9:40 GMT+02:00 Sergey Safarov <s.safarov at gmail.com <mailto:s.safarov at gmail.com>>:
> Think you not applied recommendation
> 
> If you wish to retain the previous behavior you can set fax_enable_t38=true
> in your vars.xml, You can also not set that and just set refuse_t38=true if
> you wish to never do t.38.
> 
> пт, 27 апр. 2018 г. в 10:03, Dominique Jeannerod <dominique.jeannerod at interact-iv.com <mailto:dominique.jeannerod at interact-iv.com>>:
> Thanks a lot for this answer.
> I already read this information mail, and read the changes done for t38 re-invite.
> The problem i have is NOT on the reinvite but directly on the first invite of the Call
> Do you think the behaviour also changed for the first invite ?
> 
> Anyway, i will try the new configuration parameter, and test if it fixes my problem.
> My short term fallback plan would just be to downgrade to the 1.6.8 version we already have on production.
> 
> 
> Le jeu. 26 avr. 2018 à 23:52, Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>> a écrit :
> http://lists.freeswitch.org/pipermail/freeswitch-users/2017-January/124459.html <http://lists.freeswitch.org/pipermail/freeswitch-users/2017-January/124459.html>
> 
>> On Apr 26, 2018, at 5:20 PM, Dominique Jeannerod <dominique.jeannerod at interact-iv.com <mailto:dominique.jeannerod at interact-iv.com>> wrote:
>> 
>> Hello,
>> 
>> I currently use Freeswitch 1.6.8-1 in production, and have some ISDN gateways sending SDP including T.38 for voice calls (not faxes). Calls are handled without problems by freeswitch.
>> 
>> I have another platform, with freeswitch 1.6.19 and same configuration.
>> However, voice calls received with SDP including t.38 are rejected with 488 INCOMPATIBLE_DESTINATION :
>> ba17b1eb-961d-43fe-b820-cb469a8f292d 2018-04-25 23:47:39.434491 [DEBUG] switch_core_media.c:4028 sofia/internal_1/33183817938 at gv-ics-prd-301 T38 REFUSE on request
>> ba17b1eb-961d-43fe-b820-cb469a8f292d 2018-04-25 23:47:39.434491 [NOTICE] sofia.c:7566 Hangup sofia/internal_1/33183817938 at gv-ics-prd-301 [CS_NEW] [INCOMPATIBLE_DESTINATION]
>> 
>> I didn't find any valid configuration to make those calls accepted, except configuring proxy_media, which I don't want to use.
>> 
>> Is this behaviour a "normal" behaviour ? Could it be a bug ?
>> This is a very big issue for us ...
>> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> http://www.cluecon.com <http://www.cluecon.com/>
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> http://www.freeswitch.org <http://www.freeswitch.org/>_________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> http://www.cluecon.com <http://www.cluecon.com/>
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> http://www.freeswitch.org <http://www.freeswitch.org/>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> http://www.cluecon.com <http://www.cluecon.com/>
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> http://www.freeswitch.org <http://www.freeswitch.org/>
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> http://www.cluecon.com <http://www.cluecon.com/>
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> http://www.freeswitch.org <http://www.freeswitch.org/>
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From bilaln018 at gmail.com  Fri Apr 27 19:20:27 2018
From: bilaln018 at gmail.com (Bilal Abbasi)
Date: Fri, 27 Apr 2018 19:20:27 +0000
Subject: [Freeswitch-users] [event socket][client connection halt]
Message-ID: <CADSVerBK8BOQkis+12qA1q5S_wyY-te4L1yiiXK+F6X=own05A@mail.gmail.com>

Hi Users,
I came across an issue that my golang connection to event socket halts
after few days(normally 1-2 days).
When that halts i am not able to make calls or receive events.
I am using below library for event socket
https://github.com/fiorix/go-eventsocket

Cloud service provider: amazon ec2
I am just thinking its tcp connection break that is not started again by
the library( may be some method to keep alive the port).
Can somebody help me with this issue.

Regards
Abbasi
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From prestonh at gmail.com  Fri Apr 27 20:12:48 2018
From: prestonh at gmail.com (Preston Hagar)
Date: Fri, 27 Apr 2018 15:12:48 -0500
Subject: [Freeswitch-users] Sangoma A102, PRIs, Caller ID Name, and Facility
Message-ID: <CAK6zN=2US_JUHB7JvGGU7uO_HSK12irtdJSF8SjAMqpCHeTQeQ@mail.gmail.com>

Hi all,

We are working on setting up a FreeSWITCH system using a PRI connected to a
Sangoma A102DE card.

Everything is working, except Caller ID Name (we get caller ID number).

The Caller ID is getting passed in the SETUP message, but the Sangoma
libraries don't seem to pull it out.  If we try setting facility=yes in the
freetdm.conf.xml file, FreeSwitch segfaults (in the ftmod_sangoma_isdn.so
file), so at least right now, that isn't a fix.

We have a ticket open with Sangoma, but it is taking a while.

I am just curious, does anyone out there have a FreeSwitch system working
with a Sangoma card hooked to a PRI and receiving caller ID names?

If so, are you willing to share any model/config information?

This is our last step to get this off the ground, so any help anyone can
provide would be greatly appreciated.

Thanks!

Preston
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From prestonh at gmail.com  Fri Apr 27 20:18:06 2018
From: prestonh at gmail.com (Preston Hagar)
Date: Fri, 27 Apr 2018 15:18:06 -0500
Subject: [Freeswitch-users] valet_park and caller Id number
In-Reply-To: <CAK6zN=0X06q=Xca9CgR4VxMowsxbRHe_M2uX998wJAzDNtejfA@mail.gmail.com>
References: <CAK6zN=0X06q=Xca9CgR4VxMowsxbRHe_M2uX998wJAzDNtejfA@mail.gmail.com>
Message-ID: <CAK6zN=0Xyspcu6O56wqqKEeZXZriAL929GHwT6mXGQYY=U-qZA@mail.gmail.com>

Just thought I would send a followup in case someone ever runs into the
same problem and comes across this thread later.

We ended up having to modify mod_valet_parking.c to pull in the
caller_id_number channel variable and then added it as an
event_header_string.

Then in sofia_presence.c we edited the section that builds the presence XML
for parked calls and grabbed that caller ID number and put it in the
<remote><identity display="" section instead of just having the word "park"
hardcoded like it is now.

We now get caller Id numbers on things like Voice Operator Panel by having
it monitor the presence on the parked extensions.

If anyone ever needs the code, feel free to contact me and I'm happy to
share.

On Thu, Apr 19, 2018 at 1:23 PM, Preston Hagar <prestonh at gmail.com> wrote:

> I've setup mod_valet_park to have a set of "parking slots" so that our
> receptionist can park the calls in, page for someone to pickup, and then
> that person can dial in and pick up.  That is all working well.
>
> I'm using Voice Operator Panel (VOP) for the receptionist software and
> want them to be able to monitor presence on the parking slots so that they
> can see which ones have calls parked that haven't been picked up.  I'd also
> like them to be able to see the caller id of the call parked in each slot.
>
> VOP is monitoring presence fine, but for callerId it just shows "park" as
> the number.  I traced the presence packets in the console and the problem
> is that in the remote->identity display section of the notify packet, it
> has "park" as the value to display, instead of the caller id number parked
> there.
>
> (see example XML below).
>
> I have a couple of questions:
>
> 1.  Is it even possible to get mod_valet_park/freeswitch to report the
> parked caller id number in presence requests in the current
> mod_valet_park?  If so, how is it done?
>
> 2.  If it isn't in the current code, can anyone point me to the code block
> that generates that remote XML section of the presence notification?  I
> poked around in the valet_send_presence function of mod_valet_parking.c,
> but can't seem to figure out how that block is built.  I will admit that
> I'm still getting up to speed on how Presence works in FreeSWITCH.
>
> Thanks for your help!
>
> Here is a sample notify block:
>
> <?xml version="1.0"?>
>       <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="0"
> state="full" entity="sip:park+8001 at pbx.myfreeswitchdomain.com">
>         <dialog id="8001" direction="recipient">
>          <state>confirmed</state>
>           <local>
>              <identity display="park">sip:8001 at pbx.myfreeswitchdomain.com
> ;proto=park</identity>
>               <target uri="sip:8001 at pbx.myfreeswitchdomain.com;proto=
> park">
>                 <param pname="+sip.rendering" pvalue="no"/>
>                </target>
>             </local>
>             <remote>
>               <identity display="park">sip:8001</identity>
>               <target uri="sip:park+8001 at pbx.myfreeswitchdomain.com"/>
>             </remote>
>            </dialog>
>         </dialog-info>
>
>
>
>
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From joel at gogii.net  Fri Apr 27 22:06:46 2018
From: joel at gogii.net (Joel Serrano)
Date: Fri, 27 Apr 2018 22:06:46 +0000
Subject: [Freeswitch-users] valet_park and caller Id number
In-Reply-To: <CAK6zN=0Xyspcu6O56wqqKEeZXZriAL929GHwT6mXGQYY=U-qZA@mail.gmail.com>
References: <CAK6zN=0X06q=Xca9CgR4VxMowsxbRHe_M2uX998wJAzDNtejfA@mail.gmail.com>
 <CAK6zN=0Xyspcu6O56wqqKEeZXZriAL929GHwT6mXGQYY=U-qZA@mail.gmail.com>
Message-ID: <CAMtXxQnCa_maJy-3hoek950BfoSzyMLnq00fa_Mt2X+XBGG-Uw@mail.gmail.com>

Maybe you can make that functionality configurable and submit a PR to
upstream?

I’m sure someone can/will benefit from your effort.

Joel.

On Fri, Apr 27, 2018 at 13:47 Preston Hagar <prestonh at gmail.com> wrote:

> Just thought I would send a followup in case someone ever runs into the
> same problem and comes across this thread later.
>
> We ended up having to modify mod_valet_parking.c to pull in the
> caller_id_number channel variable and then added it as an
> event_header_string.
>
> Then in sofia_presence.c we edited the section that builds the presence
> XML for parked calls and grabbed that caller ID number and put it in the
> <remote><identity display="" section instead of just having the word "park"
> hardcoded like it is now.
>
> We now get caller Id numbers on things like Voice Operator Panel by having
> it monitor the presence on the parked extensions.
>
> If anyone ever needs the code, feel free to contact me and I'm happy to
> share.
>
> On Thu, Apr 19, 2018 at 1:23 PM, Preston Hagar <prestonh at gmail.com> wrote:
>
>> I've setup mod_valet_park to have a set of "parking slots" so that our
>> receptionist can park the calls in, page for someone to pickup, and then
>> that person can dial in and pick up.  That is all working well.
>>
>> I'm using Voice Operator Panel (VOP) for the receptionist software and
>> want them to be able to monitor presence on the parking slots so that they
>> can see which ones have calls parked that haven't been picked up.  I'd also
>> like them to be able to see the caller id of the call parked in each slot.
>>
>> VOP is monitoring presence fine, but for callerId it just shows "park" as
>> the number.  I traced the presence packets in the console and the problem
>> is that in the remote->identity display section of the notify packet, it
>> has "park" as the value to display, instead of the caller id number parked
>> there.
>>
>> (see example XML below).
>>
>> I have a couple of questions:
>>
>> 1.  Is it even possible to get mod_valet_park/freeswitch to report the
>> parked caller id number in presence requests in the current
>> mod_valet_park?  If so, how is it done?
>>
>> 2.  If it isn't in the current code, can anyone point me to the code
>> block that generates that remote XML section of the presence notification?
>> I poked around in the valet_send_presence function of mod_valet_parking.c,
>> but can't seem to figure out how that block is built.  I will admit that
>> I'm still getting up to speed on how Presence works in FreeSWITCH.
>>
>> Thanks for your help!
>>
>> Here is a sample notify block:
>>
>> <?xml version="1.0"?>
>>       <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="0"
>> state="full" entity="sip:park+8001 at pbx.myfreeswitchdomain.com">
>>         <dialog id="8001" direction="recipient">
>>          <state>confirmed</state>
>>           <local>
>>              <identity display="park">sip:8001 at pbx.myfreeswitchdomain.com
>> ;proto=park</identity>
>>               <target uri="sip:8001 at pbx.myfreeswitchdomain.com
>> ;proto=park">
>>                 <param pname="+sip.rendering" pvalue="no"/>
>>                </target>
>>             </local>
>>             <remote>
>>               <identity display="park">sip:8001</identity>
>>               <target uri="sip:park+8001 at pbx.myfreeswitchdomain.com"/>
>>             </remote>
>>            </dialog>
>>         </dialog-info>
>>
>>
>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From jungleboogie0 at gmail.com  Sat Apr 28 00:35:33 2018
From: jungleboogie0 at gmail.com (jungle boogie)
Date: Fri, 27 Apr 2018 17:35:33 -0700
Subject: [Freeswitch-users] freeswitch.org redirecting to freeswitch.com
Message-ID: <787cf77b-1e08-de23-3a52-9e777580ff38@gmail.com>

Hi All,

Am I wrong...it looks like freeswitch.org is redirecting to freeswitch.com

$ curl https://freeswitch.org
<!DOCTYPE HTML PUBLIC "-//IETF//DTD HTML 2.0//EN">
<html><head>
<title>301 Moved Permanently</title>
</head><body>
<h1>Moved Permanently</h1>
<p>The document has moved <a href="https://freeswitch.com">here</a>.</p>
<hr>
<address>Apache/2.4.10 (Debian) Server at freeswitch.org Port 80</address>
</body></html>

stash, jira and confluence still load, but it looks like you need to 
type in the exact location to access them.

Thanks!


From s.safarov at gmail.com  Sat Apr 28 03:50:35 2018
From: s.safarov at gmail.com (Sergey Safarov)
Date: Sat, 28 Apr 2018 03:50:35 +0000
Subject: [Freeswitch-users] freeswitch.org redirecting to freeswitch.com
In-Reply-To: <787cf77b-1e08-de23-3a52-9e777580ff38@gmail.com>
References: <787cf77b-1e08-de23-3a52-9e777580ff38@gmail.com>
Message-ID: <CAHtxdDcvweOOw8Y8L0wZTj6COUNwT51dO2p4iCgzW_is55yt+Q@mail.gmail.com>

According "whois" freeswitch.com is owned by freeswitch team

$ whois freeswitch.com
.....

Admin Name: Anthony Minessale

Admin Organization: FreeSWITCH Solutions, LLC

Admin Street: PO BOX PO BOX 2531

Admin City: Brookfield

Admin State/Province: WI

Admin Postal Code: 53008-2531

Admin Country: US

Admin Phone: +1.2132860400

Admin Phone Ext:

Admin Fax: +1.2132860401

Admin Fax Ext:

Admin Email: domainreg at freeswitch.org

Registry Tech ID:

Tech Name: Brian West

Tech Organization: FreeSWITCH Solutions, LLC

Tech Street: 714 E Osage Ave

Tech City: McAlester

Tech State/Province: OK

Tech Postal Code: 74501-6638

Tech Country: US

Tech Phone: +1.9184249378

Tech Phone Ext:

Tech Fax: +1.9184249378

Tech Fax Ext:

Tech Email: brian at freeswitch.org

Name Server: NS2.HE.NET

Name Server: NS3.HE.NET

Name Server: NS4.HE.NET

Name Server: NS5.HE.NET

Name Server: NS1.FREESWITCH.ORG

сб, 28 апр. 2018 г. в 3:51, jungle boogie <jungleboogie0 at gmail.com>:

> Hi All,
>
> Am I wrong...it looks like freeswitch.org is redirecting to freeswitch.com
>
> $ curl https://freeswitch.org
> <!DOCTYPE HTML PUBLIC "-//IETF//DTD HTML 2.0//EN">
> <html><head>
> <title>301 Moved Permanently</title>
> </head><body>
> <h1>Moved Permanently</h1>
> <p>The document has moved <a href="https://freeswitch.com">here</a>.</p>
> <hr>
> <address>Apache/2.4.10 (Debian) Server at freeswitch.org Port 80</address>
> </body></html>
>
> stash, jira and confluence still load, but it looks like you need to
> type in the exact location to access them.
>
> Thanks!
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From jungleboogie0 at gmail.com  Sat Apr 28 04:35:01 2018
From: jungleboogie0 at gmail.com (jungle boogie)
Date: Fri, 27 Apr 2018 21:35:01 -0700
Subject: [Freeswitch-users] freeswitch.org redirecting to freeswitch.com
In-Reply-To: <CAHtxdDcvweOOw8Y8L0wZTj6COUNwT51dO2p4iCgzW_is55yt+Q@mail.gmail.com>
References: <787cf77b-1e08-de23-3a52-9e777580ff38@gmail.com>
 <CAHtxdDcvweOOw8Y8L0wZTj6COUNwT51dO2p4iCgzW_is55yt+Q@mail.gmail.com>
Message-ID: <8c81cfdd-0a72-7660-f936-38551cf64f0a@gmail.com>

Thus said Sergey Safarov on Sat, 28 Apr 2018 03:50:35 +0000
> According "whois" freeswitch.com is owned by freeswitch team
> 
> $ whois freeswitch.com
> .....
> 
>

Yes, that's right. I'm not saying the two are not connected - just 
curious why/checking to see if the redirection is accurate.


From ronybeck at themenz.biz  Mon Apr 30 08:30:06 2018
From: ronybeck at themenz.biz (Ronnie Beck)
Date: Mon, 30 Apr 2018 10:30:06 +0200
Subject: [Freeswitch-users] SIP PUBLISH and custom events
Message-ID: <8b81d58e6c57aad2343e9acfc2ec597d@themenz.biz>

Hi All, 

I have a configuration question regarding SIP Events (rfc 3903).  I have
a custom VoIP client which PUBLISHes events which are custom i.e. they
are not types which a phone would understand like "presence" or
"call-info".    When my client attempts to publish such events by
sending a publish SIP request, Freeswitch replies with a "BAD EVENT"
message.  The supported messages seem to be:  talk, hold, conference,
presence, as-feature-event, dialog, line-seize, call-info, sla,
include-session-description, presence.winfo, message-summary, refer.  Is
this list configurable?  Or is there a way that I can configure the
publishing of event, the "Event" header of which I define?  Must I write
a freeswitch module for handling the subscriptions for this or can this
be handled via a configuration and perhaps an additional event socket
module which I would have to write? 

Any help is greatly appreciated. 

Cheers, 

Aaron
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From mbodbg at gmx.net  Mon Apr 30 09:37:48 2018
From: mbodbg at gmx.net (=?utf-8?Q?Markus_B=C3=B6nke?=)
Date: Mon, 30 Apr 2018 11:37:48 +0200
Subject: [Freeswitch-users] FSBlue / FreeSWITCH 1.8
Message-ID: <D1C7F7D3-4FC0-4748-AC28-9331CCE64FCF@gmx.net>

Hello,

I just saw that the freeswitch team is offering now a hosting service FSBlue (https://freeswitch.com/index.php/fs-blue/ <https://freeswitch.com/index.php/fs-blue/>). In the description it is mentioned that the manages instances are based on Freeswitch 1.8. So is there now a commercial, closed source branch for Freeswitch 1.8 and a community edition based on the 1.6 branch?

Thanks and regards

Markus
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From ssinyagin at gmail.com  Mon Apr 30 13:34:09 2018
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Mon, 30 Apr 2018 15:34:09 +0200
Subject: [Freeswitch-users] FSBlue / FreeSWITCH 1.8
In-Reply-To: <D1C7F7D3-4FC0-4748-AC28-9331CCE64FCF@gmx.net>
References: <D1C7F7D3-4FC0-4748-AC28-9331CCE64FCF@gmx.net>
Message-ID: <CACOYK=ihgwWgjojw_WZhks=bPYLa=7jTa1u+BHuePWthgrDmVw@mail.gmail.com>

1.8 is not released yet, but all source code is available in master
branch. There are also .deb packages for 1.9.0 which are made from the
master branch on a regular (daily?) basis.



On Mon, Apr 30, 2018 at 11:37 AM, Markus Bönke <mbodbg at gmx.net> wrote:
> Hello,
>
> I just saw that the freeswitch team is offering now a hosting service FSBlue
> (https://freeswitch.com/index.php/fs-blue/). In the description it is
> mentioned that the manages instances are based on Freeswitch 1.8. So is
> there now a commercial, closed source branch for Freeswitch 1.8 and a
> community edition based on the 1.6 branch?
>
> Thanks and regards
>
> Markus
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From bilaln018 at gmail.com  Mon Apr 30 13:45:22 2018
From: bilaln018 at gmail.com (Bilal Abbasi)
Date: Mon, 30 Apr 2018 13:45:22 +0000
Subject: [Freeswitch-users] [event socket][client connection halt]
In-Reply-To: <CADSVerBK8BOQkis+12qA1q5S_wyY-te4L1yiiXK+F6X=own05A@mail.gmail.com>
References: <CADSVerBK8BOQkis+12qA1q5S_wyY-te4L1yiiXK+F6X=own05A@mail.gmail.com>
Message-ID: <CADSVerA-3Yx_Sx5cswGX5AV3sctTAcuDzu33kRjuQKLsLBWZ8Q@mail.gmail.com>

Can any body help me with this please

On Sat, 28 Apr 2018 at 12:20 AM, Bilal Abbasi <bilaln018 at gmail.com> wrote:

> Hi Users,
> I came across an issue that my golang connection to event socket halts
> after few days(normally 1-2 days).
> When that halts i am not able to make calls or receive events.
> I am using below library for event socket
> https://github.com/fiorix/go-eventsocket
>
> Cloud service provider: amazon ec2
> I am just thinking its tcp connection break that is not started again by
> the library( may be some method to keep alive the port).
> Can somebody help me with this issue.
>
> Regards
> Abbasi
>
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From steveayre at gmail.com  Mon Apr 30 14:13:14 2018
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 30 Apr 2018 15:13:14 +0100
Subject: [Freeswitch-users] FSBlue / FreeSWITCH 1.8
In-Reply-To: <CACOYK=ihgwWgjojw_WZhks=bPYLa=7jTa1u+BHuePWthgrDmVw@mail.gmail.com>
References: <D1C7F7D3-4FC0-4748-AC28-9331CCE64FCF@gmx.net>
 <CACOYK=ihgwWgjojw_WZhks=bPYLa=7jTa1u+BHuePWthgrDmVw@mail.gmail.com>
Message-ID: <CAFiqYumW2Lf7iSHLss7sumE8LOKCX8wVxjUFug9L36UjYyD_vg@mail.gmail.com>

There is also a FreeSWITCH Advantage repository (
https://freeswitch.com/index.php/freeswitch-advantage/). This might be
ahead of the public master branch?

On 30 April 2018 at 14:34, Stanislav Sinyagin <ssinyagin at gmail.com> wrote:

> 1.8 is not released yet, but all source code is available in master
> branch. There are also .deb packages for 1.9.0 which are made from the
> master branch on a regular (daily?) basis.
>
>
>
> On Mon, Apr 30, 2018 at 11:37 AM, Markus Bönke <mbodbg at gmx.net> wrote:
> > Hello,
> >
> > I just saw that the freeswitch team is offering now a hosting service
> FSBlue
> > (https://freeswitch.com/index.php/fs-blue/). In the description it is
> > mentioned that the manages instances are based on Freeswitch 1.8. So is
> > there now a commercial, closed source branch for Freeswitch 1.8 and a
> > community edition based on the 1.6 branch?
> >
> > Thanks and regards
> >
> > Markus
> >
> > ____________________________________________________________
> _____________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://confluence.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at 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 alexandr.popov at iqoption.com  Mon Apr 30 14:27:57 2018
From: alexandr.popov at iqoption.com (Alexandr Popov)
Date: Mon, 30 Apr 2018 17:27:57 +0300
Subject: [Freeswitch-users] [event socket][client connection halt]
In-Reply-To: <CADSVerA-3Yx_Sx5cswGX5AV3sctTAcuDzu33kRjuQKLsLBWZ8Q@mail.gmail.com>
References: <CADSVerBK8BOQkis+12qA1q5S_wyY-te4L1yiiXK+F6X=own05A@mail.gmail.com>
 <CADSVerA-3Yx_Sx5cswGX5AV3sctTAcuDzu33kRjuQKLsLBWZ8Q@mail.gmail.com>
Message-ID: <CAKeZRhv9AmiXCMQnrDBFcOgpiRnq1gge4QZ=db=-qjNNp0guGQ@mail.gmail.com>

yeah i met it after some fs update - so i just move to another lib

i made issue but it was closed https://freeswitch.org/jira/browse/FS-10412

2018-04-30 16:45 GMT+03:00 Bilal Abbasi <bilaln018 at gmail.com>:

> Can any body help me with this please
>
> On Sat, 28 Apr 2018 at 12:20 AM, Bilal Abbasi <bilaln018 at gmail.com> wrote:
>
>> Hi Users,
>> I came across an issue that my golang connection to event socket halts
>> after few days(normally 1-2 days).
>> When that halts i am not able to make calls or receive events.
>> I am using below library for event socket
>> https://github.com/fiorix/go-eventsocket
>>
>> Cloud service provider: amazon ec2
>> I am just thinking its tcp connection break that is not started again by
>> the library( may be some method to keep alive the port).
>> Can somebody help me with this issue.
>>
>> Regards
>> Abbasi
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 bipin at xbipin.com  Mon Apr 30 14:35:27 2018
From: bipin at xbipin.com (Bipin Patel)
Date: Mon, 30 Apr 2018 18:35:27 +0400
Subject: [Freeswitch-users] FSBlue / FreeSWITCH 1.8
In-Reply-To: <CACOYK=ihgwWgjojw_WZhks=bPYLa=7jTa1u+BHuePWthgrDmVw@mail.gmail.com>
References: <D1C7F7D3-4FC0-4748-AC28-9331CCE64FCF@gmx.net>
 <CACOYK=ihgwWgjojw_WZhks=bPYLa=7jTa1u+BHuePWthgrDmVw@mail.gmail.com>
Message-ID: <5043bf3c-6c87-9eee-df1e-4ba804e7486e@xbipin.com>

hi,

doesnt look like patches pushed to 1.8 are in master

Regards,
Bipin



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

-------- Original Message  --------
Subject: Re: [Freeswitch-users] FSBlue / FreeSWITCH 1.8
From: Stanislav Sinyagin <ssinyagin at gmail.com>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Date: 4/30/2018, 5:34:09 PM

> 1.8 is not released yet, but all source code is available in master
> branch. There are also .deb packages for 1.9.0 which are made from the
> master branch on a regular (daily?) basis.
> 
> 
> 
> On Mon, Apr 30, 2018 at 11:37 AM, Markus Bönke <mbodbg at gmx.net> wrote:
>> Hello,
>>
>> I just saw that the freeswitch team is offering now a hosting service FSBlue
>> (https://freeswitch.com/index.php/fs-blue/). In the description it is
>> mentioned that the manages instances are based on Freeswitch 1.8. So is
>> there now a commercial, closed source branch for Freeswitch 1.8 and a
>> community edition based on the 1.6 branch?
>>
>> Thanks and regards
>>
>> Markus
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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
> 



From prestonh at gmail.com  Mon Apr 30 15:35:39 2018
From: prestonh at gmail.com (Preston Hagar)
Date: Mon, 30 Apr 2018 10:35:39 -0500
Subject: [Freeswitch-users] valet_park and caller Id number
In-Reply-To: <CAMtXxQnCa_maJy-3hoek950BfoSzyMLnq00fa_Mt2X+XBGG-Uw@mail.gmail.com>
References: <CAK6zN=0X06q=Xca9CgR4VxMowsxbRHe_M2uX998wJAzDNtejfA@mail.gmail.com>
 <CAK6zN=0Xyspcu6O56wqqKEeZXZriAL929GHwT6mXGQYY=U-qZA@mail.gmail.com>
 <CAMtXxQnCa_maJy-3hoek950BfoSzyMLnq00fa_Mt2X+XBGG-Uw@mail.gmail.com>
Message-ID: <CAK6zN=1PY36503pHxsB7ujx3BSNVRJCzmKdUmfLnhR9-qqK49Q@mail.gmail.com>

I'll make myself a note to try and see if I can circle back around someday
if possible and do that.  I'm happy to share the code if anyone else wants
to do the extra work of adding a config option and making a patch request.
I'm unfortunately too tied up right now trying to get my final caller ID
name bug fixed.  It seems I may be the last person still using PRIs :)  (or
at least no one uses PRIs with FreeSWITCH).



On Fri, Apr 27, 2018 at 5:06 PM, Joel Serrano <joel at gogii.net> wrote:

> Maybe you can make that functionality configurable and submit a PR to
> upstream?
>
> I’m sure someone can/will benefit from your effort.
>
> Joel.
>
> On Fri, Apr 27, 2018 at 13:47 Preston Hagar <prestonh at gmail.com> wrote:
>
>> Just thought I would send a followup in case someone ever runs into the
>> same problem and comes across this thread later.
>>
>> We ended up having to modify mod_valet_parking.c to pull in the
>> caller_id_number channel variable and then added it as an
>> event_header_string.
>>
>> Then in sofia_presence.c we edited the section that builds the presence
>> XML for parked calls and grabbed that caller ID number and put it in the
>> <remote><identity display="" section instead of just having the word "park"
>> hardcoded like it is now.
>>
>> We now get caller Id numbers on things like Voice Operator Panel by
>> having it monitor the presence on the parked extensions.
>>
>> If anyone ever needs the code, feel free to contact me and I'm happy to
>> share.
>>
>> On Thu, Apr 19, 2018 at 1:23 PM, Preston Hagar <prestonh at gmail.com>
>> wrote:
>>
>>> I've setup mod_valet_park to have a set of "parking slots" so that our
>>> receptionist can park the calls in, page for someone to pickup, and then
>>> that person can dial in and pick up.  That is all working well.
>>>
>>> I'm using Voice Operator Panel (VOP) for the receptionist software and
>>> want them to be able to monitor presence on the parking slots so that they
>>> can see which ones have calls parked that haven't been picked up.  I'd also
>>> like them to be able to see the caller id of the call parked in each slot.
>>>
>>> VOP is monitoring presence fine, but for callerId it just shows "park"
>>> as the number.  I traced the presence packets in the console and the
>>> problem is that in the remote->identity display section of the notify
>>> packet, it has "park" as the value to display, instead of the caller id
>>> number parked there.
>>>
>>> (see example XML below).
>>>
>>> I have a couple of questions:
>>>
>>> 1.  Is it even possible to get mod_valet_park/freeswitch to report the
>>> parked caller id number in presence requests in the current
>>> mod_valet_park?  If so, how is it done?
>>>
>>> 2.  If it isn't in the current code, can anyone point me to the code
>>> block that generates that remote XML section of the presence notification?
>>> I poked around in the valet_send_presence function of mod_valet_parking.c,
>>> but can't seem to figure out how that block is built.  I will admit that
>>> I'm still getting up to speed on how Presence works in FreeSWITCH.
>>>
>>> Thanks for your help!
>>>
>>> Here is a sample notify block:
>>>
>>> <?xml version="1.0"?>
>>>       <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info"
>>> version="0" state="full" entity="sip:park+8001 at pbx.
>>> myfreeswitchdomain.com">
>>>         <dialog id="8001" direction="recipient">
>>>          <state>confirmed</state>
>>>           <local>
>>>              <identity display="park">sip:8001 at pbx.
>>> myfreeswitchdomain.com;proto=park</identity>
>>>               <target uri="sip:8001 at pbx.myfreeswitchdomain.com;proto=
>>> park">
>>>                 <param pname="+sip.rendering" pvalue="no"/>
>>>                </target>
>>>             </local>
>>>             <remote>
>>>               <identity display="park">sip:8001</identity>
>>>               <target uri="sip:park+8001 at pbx.myfreeswitchdomain.com"/>
>>>             </remote>
>>>            </dialog>
>>>         </dialog-info>
>>>
>>>
>>>
>>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 ismael.blanco18 at gmail.com  Wed Apr 25 19:09:50 2018
From: ismael.blanco18 at gmail.com (Ismael Blanco)
Date: Wed, 25 Apr 2018 22:09:50 +0300
Subject: [Freeswitch-users] MoH/IVR & Play_and_get_digits or Playback
Message-ID: <CAO-EmkjF1FC+7VfDtE6trj5sJ4m2f8gNWKEdTj=e5pLEVgmtzw@mail.gmail.com>

Hello,

I'm working on a scenario where inbound calls to a DID are bridged to a
termination provider. At the same time, I'm looking to inject a whisper to
the b-leg so that agents can have knowledge on where the call is coming
from.

On a normal scenario where there is no IVR on MoH (hosted from the b-leg)
everything works as expected.

However, when there is MoH or an IVR, the caller only hears ringing and not
the actual media from the above.

I've tested a couple of different scenarios, each without success.
Both using a nolocal:execute_on_answer lua script:

scenario a) session:execute( "play_and_get_digits", "1 1 10 2000 ''
"..playbackstring );

scenario b) session:execute("set","playback_terminators=any")
                   session:execute( "playback", playbackstring )

Any ideas on how the audio can be maintained for the caller would be really
appreciated.

Thanks in advance,
Ismael
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From stefano.bertuola at athonet.com  Mon Apr 30 14:26:39 2018
From: stefano.bertuola at athonet.com (Stefano Bertuola)
Date: Mon, 30 Apr 2018 14:26:39 +0000
Subject: [Freeswitch-users] FreeSwitch not preserving the 'Via' in the
	answer to the BYE message
Message-ID: <VI1PR0202MB2736B9ADEEC20DA1DFB7DF0184820@VI1PR0202MB2736.eurprd02.prod.outlook.com>

Hi all.

I have an issue/doubt regarding Via header. The scenario is a FreeSwitch connected to a Mavenir's IMS; call is generated by FS towards Mavenir's IMS (it is ok) and release (BYE) is coming from IMS:

Request-Line: BYE sip:gw+ibcf_gateway at 192.168.242.42:5060;transport=udp;gw=ibcf_gateway SIP/2.0
Message Header
     [truncated]Via: SIP/2.0/UDP scscf.ims.mnc001.mcc001.3gppnetwork.org:5060;rport;branch=z9hG4bKmavodi-0-4c-bf-2-640020-4df45e927c208f36,SIP/2.0/UDP tas.ims.mnc001.mcc001.3gppnetwork.org:5060;received=192.168.239.37;rport=5060;branch=z9hG4bK
        Transport: UDP
        Sent-by Address: scscf.ims.mnc001.mcc001.3gppnetwork.org
        Sent-by port: 5060
        RPort: rport
        Branch: z9hG4bKmavodi-0-4c-bf-2-640020-4df45e927c208f36
        Transport: UDP
        Sent-by Address: tas.ims.mnc001.mcc001.3gppnetwork.org
        Sent-by port: 5060
        Received: 192.168.239.37
        RPort: 5060
        Branch: z9hG4bKmavodi-0-7e-76-1-ffffffff-3029420078-24076
    Max-Forwards: 69
    From: <sip:+628811451570 at ims.mnc001.mcc001.3gppnetwork.org>;tag=mavodi-0-7e-76-1-ffffffff-_000C2939994E-5e0c-a2ac4700-5df-5ae2f6d8-24075
    To: "+628811452131" <sip:+628811452131 at 192.168.242.42>;tag=HHXppypZBv2aS
    Call-ID: f4973e3c-c4a8-1236-15b0-000c29581a5f
    CSeq: 122076474 BYE
    Content-Length: 0
    P-Access-Network-Info: 3GPP-E-UTRAN-TDD;utran-cell-id-3gpp=5102800640033101
    P-Charging-Vector: icid-value=0.82.760-1524823768.21;icid-generated-at=192.168.239.35;term-ioi=Ioi1;orig-ioi=22345
    P-Charging-Function-Addresses: ccf=ims.mnc001.mcc001.3gppnetwork.org;ecf=ims.mnc001.mcc001.3gppnetwork.org;ccf=ims.mnc001.mcc001.3gppnetwork.org;ecf=ims.mnc001.mcc001.3gppnetwork.org


Status-Line: SIP/2.0 200 OK
Message Header
    Via: SIP/2.0/UDP scscf.ims.mnc001.mcc001.3gppnetwork.org:5060;rport=5060;branch=z9hG4bKmavodi-0-4c-bf-2-640020-4df45e927c208f36;received=192.168.239.36
        Transport: UDP
        Sent-by Address: scscf.ims.mnc001.mcc001.3gppnetwork.org
        Sent-by port: 5060
        RPort: 5060
        Branch: z9hG4bKmavodi-0-4c-bf-2-640020-4df45e927c208f36
        Received: 192.168.239.36
    From: <sip:+628811451570 at ims.mnc001.mcc001.3gppnetwork.org>;tag=mavodi-0-7e-76-1-ffffffff-_000C2939994E-5e0c-a2ac4700-5df-5ae2f6d8-24075
    To: "+628811452131" <sip:+628811452131 at 192.168.242.42>;tag=HHXppypZBv2aS
    Call-ID: f4973e3c-c4a8-1236-15b0-000c29581a5f
    CSeq: 122076474 BYE
    User-Agent: FreeSwitch
    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
    Supported: timer, path, replaces
    Content-Length: 0

In the 200 OK replied back by FS to the BYE message, part of the 'Via' header is "missing".

According to RFC 3261:  "The Via header field values in the response MUST equal the Via header field values in the request and MUST maintain the same ordering".

Is it a expected behavior? Any variable to set in order to get the "original" 'Via' header in the answer?

Br. Stefano
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From mike at jerris.com  Mon Apr 30 16:12:01 2018
From: mike at jerris.com (Michael Jerris)
Date: Mon, 30 Apr 2018 12:12:01 -0400
Subject: [Freeswitch-users] freeswitch.org redirecting to freeswitch.com
In-Reply-To: <8c81cfdd-0a72-7660-f936-38551cf64f0a@gmail.com>
References: <787cf77b-1e08-de23-3a52-9e777580ff38@gmail.com>
 <CAHtxdDcvweOOw8Y8L0wZTj6COUNwT51dO2p4iCgzW_is55yt+Q@mail.gmail.com>
 <8c81cfdd-0a72-7660-f936-38551cf64f0a@gmail.com>
Message-ID: <F59D925A-8F05-46B2-954D-34B3348B5401@jerris.com>

It’s accurate, just doing some rework at unifying our web presence.  If there is anything you can't get to easily please let us know, we tried to make sure everything was working correctly after the changes.

> On Apr 28, 2018, at 12:35 AM, jungle boogie <jungleboogie0 at gmail.com> wrote:
> 
> Thus said Sergey Safarov on Sat, 28 Apr 2018 03:50:35 +0000
>> According "whois" freeswitch.com is owned by freeswitch team
>> $ whois freeswitch.com
>> .....
>> 
> 
> Yes, that's right. I'm not saying the two are not connected - just curious why/checking to see if the redirection is accurate.



From mike at jerris.com  Mon Apr 30 16:15:52 2018
From: mike at jerris.com (Michael Jerris)
Date: Mon, 30 Apr 2018 12:15:52 -0400
Subject: [Freeswitch-users] FSBlue / FreeSWITCH 1.8
In-Reply-To: <CAFiqYumW2Lf7iSHLss7sumE8LOKCX8wVxjUFug9L36UjYyD_vg@mail.gmail.com>
References: <D1C7F7D3-4FC0-4748-AC28-9331CCE64FCF@gmx.net>
 <CACOYK=ihgwWgjojw_WZhks=bPYLa=7jTa1u+BHuePWthgrDmVw@mail.gmail.com>
 <CAFiqYumW2Lf7iSHLss7sumE8LOKCX8wVxjUFug9L36UjYyD_vg@mail.gmail.com>
Message-ID: <A4CE9A9D-5C4D-4994-843F-58F81CEC700A@jerris.com>

FreeSWITCH Advantage does indeed have fixes that are not yet released into our public repositories.  One of the advantages to FreeSWITCH Advantage would be that you can advanced access to some fixes and releases.  All of these will eventually make it down to the public releases, but on a delayed basis.

Mike


> On Apr 30, 2018, at 10:13 AM, Steven Ayre <steveayre at gmail.com> wrote:
> 
> There is also a FreeSWITCH Advantage repository (https://freeswitch.com/index.php/freeswitch-advantage/ <https://freeswitch.com/index.php/freeswitch-advantage/>). This might be ahead of the public master branch?
> 
> On 30 April 2018 at 14:34, Stanislav Sinyagin <ssinyagin at gmail.com <mailto:ssinyagin at gmail.com>> wrote:
> 1.8 is not released yet, but all source code is available in master
> branch. There are also .deb packages for 1.9.0 which are made from the
> master branch on a regular (daily?) basis.
> 
> 
> 
> On Mon, Apr 30, 2018 at 11:37 AM, Markus Bönke <mbodbg at gmx.net <mailto:mbodbg at gmx.net>> wrote:
> > Hello,
> >
> > I just saw that the freeswitch team is offering now a hosting service FSBlue
> > (https://freeswitch.com/index.php/fs-blue/ <https://freeswitch.com/index.php/fs-blue/>). In the description it is
> > mentioned that the manages instances are based on Freeswitch 1.8. So is
> > there now a commercial, closed source branch for Freeswitch 1.8 and a
> > community edition based on the 1.6 branch?
> >
> > Thanks and regards
> >
> > Markus
> >
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From mike at jerris.com  Mon Apr 30 16:17:47 2018
From: mike at jerris.com (Michael Jerris)
Date: Mon, 30 Apr 2018 12:17:47 -0400
Subject: [Freeswitch-users] [event socket][client connection halt]
In-Reply-To: <CAKeZRhv9AmiXCMQnrDBFcOgpiRnq1gge4QZ=db=-qjNNp0guGQ@mail.gmail.com>
References: <CADSVerBK8BOQkis+12qA1q5S_wyY-te4L1yiiXK+F6X=own05A@mail.gmail.com>
 <CADSVerA-3Yx_Sx5cswGX5AV3sctTAcuDzu33kRjuQKLsLBWZ8Q@mail.gmail.com>
 <CAKeZRhv9AmiXCMQnrDBFcOgpiRnq1gge4QZ=db=-qjNNp0guGQ@mail.gmail.com>
Message-ID: <75E555AC-05BD-40E9-9E5B-484351D536E0@jerris.com>

Seems to be some sort of issue with the client library from the information I’ve seen.

> On Apr 30, 2018, at 10:27 AM, Alexandr Popov <alexandr.popov at iqoption.com> wrote:
> 
> yeah i met it after some fs update - so i just move to another lib
> 
> i made issue but it was closed https://freeswitch.org/jira/browse/FS-10412 <https://freeswitch.org/jira/browse/FS-10412>
> 
> 2018-04-30 16:45 GMT+03:00 Bilal Abbasi <bilaln018 at gmail.com <mailto:bilaln018 at gmail.com>>:
> Can any body help me with this please
> 
> On Sat, 28 Apr 2018 at 12:20 AM, Bilal Abbasi <bilaln018 at gmail.com <mailto:bilaln018 at gmail.com>> wrote:
> Hi Users,
> I came across an issue that my golang connection to event socket halts after few days(normally 1-2 days).
> When that halts i am not able to make calls or receive events.
> I am using below library for event socket
> https://github.com/fiorix/go-eventsocket <https://github.com/fiorix/go-eventsocket>
> 
> Cloud service provider: amazon ec2
> I am just thinking its tcp connection break that is not started again by the library( may be some method to keep alive the port).
> Can somebody help me with this issue.
> 
> Regards
> Abbasi

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From bilaln018 at gmail.com  Mon Apr 30 16:43:17 2018
From: bilaln018 at gmail.com (Bilal Abbasi)
Date: Mon, 30 Apr 2018 16:43:17 +0000
Subject: [Freeswitch-users] [event socket][client connection halt]
In-Reply-To: <CAKeZRhv9AmiXCMQnrDBFcOgpiRnq1gge4QZ=db=-qjNNp0guGQ@mail.gmail.com>
References: <CADSVerBK8BOQkis+12qA1q5S_wyY-te4L1yiiXK+F6X=own05A@mail.gmail.com>
 <CADSVerA-3Yx_Sx5cswGX5AV3sctTAcuDzu33kRjuQKLsLBWZ8Q@mail.gmail.com>
 <CAKeZRhv9AmiXCMQnrDBFcOgpiRnq1gge4QZ=db=-qjNNp0guGQ@mail.gmail.com>
Message-ID: <CADSVerD01T8PWeO=mwt=PYZryjPfbwC5NuTuTRxBd3JWEd=sog@mail.gmail.com>

Can i ask which library you shifted to, i am planing to move to
https://freeswitch.org/confluence/plugins/servlet/mobile#content/view/7143958

Regards
Abbasi

On Mon, 30 Apr 2018 at 8:23 PM, Alexandr Popov <alexandr.popov at iqoption.com>
wrote:

> yeah i met it after some fs update - so i just move to another lib
>
> i made issue but it was closed https://freeswitch.org/jira/browse/FS-10412
>
> 2018-04-30 16:45 GMT+03:00 Bilal Abbasi <bilaln018 at gmail.com>:
>
>> Can any body help me with this please
>>
>> On Sat, 28 Apr 2018 at 12:20 AM, Bilal Abbasi <bilaln018 at gmail.com>
>> wrote:
>>
>>> Hi Users,
>>> I came across an issue that my golang connection to event socket halts
>>> after few days(normally 1-2 days).
>>> When that halts i am not able to make calls or receive events.
>>> I am using below library for event socket
>>> https://github.com/fiorix/go-eventsocket
>>>
>>> Cloud service provider: amazon ec2
>>> I am just thinking its tcp connection break that is not started again by
>>> the library( may be some method to keep alive the port).
>>> Can somebody help me with this issue.
>>>
>>> Regards
>>> Abbasi
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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 mbodbg at gmx.net  Mon Apr 30 17:21:11 2018
From: mbodbg at gmx.net (=?utf-8?Q?Markus_B=C3=B6nke?=)
Date: Mon, 30 Apr 2018 19:21:11 +0200
Subject: [Freeswitch-users] FSBlue / FreeSWITCH 1.8
In-Reply-To: <A4CE9A9D-5C4D-4994-843F-58F81CEC700A@jerris.com>
References: <D1C7F7D3-4FC0-4748-AC28-9331CCE64FCF@gmx.net>
 <CACOYK=ihgwWgjojw_WZhks=bPYLa=7jTa1u+BHuePWthgrDmVw@mail.gmail.com>
 <CAFiqYumW2Lf7iSHLss7sumE8LOKCX8wVxjUFug9L36UjYyD_vg@mail.gmail.com>
 <A4CE9A9D-5C4D-4994-843F-58F81CEC700A@jerris.com>
Message-ID: <4DAACD22-C0C9-4FA1-BD08-93775CBDAF80@gmx.net>

OK, thanks for clarification.

Markus

> Am 30.04.2018 um 18:15 schrieb Michael Jerris <mike at jerris.com>:
> 
> FreeSWITCH Advantage does indeed have fixes that are not yet released into our public repositories.  One of the advantages to FreeSWITCH Advantage would be that you can advanced access to some fixes and releases.  All of these will eventually make it down to the public releases, but on a delayed basis.
> 
> Mike
> 
> 
>> On Apr 30, 2018, at 10:13 AM, Steven Ayre <steveayre at gmail.com <mailto:steveayre at gmail.com>> wrote:
>> 
>> There is also a FreeSWITCH Advantage repository (https://freeswitch.com/index.php/freeswitch-advantage/ <https://freeswitch.com/index.php/freeswitch-advantage/>). This might be ahead of the public master branch?
>> 
>> On 30 April 2018 at 14:34, Stanislav Sinyagin <ssinyagin at gmail.com <mailto:ssinyagin at gmail.com>> wrote:
>> 1.8 is not released yet, but all source code is available in master
>> branch. There are also .deb packages for 1.9.0 which are made from the
>> master branch on a regular (daily?) basis.
>> 
>> 
>> 
>> On Mon, Apr 30, 2018 at 11:37 AM, Markus Bönke <mbodbg at gmx.net <mailto:mbodbg at gmx.net>> wrote:
>> > Hello,
>> >
>> > I just saw that the freeswitch team is offering now a hosting service FSBlue
>> > (https://freeswitch.com/index.php/fs-blue/ <https://freeswitch.com/index.php/fs-blue/>). In the description it is
>> > mentioned that the manages instances are based on Freeswitch 1.8. So is
>> > there now a commercial, closed source branch for Freeswitch 1.8 and a
>> > community edition based on the 1.6 branch?
>> >
>> > Thanks and regards
>> >
>> > Markus
>> >
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 ssinyagin at gmail.com  Mon Apr 30 19:26:34 2018
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Mon, 30 Apr 2018 21:26:34 +0200
Subject: [Freeswitch-users] FSBlue / FreeSWITCH 1.8
In-Reply-To: <A4CE9A9D-5C4D-4994-843F-58F81CEC700A@jerris.com>
References: <D1C7F7D3-4FC0-4748-AC28-9331CCE64FCF@gmx.net>
 <CACOYK=ihgwWgjojw_WZhks=bPYLa=7jTa1u+BHuePWthgrDmVw@mail.gmail.com>
 <CAFiqYumW2Lf7iSHLss7sumE8LOKCX8wVxjUFug9L36UjYyD_vg@mail.gmail.com>
 <A4CE9A9D-5C4D-4994-843F-58F81CEC700A@jerris.com>
Message-ID: <CACOYK=inDn6pacBMHtwuaBixPpwc2734+Bwg3mNge+p5QTHPAg@mail.gmail.com>

well, that's a surprising move. Did you consider that this idea would
discourage the community from contributing and bug reporting?

On Mon, Apr 30, 2018 at 6:15 PM, Michael Jerris <mike at jerris.com> wrote:
> FreeSWITCH Advantage does indeed have fixes that are not yet released into
> our public repositories.  One of the advantages to FreeSWITCH Advantage
> would be that you can advanced access to some fixes and releases.  All of
> these will eventually make it down to the public releases, but on a delayed
> basis.
>
> Mike
>
>
> On Apr 30, 2018, at 10:13 AM, Steven Ayre <steveayre at gmail.com> wrote:
>
> There is also a FreeSWITCH Advantage repository
> (https://freeswitch.com/index.php/freeswitch-advantage/). This might be
> ahead of the public master branch?
>
> On 30 April 2018 at 14:34, Stanislav Sinyagin <ssinyagin at gmail.com> wrote:
>>
>> 1.8 is not released yet, but all source code is available in master
>> branch. There are also .deb packages for 1.9.0 which are made from the
>> master branch on a regular (daily?) basis.
>>
>>
>>
>> On Mon, Apr 30, 2018 at 11:37 AM, Markus Bönke <mbodbg at gmx.net> wrote:
>> > Hello,
>> >
>> > I just saw that the freeswitch team is offering now a hosting service
>> > FSBlue
>> > (https://freeswitch.com/index.php/fs-blue/). In the description it is
>> > mentioned that the manages instances are based on Freeswitch 1.8. So is
>> > there now a commercial, closed source branch for Freeswitch 1.8 and a
>> > community edition based on the 1.6 branch?
>> >
>> > Thanks and regards
>> >
>> > Markus
>> >
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From ksrigo at gmail.com  Mon Apr 30 19:58:55 2018
From: ksrigo at gmail.com (Srigo Kanapathipillai)
Date: Mon, 30 Apr 2018 21:58:55 +0200
Subject: [Freeswitch-users] MoH/IVR & Play_and_get_digits or Playback
In-Reply-To: <CAO-EmkjF1FC+7VfDtE6trj5sJ4m2f8gNWKEdTj=e5pLEVgmtzw@mail.gmail.com>
References: <CAO-EmkjF1FC+7VfDtE6trj5sJ4m2f8gNWKEdTj=e5pLEVgmtzw@mail.gmail.com>
Message-ID: <88a7ce7e-75dd-4e90-b7f3-bfcc85523afb@gmail.com>

Hi,

Did you try session.answer(); before playing and get digits?

Srigo


On 30 Apr 2018, 19:13, at 19:13, Ismael Blanco <ismael.blanco18 at gmail.com> wrote:
>Hello,
>
>I'm working on a scenario where inbound calls to a DID are bridged to a
>termination provider. At the same time, I'm looking to inject a whisper
>to
>the b-leg so that agents can have knowledge on where the call is coming
>from.
>
>On a normal scenario where there is no IVR on MoH (hosted from the
>b-leg)
>everything works as expected.
>
>However, when there is MoH or an IVR, the caller only hears ringing and
>not
>the actual media from the above.
>
>I've tested a couple of different scenarios, each without success.
>Both using a nolocal:execute_on_answer lua script:
>
>scenario a) session:execute( "play_and_get_digits", "1 1 10 2000 ''
>"..playbackstring );
>
>scenario b) session:execute("set","playback_terminators=any")
>                   session:execute( "playback", playbackstring )
>
>Any ideas on how the audio can be maintained for the caller would be
>really
>appreciated.
>
>Thanks in advance,
>Ismael
>
>
>------------------------------------------------------------------------
>
>_________________________________________________________________________
>Professional FreeSWITCH Consulting Services:
>consulting at freeswitch.org
>http://www.freeswitchsolutions.com
>
>Official FreeSWITCH Sites
>http://www.freeswitch.org
>http://confluence.freeswitch.org
>http://www.cluecon.com
>
>FreeSWITCH-users mailing list
>FreeSWITCH-users at 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 nneul at mst.edu  Mon Apr 30 20:17:21 2018
From: nneul at mst.edu (Nathan Neulinger)
Date: Mon, 30 Apr 2018 15:17:21 -0500
Subject: [Freeswitch-users] FSBlue / FreeSWITCH 1.8
In-Reply-To: <CACOYK=inDn6pacBMHtwuaBixPpwc2734+Bwg3mNge+p5QTHPAg@mail.gmail.com>
References: <D1C7F7D3-4FC0-4748-AC28-9331CCE64FCF@gmx.net>
 <CACOYK=ihgwWgjojw_WZhks=bPYLa=7jTa1u+BHuePWthgrDmVw@mail.gmail.com>
 <CAFiqYumW2Lf7iSHLss7sumE8LOKCX8wVxjUFug9L36UjYyD_vg@mail.gmail.com>
 <A4CE9A9D-5C4D-4994-843F-58F81CEC700A@jerris.com>
 <CACOYK=inDn6pacBMHtwuaBixPpwc2734+Bwg3mNge+p5QTHPAg@mail.gmail.com>
Message-ID: <eb8631b8-cef2-2acb-e11c-cdc6aa237d7c@mst.edu>

You should file a jira for that... Bugs aren't discussed on the mailing list. ;)

-- Nathan

On 4/30/18 2:26 PM, Stanislav Sinyagin wrote:
> well, that's a surprising move. Did you consider that this idea would
> discourage the community from contributing and bug reporting?
>
> On Mon, Apr 30, 2018 at 6:15 PM, Michael Jerris <mike at jerris.com> wrote:
>> FreeSWITCH Advantage does indeed have fixes that are not yet released into
>> our public repositories.  One of the advantages to FreeSWITCH Advantage
>> would be that you can advanced access to some fixes and releases.  All of
>> these will eventually make it down to the public releases, but on a delayed
>> basis.
>>
>> Mike
>>
>>
>> On Apr 30, 2018, at 10:13 AM, Steven Ayre <steveayre at gmail.com> wrote:
>>
>> There is also a FreeSWITCH Advantage repository
>> (https://freeswitch.com/index.php/freeswitch-advantage/). This might be
>> ahead of the public master branch?
>>
>> On 30 April 2018 at 14:34, Stanislav Sinyagin <ssinyagin at gmail.com> wrote:
>>> 1.8 is not released yet, but all source code is available in master
>>> branch. There are also .deb packages for 1.9.0 which are made from the
>>> master branch on a regular (daily?) basis.
>>>
>>>
>>>
>>> On Mon, Apr 30, 2018 at 11:37 AM, Markus Bönke <mbodbg at gmx.net> wrote:
>>>> Hello,
>>>>
>>>> I just saw that the freeswitch team is offering now a hosting service
>>>> FSBlue
>>>> (https://freeswitch.com/index.php/fs-blue/). In the description it is
>>>> mentioned that the manages instances are based on Freeswitch 1.8. So is
>>>> there now a commercial, closed source branch for Freeswitch 1.8 and a
>>>> community edition based on the 1.6 branch?
>>>>
>>>> Thanks and regards
>>>>
>>>> Markus
>>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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

-- 
------------------------------------------------------------
Nathan Neulinger                       nneul at mst.edu
Missouri S&T Information Technology    (573) 612-1412
System Administrator - Architect

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From jungleboogie0 at gmail.com  Mon Apr 30 20:27:47 2018
From: jungleboogie0 at gmail.com (jungle Boogie)
Date: Mon, 30 Apr 2018 13:27:47 -0700
Subject: [Freeswitch-users] FSBlue / FreeSWITCH 1.8
In-Reply-To: <CACOYK=inDn6pacBMHtwuaBixPpwc2734+Bwg3mNge+p5QTHPAg@mail.gmail.com>
References: <D1C7F7D3-4FC0-4748-AC28-9331CCE64FCF@gmx.net>
 <CACOYK=ihgwWgjojw_WZhks=bPYLa=7jTa1u+BHuePWthgrDmVw@mail.gmail.com>
 <CAFiqYumW2Lf7iSHLss7sumE8LOKCX8wVxjUFug9L36UjYyD_vg@mail.gmail.com>
 <A4CE9A9D-5C4D-4994-843F-58F81CEC700A@jerris.com>
 <CACOYK=inDn6pacBMHtwuaBixPpwc2734+Bwg3mNge+p5QTHPAg@mail.gmail.com>
Message-ID: <CAKE2PDt-7dfXGvPFAQ0Wz3KoJ+=SnBKE24b3YVPBWPQWYrydgg@mail.gmail.com>

On 30 April 2018 at 12:26, Stanislav Sinyagin <ssinyagin at gmail.com> wrote:
> well, that's a surprising move. Did you consider that this idea would
> discourage the community from contributing and bug reporting?
>
> On Mon, Apr 30, 2018 at 6:15 PM, Michael Jerris <mike at jerris.com> wrote:
>> FreeSWITCH Advantage does indeed have fixes that are not yet released into
>> our public repositories.  One of the advantages to FreeSWITCH Advantage
>> would be that you can advanced access to some fixes and releases.  All of
>> these will eventually make it down to the public releases, but on a delayed
>> basis.


I think perhaps a blog post could be made to announce this and/or a
cluecon call to discuss all these changes.


From steveayre at gmail.com  Mon Apr 30 20:53:45 2018
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 30 Apr 2018 21:53:45 +0100
Subject: [Freeswitch-users] FSBlue / FreeSWITCH 1.8
In-Reply-To: <A4CE9A9D-5C4D-4994-843F-58F81CEC700A@jerris.com>
References: <D1C7F7D3-4FC0-4748-AC28-9331CCE64FCF@gmx.net>
 <CACOYK=ihgwWgjojw_WZhks=bPYLa=7jTa1u+BHuePWthgrDmVw@mail.gmail.com>
 <CAFiqYumW2Lf7iSHLss7sumE8LOKCX8wVxjUFug9L36UjYyD_vg@mail.gmail.com>
 <A4CE9A9D-5C4D-4994-843F-58F81CEC700A@jerris.com>
Message-ID: <CAFiqYukb_oSdcCEOMt3EU17JTU8AcK0kNd7YMxORUTNemFyx=w@mail.gmail.com>

When you say on a 'delayed' basis, what timeframe are we talking about?


On 30 April 2018 at 17:15, Michael Jerris <mike at jerris.com> wrote:

> FreeSWITCH Advantage does indeed have fixes that are not yet released into
> our public repositories.  One of the advantages to FreeSWITCH Advantage
> would be that you can advanced access to some fixes and releases.  All of
> these will eventually make it down to the public releases, but on a delayed
> basis.
>
> Mike
>
>
> On Apr 30, 2018, at 10:13 AM, Steven Ayre <steveayre at gmail.com> wrote:
>
> There is also a FreeSWITCH Advantage repository (
> https://freeswitch.com/index.php/freeswitch-advantage/). This might be
> ahead of the public master branch?
>
> On 30 April 2018 at 14:34, Stanislav Sinyagin <ssinyagin at gmail.com> wrote:
>
>> 1.8 is not released yet, but all source code is available in master
>> branch. There are also .deb packages for 1.9.0 which are made from the
>> master branch on a regular (daily?) basis.
>>
>>
>>
>> On Mon, Apr 30, 2018 at 11:37 AM, Markus Bönke <mbodbg at gmx.net> wrote:
>> > Hello,
>> >
>> > I just saw that the freeswitch team is offering now a hosting service
>> FSBlue
>> > (https://freeswitch.com/index.php/fs-blue/). In the description it is
>> > mentioned that the manages instances are based on Freeswitch 1.8. So is
>> > there now a commercial, closed source branch for Freeswitch 1.8 and a
>> > community edition based on the 1.6 branch?
>> >
>> > Thanks and regards
>> >
>> > Markus
>> >
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 darren at dmmhosting.co.uk  Mon Apr 30 21:04:12 2018
From: darren at dmmhosting.co.uk (Darren Williams)
Date: Mon, 30 Apr 2018 22:04:12 +0100
Subject: [Freeswitch-users] FSBlue / FreeSWITCH 1.8
Message-ID: <1814008362-3872@mail.email-me.co.uk>

Look at it on the positive side, back in the day when working in the enterprise we would not consider rolling an upgrade until the fix had been in the field a good few months.


Strangely, these Freeswitch Advantage people become the 'testers' for the rest of the community, by the time the community gets these fixes they should be solid and well tested.



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From mike at jerris.com  Mon Apr 30 21:16:41 2018
From: mike at jerris.com (Michael Jerris)
Date: Mon, 30 Apr 2018 17:16:41 -0400
Subject: [Freeswitch-users] FSBlue / FreeSWITCH 1.8
In-Reply-To: <CACOYK=inDn6pacBMHtwuaBixPpwc2734+Bwg3mNge+p5QTHPAg@mail.gmail.com>
References: <D1C7F7D3-4FC0-4748-AC28-9331CCE64FCF@gmx.net>
 <CACOYK=ihgwWgjojw_WZhks=bPYLa=7jTa1u+BHuePWthgrDmVw@mail.gmail.com>
 <CAFiqYumW2Lf7iSHLss7sumE8LOKCX8wVxjUFug9L36UjYyD_vg@mail.gmail.com>
 <A4CE9A9D-5C4D-4994-843F-58F81CEC700A@jerris.com>
 <CACOYK=inDn6pacBMHtwuaBixPpwc2734+Bwg3mNge+p5QTHPAg@mail.gmail.com>
Message-ID: <B11E89B5-E009-4CEE-8686-402054512B7A@jerris.com>

We think we’ve found a good balance that allows for customers to get early access to the core teams work, while still proving for a vibrant open source experience and allow everyone access to public contributions in a timely manner.  All code submitted through pull requests of course remains 100% open source, and the core teams contributions will continue to make their way into the open source releases.  This helps solidify the stability and funding for the project, while maintaining its availability as open source software for all.  We have seen that the open source contributions continue to be a vibrant and dynamic addition to the FreeSWITCH ecosystem as they always have been, and we look forward to more of the same in the future.  If you have any questions or concerns, feel free to reach out to me privately and I would be happy to discuss.

Mike

> On Apr 30, 2018, at 3:26 PM, Stanislav Sinyagin <ssinyagin at gmail.com> wrote:
> 
> well, that's a surprising move. Did you consider that this idea would
> discourage the community from contributing and bug reporting?
> 
> On Mon, Apr 30, 2018 at 6:15 PM, Michael Jerris <mike at jerris.com> wrote:
>> FreeSWITCH Advantage does indeed have fixes that are not yet released into
>> our public repositories.  One of the advantages to FreeSWITCH Advantage
>> would be that you can advanced access to some fixes and releases.  All of
>> these will eventually make it down to the public releases, but on a delayed
>> basis.
>> 
>> Mike
>> 
>> 
>> On Apr 30, 2018, at 10:13 AM, Steven Ayre <steveayre at gmail.com> wrote:
>> 
>> There is also a FreeSWITCH Advantage repository
>> (https://freeswitch.com/index.php/freeswitch-advantage/). This might be
>> ahead of the public master branch?
>> 
>> On 30 April 2018 at 14:34, Stanislav Sinyagin <ssinyagin at gmail.com> wrote:
>>> 
>>> 1.8 is not released yet, but all source code is available in master
>>> branch. There are also .deb packages for 1.9.0 which are made from the
>>> master branch on a regular (daily?) basis.
>>> 
>>> 
>>> 
>>> On Mon, Apr 30, 2018 at 11:37 AM, Markus Bönke <mbodbg at gmx.net> wrote:
>>>> Hello,
>>>> 
>>>> I just saw that the freeswitch team is offering now a hosting service
>>>> FSBlue
>>>> (https://freeswitch.com/index.php/fs-blue/). In the description it is
>>>> mentioned that the manages instances are based on Freeswitch 1.8. So is
>>>> there now a commercial, closed source branch for Freeswitch 1.8 and a
>>>> community edition based on the 1.6 branch?
>>>> 
>>>> Thanks and regards
>>>> 
>>>> Markus
>>>> 
>> 



From jungleboogie0 at gmail.com  Mon Apr 30 23:13:33 2018
From: jungleboogie0 at gmail.com (jungle Boogie)
Date: Mon, 30 Apr 2018 16:13:33 -0700
Subject: [Freeswitch-users] FSBlue / FreeSWITCH 1.8
In-Reply-To: <B11E89B5-E009-4CEE-8686-402054512B7A@jerris.com>
References: <D1C7F7D3-4FC0-4748-AC28-9331CCE64FCF@gmx.net>
 <CACOYK=ihgwWgjojw_WZhks=bPYLa=7jTa1u+BHuePWthgrDmVw@mail.gmail.com>
 <CAFiqYumW2Lf7iSHLss7sumE8LOKCX8wVxjUFug9L36UjYyD_vg@mail.gmail.com>
 <A4CE9A9D-5C4D-4994-843F-58F81CEC700A@jerris.com>
 <CACOYK=inDn6pacBMHtwuaBixPpwc2734+Bwg3mNge+p5QTHPAg@mail.gmail.com>
 <B11E89B5-E009-4CEE-8686-402054512B7A@jerris.com>
Message-ID: <CAKE2PDu2jhk_3vywjUzDsd_97R6UoDJ_nnNG5+pXmY5Huyu0yA@mail.gmail.com>

On 30 April 2018 at 14:16, Michael Jerris <mike at jerris.com> wrote:
> We think we’ve found a good balance that allows for customers to get early access to the core teams work, while still proving for a vibrant open source experience and allow everyone access to public contributions in a timely manner.  All code submitted through pull requests of course remains 100% open source, and the core teams contributions will continue to make their way into the open source releases.  This helps solidify the stability and funding for the project, while maintaining its availability as open source software for all.  We have seen that the open source contributions continue to be a vibrant and dynamic addition to the FreeSWITCH ecosystem as they always have been, and we look forward to more of the same in the future.  If you have any questions or concerns, feel free to reach out to me privately and I would be happy to discuss.
>

With freeswitch.org being redirected to your freeswitch.com, are you
no longer taking donations?
https://freeswitch.com/donate

That doesn't exist.

I think most of us want to see the company grow and foster and see
your customer list increase, it's already quite impressive. How about
a cluecon call that talks about your current customers and your future
plans with FS advantage?

> Mike
>